Lecture Notes

Subject: Data Communication (CIE3304)

College of Electronic Engineering
Information and Computer Engineering
3rd Year

Azhar Sabah Abdulaziz

October 16, 2019

Contents

1 Data Communication Principles 3

1.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.2 Signal Definition . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.2.1 Continuous and Discrete . . . . . . . . . . . . . . . . . 4
1.2.2 Analog and Digital . . . . . . . . . . . . . . . . . . . . 4
1.2.3 Periodic and Non-periodic . . . . . . . . . . . . . . . . 5
1.2.4 Energy and Power Signals . . . . . . . . . . . . . . . . 6
1.2.5 Deterministic and probabilistic signal . . . . . . . . . . 7
1.3 Communication Channels . . . . . . . . . . . . . . . . . . . . 7
1.3.1 Wireline Channels . . . . . . . . . . . . . . . . . . . . 8
1.3.2 Fiber Optic Channels . . . . . . . . . . . . . . . . . . . 10
1.3.3 Wireless Electromagnetic Channels . . . . . . . . . . . 10
1.3.4 Underwater Acoustic Channels . . . . . . . . . . . . . 14
1.3.5 Storage Channels . . . . . . . . . . . . . . . . . . . . . 14
1.4 Multiple Access to the Channel . . . . . . . . . . . . . . . . . 16
1.4.1 Time Division Multiple Access (TDMA) . . . . . . . . 16
1.4.2 Frequency Division Multiple Access (FDMA) . . . . . . 17
1.4.3 Hybrid TDMA-FDMA . . . . . . . . . . . . . . . . . . 18
1.4.4 Code Division Multiple Access (CDMA) . . . . . . . . 19

2 Discrete Pulse Modulation 21

2.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
2.2 Sampling Theory . . . . . . . . . . . . . . . . . . . . . . . . . 21
2.3 Nyquest Theorem and Aliasing . . . . . . . . . . . . . . . . . 24
2.4 Pulse Amplitude Modulation . . . . . . . . . . . . . . . . . . . 28
2.5 Pulse Width Modulation . . . . . . . . . . . . . . . . . . . . . 31
2.6 Time Division Multiplexing . . . . . . . . . . . . . . . . . . . 33
2.7 Quantization . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
2.7.1 Uniform Quantization . . . . . . . . . . . . . . . . . . 35
2.7.2 Non-uniform quantization . . . . . . . . . . . . . . . . 37
2.8 Encoding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

1
CONTENTS CONTENTS

2.9 The Analog to Digital Converter . . . . . . . . . . . . . . . . . 39

2.10 Pulse Code Modulation . . . . . . . . . . . . . . . . . . . . . . 40
2.10.1 PCM Signal to Noise Ratio (SNR) . . . . . . . . . . . 40

3 Digital Transmission and Modulation 43

3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
3.2 Digital Baseband Transmission . . . . . . . . . . . . . . . . . 44
3.3 Intersymbol Interference (ISI) . . . . . . . . . . . . . . . . . . 47
3.4 Pulse Shapes . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
3.5 Passband Data Transmission . . . . . . . . . . . . . . . . . . . 55
3.5.1 Amplitude Shift Keying (ASK) . . . . . . . . . . . . . 56
3.5.2 Frequency Shift Keying . . . . . . . . . . . . . . . . . . 57
3.5.3 Phase Shift Keying (PSK) . . . . . . . . . . . . . . . . 58
3.5.4 Differential Phase-Shift Keying (DPSK) . . . . . . . . 62
3.6 Multi-level Digital Modulation Schemes . . . . . . . . . . . . . 63
3.6.1 Quadrature Phase-Shift Keying (QPSK) . . . . . . . . 64
3.6.2 M-Ary PSK . . . . . . . . . . . . . . . . . . . . . . . . 69
3.6.3 Quadrature Amplitude Modulation (QAM) . . . . . . . 73

2
Chapter 1

Data Communication
Principles

1.1 Introduction
The goal of any communication system is to deliver the information from
the source to the receiver in an inexpensive, fast, and a reliable procedure.
There are many factors that have to be considered in designing such a system,
which are: available bandwidth, signal-to-noise ratio (SNR), and the design
complexity. Whereas the first two factors are related to the regulations or
uncontrolled environmental specifications, the complexity is in the hand of
the engineer who is responsible to reduce the cost, and hence increase his
company’s profit.
Basically, any communication system has to have a transmitter, channel
and a receiver part.

1.2 Signal Definition

A signal is function that convey information about systems or attributes of
some phenomena. Signal is not necessarily an electrical quantities, may it
physical quantities such as sound wave, light wave, and earthquake wave.
However, to perform activities such as synthesizing, transporting, recording,
analyzing and modifying the signal, it is often utilize the signal in electrical
quantity.

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Figure 1.1: Communication system pattern.

Alternatively, signal is the physical quantity by which information is car-

ried between at least two individuals devices. In other words, signal by
which physical quantity communication is done. In electrical and electronic
communication systems, signals could be:
• Continuous and discrete time signals.

• Analog and digital signals (continuous and discrete in amplitude)

• Periodic and aperiodic signals

• Energy and power signals

• Deterministic and probabilistic signals

1.2.1 Continuous and Discrete

An analog signal x(t) is a continuous function of time is defined for all t.
It could represent different physical quantities in the electrical domain, like
speech signal captured by a microphone. The analog signal would be sampled
in time, so that it will be digitize later. Sampling and digitizing are known
as Analog-to-Digital conversion process (ADC).

1.2.2 Analog and Digital

A signal whose amplitude can take any value in a continuous range (different
values) is an analog signal. This means that an analog signal amplitude can

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Figure 1.2: Continuous and discrete signal in time domain.

Figure 1.3: Analog and digital signals.

take on an infinite number of values. A digital signal can take only a finite
number of values at time t. The terms continuous time and discrete time
define the nature of signal along the time (horizontal axis). The terms digital
and analog define the nature of the signal amplitude (vertical axis). Fig 1.3
shows examples of various types of signals.

1.2.3 Periodic and Non-periodic

A signal x(t) is called periodic in time if there exist a constant T o > 0 such
that :
x(t) = x(t + kT0 ) k ∈ R+ (1.1)
where t denotes time. The resultant value T0 that satisfies this condition is
called the periodic of x(t).The period T0 defines the duration of one complete
cycle of x(t) or it is called fundamental period. A signal for which there is no
value of that satisfies equation 1.1 is called non-periodic, or aperiodic signal.

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Figure 1.4: Periodic and non-periodic (aperiodic) signals.

1.2.4 Energy and Power Signals

A signal x(t) with finite energy, is energy Ex signal, and a signal with finite
power Px is a power signal. Where the energy and the power of x(t) are:

Z ∞
x(t)2 dt
 
Ex = (1.2)
−∞
Z T /2
x(t)2 dt
 
Px = lim (1.3)
T →∞ −T /2

A signal is energy-type if:

Ex < ∞ (1.4)
and is power-type if :
0 < Px < ∞ (1.5)
A signal cannot be both power and energy type at the same time; because
for energy type signals Px = 0 and for power type signals Ex = ∞. A
signal can be neither energy type nor power type, for example ramp signals.
Another note is, every periodic signal is a power signal. In real life all the
signal generated in the lab are energy signal because power signal required
infinite interval. 2
In fact, power is v(t) × i(t) , i(t)2 × R or v(t)
R
. Where, the voltage v(t)
represents the x(t) amplitude.

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Figure 1.5: Deterministic and random signals.

1.2.5 Deterministic and probabilistic signal

Deterministic signals are those signals whose values are completely specified
for any given time. Thus, a deterministic signal can be modeled by a known
function of time t.
Random signals are also called non deterministic signals are those signals
that take random values at any given time and must be characterized sta-
tistically. Characterized statistically is mean first we make the experiment,
then we collect the data (output event), finally we name the data to suitable
function named as random variable. Deterministic signals can be described
by functions in the usual mathematical sense with time t as the independent
variable. In contrast to a deterministic signal, a random signal always has
some element of uncertainty associated with it, and hence it is not possible
to determine its value with certainty at any given point in time.

1.3 Communication Channels

As indicated in our preceding discussion, the communication channel pro-
vides the connection between the transmitter and the receiver. The physical
channel may be a pair of wires that carry the electrical signal, or an optical
fiber that carries the information on a modulated light beam, or an under-
water ocean channel in which the information is transmitted acoustically, or
free space over which the information-bearing signal is radiated by use of an

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antenna. Other media that can be characterized as communication channels

are data storage media, such as magnetic tape, magnetic disks, and optical
disks.
One common problem in signal transmission through any channel is addi-
tive noise. In general, additive noise is generated internally by components,
such as resistors and solid-state devices, used to implement system. This
type of noise is usually called thermal noise. Other sources of noise and in-
terference may arise externally to the system, such as interference from other
users of the channel.
When such noise and interference occupy the same frequency band as the
desired signal, the effect can be minimized by proper design of the trans-
mitted signal and the demodulator at the receiver. Other types of signal
degradation may be encountered in transmission over the channel, such as
signal attenuation, amplitude and phase distortion, as well as multi-path
distortion [1].
The effects of noise may be minimized by increasing the power in the
transmitted signal. However, equipment and other practical constraints limit
the power level in the transmitted signal. Another basic limitation is the
available channel bandwidth. A bandwidth constraint is usually due to the
physical limitations of the medium and the electronic components used to
implement the transmitter and the receiver. These two limitations result
in constraining the amount of data that can be transmitted reliably over
any communication channel. The characteristics of several communication
channels is described in the following subsections[1].

1.3.1 Wireline Channels

The telephone network makes extensive use of wirelines for voice signal trans-
mission, as well as data and video transmission. Twisted-pair wirelines and
coaxial cable are basically guided electromagnetic channels that provide rel-
atively modest bandwidths. Telephone wire generally used to connect a cus-
tomer to a central office has a bandwidth of several hundred kilohertz (kHz).
On the other hand, coaxial cable has a usable bandwidth of several megahertz
(MHz). Figure 1.6 illustrates the frequency range of guided electromagnetic
channels, which include waveguides and optical fibers. Signals transmitted
through such channels are distorted in both amplitude and phase, and they
are further corrupted by additive noise. Twisted-pair wireline channels are
also prone to crosstalk interference from physically adjacent channels.

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Figure 1.6: Frequency range for different channels starting from wireline up
to fiber optic channels.

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1.3.2 Fiber Optic Channels

Optical fibers offer the communication system designer a channel bandwidth
that is several orders of magnitude larger than coaxial cable channels. During
the past decade, researchers have developed optical fiber cables, which have a
relatively low signal attenuation, and highly reliable photonic devices, which
improve signal generation and signal detection. These technological advances
have resulted in a rapid deployment of fiber optic channels both in domestic
telecommunication systems as well as for transatlantic and transpacific com-
munications. With the large bandwidth available on fiber optic channels, it
is possible for the telephone companies to offer subscribers a wide array of
telecommunication services, including voice, data, facsimile, and video.
The transmitter or modulator in a fiber-optic communication system is
a light source, either a light-emitting diode (LED) or a laser. Information
is transmitted by varying (modulating) the intensity of the light source with
the message signal. The light propagates through the fiber as a light wave
and is amplified periodically (in the case of digital transmission, it is detected
and regenerated by repeaters) along the transmission path to compensate for
signal attenuation. At the receiver, the light intensity is detected by a pho-
todiode, whose output is an electrical signal that varies in direct proportion
to the power of the light impinging on the photodiode

1.3.3 Wireless Electromagnetic Channels

In radio communication systems, electromagnetic energy is coupled to the
propagation medium by an antenna, which serves as the radiator. The phys-
ical size and the configuration of the antenna depend primarily on the fre-
quency of operation. To obtain efficient radiation of electromagnetic energy,
the antenna must be longer than 1/10 of the wavelength. Consequently, a
radio station transmitting in the AM frequency band, say, at 1 MHz (cor-
responding to a wavelength of λ = c/fc = 300m) requires an antenna of at
least 30 meters. Figure 1.7 illustrates the various frequency bands of the
electromagnetic spectrum.
The mode of propagation of electromagnetic waves in the atmosphere and
in free space may be subdivided into three categories, namely, ground-wave
propagation, sky-wave propagation, and line-of-sight (LOS) propagation. In
the very low frequency (VLF) and extremely low frequency bands where
the wavelengths exceed 10 kilometers, the earth and the ionosphere act as a
waveguide for electromagnetic wave propagation.

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Figure 1.7: Frequency range for wireless electromagnetic channels (wireless)

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In these frequency ranges, communication signals practically propagate

around the globe. For this reason, these frequency bands are primarily used
to provide navigational aids from shore to ships around the world. The
channel bandwidths available in these frequency bands are relatively small
(usually 1%-10% of the center frequency); hence, the information that is
transmitted through these channels is relatively of slow speed and generally
confined to digital transmission.
A dominant type of noise at these frequencies is generated from thunder-
storm activity around the globe, especially in tropical regions. Interference
results from the many users of these frequency bands.
Ground-wave propagation, illustrated in Figure 1.8b, is the dominant
mode of propagation for frequencies in the medium frequency (MF) band
(0.3-3 MHz). This is the frequency band used for AM broadcasting and
maritime radio broadcasting. In AM broadcast, ground-wave propagation
limits the range of even the most powerful radio stations to about 100 miles.
Atmospheric noise, man-made noise, and thermal noise from electronic com-
ponents at the receiver are dominant disturbances for signal transmission at
MF.
Sky-wave propagation, as illustrated in Figure 1.8c, results from trans-
mitted signals being reflected (bent or refracted) from the ionosphere, which
consists of several layers of charged particles ranging in altitude from 30 to
250 miles above the surface of the earth. During the daytime hours, the
heating of the lower atmosphere by the sun causes the formation of the lower
layers at altitudes below 75 miles. These lower layers, especially the D-layer,
absorb frequencies below 2 MHz; thus, they severely limit sky-wave prop-
agation of AM radio broadcast. However, during the nighttime hours, the
electron density in the lower layers of the ionosphere drops sharply and the
frequency absorption that occurs during the day is significantly reduced. As
a consequence, powerful AM radio broadcast stations can propagate over
large distances via sky wave over the F-layer of the ionosphere, which ranges
from 90 miles to 250 miles above the surface of the earth.
A common problem with electromagnetic wave propagation via sky wave
in the high frequency (HF) range is signal multipath. Signal multipath occurs
when the transmitted signal arrives at the receiver via multiple propagation
paths at different delays. Signal multipath generally results in intersym-
bol interference in a digital communication system. Moreover, the signal
components arriving via different propagation paths may add destructively,
resulting in a phenomenon called signal fading. Most people have experi-
enced fading phenomenon when listening to a distant radio station at night,
when sky wave is the dominant propagation mode.
Additive noise at HF is a combination of atmospheric noise and thermal

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(a) Line of sight (LOS)

(b) Ground-wave

(c) Sky-wave

Figure 1.8: Electromagnetic waveguide (wireless) possible propagation

schemes.

noise. Sky-wave ionospheric propagation ceases to exist at frequencies above

approximately 30 MHz, which is the end of the HF band. However, it is
possible to have ionospheric scatter propagation at frequencies in the range
of 30-60 MHz; this is a result of signal scattering from the lower ionosphere.
It is also possible to communicate over distances of several hundred miles
using tropospheric scattering at frequencies in the range of 40-300 MHz.
Troposcatter results from signal scattering due to particles in the atmo-
sphere at altitudes of 10 miles or less. Generally, ionospheric scatter and
tropospheric scatter involve large signal propagation losses and require a
large amount of transmitter power and relatively large antennas. Frequen-
cies above 30 MHz propagate through the ionosphere with relatively little
loss and make satellite and extraterrestrial communications possible. Hence,
at frequencies in the VHF band and higher, the dominant mode of electro-
magnetic propagation is LOS propagation. For terrestrial communication
systems, this means that the transmitter and receiver antennas must be in
direct LOS with relatively little or no obstruction. For this reason, television
stations transmitting in the very high frequency (VHF) and ultra high fre-
quency (UHF) bands mount their antennas on high towers in order to achieve
a broad coverage area.

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1.3.4 Underwater Acoustic Channels

Over the past few decades, ocean exploration activity has been steadily in-
creasing. Coupled with this increase in ocean exploration is the need to
transmit data, which is collected by sensors placed underwater, to the sur-
face of the ocean. From there, it is possible to relay the data via a satellite
to a data collection center.
Electromagnetic waves do not propagate over long distances underwater,
except at extremely low frequencies. However, the transmission of signals
at such low frequencies is prohibitively expensive because of the large and
powerful transmitters required. The attenuation of electromagnetic waves in
water can be expressed in terms of the skin depth, which is the distance a
signal is attenuated by 1/e. For seawater, the skin depth is:
p
δ = 250/ f (1.6)

where f is expressed in Hertz and δ is in meters. For example, at 10 kHz,

the skin depth is 2.5 meters, which means the electromagnetic signal will be
weakened within couple of meters in water!.
In contrast, acoustic signals propagate over distances of tens and even
hundreds of kilometers. A shallow-water acoustic channel is characterized
as a multipath channel due to signal reflections from the surface and the
bottom of the sea. Due to wave motion, the signal multipath components
undergo time-varying propagation delays that result in signal fading. In
addition, there is frequency-dependent attenuation, which is approximately
proportional to the square of the signal frequency. Ambient ocean acoustic
noise is caused by shrimp, fish, and various mammals. Additionally, man-
made acoustic noise exists near harbors. In spite of this hostile environment,
it is possible to design and implement efficient and highly reliable underwater
acoustic communication systems for transmitting digital signals over large
distances.

1.3.5 Storage Channels

Information storage and retrieval systems constitute a significant part of our
data-handling activities on a daily basis. Magnetic tape (including digital
audio tape and video tape), magnetic disks (used for storing large amounts
of computer data), and optical disks (or compact disk (CD) which is used
for computer data storage, music, and video) are examples of data storage
systems that can be characterized as communication channels.
The process of storing data on a magnetic tape, magnetic disk, or optical
disk (CD or DVD) is equivalent to transmitting a signal over a telephone or

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a radio channel. The read-back process and the signal processing used to
recover the stored information is equivalent to the functions performed by a
telephone receiver or radio communication system to recover the transmitted
information
Additive noise generated by the electronic components and interference
from adjacent tracks is generally present in the readback signal of a storage
system. The amount of data that can be stored is generally limited by the
size of the disk or tape and the density (number of bits stored per square
inch) that can be achieved by the write/read electronic systems and heads.
The speed at which data can be written on a disk or tape and the speed at
which it can be read back is also limited by the associated mechanical and
electrical subsystems that constitute an information storage system. Channel
coding and modulation are essential components of a well-designed digital
magnetic or optical storage system. In the readback process, the signal is
demodulated and the added redundancy introduced by the channel encoder
is used to correct errors in the readback signal.

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Figure 1.9: Simplex vs. Duplex Communication scheme.

1.4 Multiple Access to the Channel

The communication is either one-way transmission, called simplex, or two-
way transmission or duplex. Multiple access to the channel resources varied
based on the system requirements. Duplex multiple access could be half-
duplex or full-duplex communication as shown in figure 1.9.
In order to design seamless communication between two nodes, transmission-
reception should be implemented without delay. Full-duplex communication
is important in audio/video communication systems, as human should inter-
act without any noticed barrier.
Duplex is achieved by either divide time, frequency or code between the
communicating parties. Hence, multiple access could be achieved by:

1.4.1 Time Division Multiple Access (TDMA)

in TDMA, time is divided among the communicating parties while frequency
is shared. Data is transmitted in time slots, which make up frames. Guard
time caters for time inaccuracies due to clock instability, Delay Spread, Prop-
agation delay, Pulse tails.
TDMA has the following characteristics:

1. Forward and reverse channels are transmitted on the same frequency,

but not the same time.

2. Although half-duplex in theory (two-way transmission, single direction

at any one time), in practice it is perceived as full duplex due to the

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Figure 1.10: Time Division Multiple Access (TDMA)

very high switching rate.

3. Better use of spectral resources

4. Expensive duplexers are not required in half-duplex mode

5. RF switch is used instead for the antenna to either transmit or receive

Advantages of TDMA are:
• No need for duplexers.

• Flexible bit rate (multiple and sub-multiple)

• Frame by frame signal strength/bit error monitoring, to enable hand-

offs.

• Bandwidth efficient: No guard bands between channels

Disadvantages of TDMA are:
• Guard and synchronisation require large overheads.

• High data rate transmission requires high clock devices.

1.4.2 Frequency Division Multiple Access (FDMA)

In FDMA. frequency is divided among the communicating parties while time
is shared.
Features of FDMA are:
1. Simple hardware with Band Pass Filters to isolate users.

2. No timing or synchronization is necessary

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Figure 1.11: Division Multiple Access (FDMA)

3. Narrow bands imply flat fading and hence equalizers are not needed.

Problems of FDMA are:

1. Very good cut off characteristics required for Band pass filters

2. Side lobe cross talk (Adjacent Channel Interference) produced by non-

linearities.

3. Not suitable for variable rate transmission

4. Limited capacity improvement since channel sits idle when not in use

5. Guard bands decrease capacity

6. Hardware cost (Duplexers)

1.4.3 Hybrid TDMA-FDMA

• Both time and frequency are divided among the communicating parties
in this scheme,none is shared.

• Transmission occurs in bursts, whereby each user is allocated part of

the bandwidth for the duration of a burst

• Currently used in North American Digital Cellular USDC, Global Sys-

tem for mobile communications GSM, and personal digital cellular PDC

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Figure 1.12: Hybrid TDMA-FDMA

1.4.4 Code Division Multiple Access (CDMA)

Both time and frequency are shared among the communicating parties. How-
ever, they have to be separated by having specific code for each one of them.
CDMA has the following properties:
1. Recent concept for cellular systems
2. Old Technology in military applications:
3. Anti-jamming as spread spectrum is difficult to jam
4. Ranging by measuring the distance from signal transmission time
5. Secure communications since spread spectrum is hard to detect
Direct Sequence CDMA (DS-CDMA) could be directly applied, so that
the allocated frequency band is shared between multiple users at the same
time. Users are separated by modulating their information by unique high bit
rate code sequences. Codes are essentially uncorrelated (orthogonal codes).
For security and jamming prevention, Frequency-Hopping CDMA (FH-
CDMA) was introduced. So that:
• The spectrum is divided into a number of frequency bands.
• A channel is assigned upon demand by assigning each user a unique
frequency-hopping sequence such that each segment of information is
transmitted on a different frequency to the previous segment.
• No two users transmit using the same band at the same time.
• The capacity of the system is a function of the number of unique hop-
ping sequences that can be used.

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Figure 1.13: Code Division Multiple Access (CDMA)

CHAPTER 1. DATA COMMUNICATION PRINCIPLES 20

Chapter 2

Discrete Pulse Modulation

2.1 Introduction
Information could be transmitted in analog, discrete or digital form. Discrete
signal transmission was implemented long time ago before digital signal were
experienced. The discretization of the analog signal led to the digital com-
munication approaches later.
In this chapter, generating the discrete and binary signals are introduced
by introducing sampling theory.

2.2 Sampling Theory

In order to transmit the analog information using digital system, it has to
be converted into a digital form. Analog-to-digital conversion (ADC) is the
most popular process that is used in almost all digital processing. The ADC
samples the analog signal, hold its value for short period of time, and then
quantize the sampled value to certain discrete voltage levels. This process
is shortened in the term sample and hold (S/H), and a basic electrical
circuit that can implement this process is shown in Figure 2.1.

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Ts

Analog signal g(t) C Sampled signal gδ (t)

Figure 2.1: Sample and hold block diagram

Figure 2.2: Sampling g(t) in time domain.

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The samples are taken for each Ts span or interval of time, where Ts =
1/fs and fs is the sampling rate. Suppose the energy signal x(t) has a
bandwidth of B Hz, then its sampled version in time domain xδ (t) is:
∞
X
xδ (t) = x(t)δ(t − nTs ) (2.1)
n=−∞

where T s = 1/fs is the sampling interval, i.e. the time you wait after taking
a sample before you pick the next one.
Note that equation 2.1 is simply the sum of the analog signal multiplied
by an impulse train δ(t), where the impulses are separated by the sampling
interval Ts .
It is important to understand the spectrum of the sampled signal in the
frequency domain, because it will facilitate the communication system design.
Therefore, Fourier transform is used to obtain Xδ (f ), as the signal in time
domain could be expressed in frequency domain using the following Fourier
transform relationship:
FT
xδ (t) ←−−−−→ Xδ (f ) (2.2)
Hence: ∞
 X 
Xδ (f ) = FT xδ (t) δ(t − nTs ) (2.3)
n=−∞

which is simply:
 
FT Product of x(t) and impulse train (2.4)

One of the Fourier transform properties stated that the multiplication in

time domain is a convolution of the spectrum of each signal in the frequency
domain. Therefore, the relationship in formula 2.4 will be:

Xδ(f ) = X(f ) ∗ FT {δ(t − nTs )} (2.5)

∞
1 X n
= X(f ) ∗ δ(f − ) (2.6)
Ts n=−∞ Ts
∞
X
= X(f ) ∗ fs δ(f − nfs ) (2.7)
n=−∞

Using the linearity of convolution, the last equation will be:

CHAPTER 2. DISCRETE PULSE MODULATION 23

Data Communication CIE3304 Azhar Abdulaziz

Figure 2.3: The spectrum of the analog signal compared to the spectrum of
the sampled version of it. Here A is the amplitude, T1s is the sampling
frequency fs and 2W is the signal bandwidth.

∞
X n
Xδ (f ) = fs X(f ) ∗ δ(f − ) (2.8)
n=−∞
Ts

The right-hand-side (RHS) of the final equation could be interpreted as

follows:
Xδ (f ) consists of replicas of the analog signal spectrum X(f ),
where each one of them is shifted by sampling frequency fs as
shown in 2.3.
Mathematically, the spectrum of the sampled signal is:
∞
X
Xδ (f ) = fs X(f − nfs ) (2.9)
n=−∞

2.3 Nyquest Theorem and Aliasing

In order to retrieve the original analog signal from its samples, the sample
rate fs has to be equal or higher than the maximum frequency of it. This
rule is commonly known as Nyquest theorem.
A band limited signal of finite energy which has no frequency higher than
fmax Hertz may be completely recovered into its analog form if:

fs > 2fmax (2.10)

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.4: (a) Spectrum of a signal. (b) Spectrum of an undersampled

version of the signal, exhibiting the aliasing phenomenon.

The digital to analog conversion (DAC) is required for some applications

like speech communication. If the condition in equation 2.10 is met, then the
DAC will produce the analog signal exactly as it was before sampling.
On the other hand, if the analog signal is sampled with a rate that is
lower than fmax , then the DAC will retrieve a different signal. In this case,
the DAC output is called alias of the original signal. The aliasing problem
could be solved by making sure that the analog signal is sampled in rate that
is higher than it maximum frequency.
In many cases, the analog signal might have undesired frequency com-
ponents that are higher than the designer thought. If removing those unex-
pected frequency components do not affect the message that will transmitted,
then they should be filtered out using what is known as Anti-Aliasing Filter
(AAF). In most cases, AAF is a LPF; as most of the messages are low-pass
signals. However, the AAF could be BPF in some cases, like when the ADC
is required to deal with specific tone in music application.

Example 2.1 Determine the sampling rate and interval of the signal:

x(t) = 2cos(4000πt)cos(1000πt)

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.5: Spectrum of the message signal.

Solution:
x(t) = 2cos(4000πt)cos(1000πt)
1 

=2 cos((4000 − 1000)πt) + cos((4000 + 1000)πt)
2
= cos(3000πt) + cos(5000πt)
= cos(2πf1 ) + cos(2πf2 )

Comparing the frequencies, f1 = 1500 and f2 = 2500. Hence, the highest

frequency of the signal x(t) is:

fmax = f2 = 2500

Hence, sampling rate is:

fs = 2fmax = 2f2 = 5000

Example 2.2 Figure 2.5 shows the spectrum of a message signal, which was
sampled by 1.5fmax , where fmax = 1 Hz is the maximum signal frequency.
1. Sketch the spectrum of the sampled version of the signal.

2. When the sampled signal is received, it has to go through a low-pass

filter (LPF). If the cut-off frequency of the LPF is 1 Hz, sketch the
spectrum of the output signal from this filter.
Solution:
1. When x(t) is sampled, its spectrum will be:

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.6: (a) Spectrum of sampled signal. (b) Transfer function of LPF.
(c) Filtered signal.

∞
X
Xδ (f ) = fs X(f − nfs )
n=−∞

Here fs = 1.5 ;max , hence it will be:

∞
X
Xδ (f ) = 1.5fmax X(f − 1.5nfmax )
n=−∞

With Fmax = 1 Hz, the above equation will be:

∞
X
Xδ (f ) = 1.5 X(f − 1.5n)
n=−∞

Figure 2.6(a) is the plot of the above equation.

2. As the signal is sampled with fs = 1.5fmax an aliasing effect occurs
as shown in 2.6(a). When the sampled signal is passed through a LPF of
BW = fmax [see figure 2.6(b)], the output will be as shown in 2.6(c). Note
that the signal in 2.5 and 2.6(c) are not the same because of the aliasing
effect phenomenon.

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.7: Pulse Amplitude Modulation (PAM).

2.4 Pulse Amplitude Modulation

In pulse-amplitude modulation (PAM), the amplitudes of regularly spaced
pulses are varied in proportion to the corresponding sample values of a con-
tinuous message signal; the pulses can be of a rectangular form or some other
appropriate shape. Pulse-amplitude modulation as defined here is somewhat
similar to natural sampling, where the message signal is multiplied by a pe-
riodic train of rectangular pulses. However, in natural sampling, the top of
each modulated pulse is allowed to vary with the message signal, whereas in
PAM it is retained flat.
The waveform of a PAM signal is illustrated in Figure 2.7. The dashed
curve in this figure depicts the waveform of the message signal m(t), and
the sequence of amplitude-modulated rectangular pulses shown as solid lines
represents the corresponding PAM signal s(t). There are two operations
involved in the generation of the PAM signal:

1. Sampling the signal of the message m(t) every sTs seconds, where the
sampling rate fs = 1/T s is selected based on sampling theorem.

2. Increasing the duration length of each sample, so that it occupies some

finite value T .

Those operations are the same function of the sample-and-hold (S/H)

circuit discussed earlier in section 2.2. The reason for increasing the duration
length of a sample is to minimize the required channel bandwidth (BW), as
BW is inversely proportional to pulse duration T .

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Data Communication CIE3304 Azhar Abdulaziz

If the sequence of pulses in PAM in figure 2.7 is denoted as s(t), then

PAM signal could be described as:
∞
X
s(t) = m(nTs )h(t − nTs ) (2.11)
n=−∞

where Ts is the sampling interval, m(nTs ) is the sampled version of m(t)

at time t = nTs and h(t) is the standard pulse train. The latter is has a
rectangular shape, duration T and repeated at each sampling interval (this
is why it has the argument t − nTs ). Therefore:


t − T2
  1, 0 < t < T

1
h(t) = rect = 2
, t = 0, t = T (2.12)
T 
 0, otherwise

PAM Receiver At the receiver, the PAM is sampled using the same sam-
pling rate fs = T1s , to make mδ (t) which is:
∞
X
mδ (t) = m(nTs )δ(t − nTs ) (2.13)
n=−∞

where δ(t − nTs ) is the time-shifted delta function (comb function).

Convolve mδ (t) with pulse in equation 2.12 to get:
Z ∞
mδ (t) ∗ h(t) = mδ (τ )h(t − τ )dτ
−∞
Z ∞ ∞
X
= m(nTs )δ(t − nTs )h(t − τ )dτ
−∞ n=−∞
Z ∞ X ∞ (2.14)
= m(nTs )δ(t − nTs )h(t − τ )dτ
−∞ n=−∞
∞
X Z ∞
= m(nTs ) δ(t − nTs )h(t − τ )dτ
n=−∞ −∞

As delta function has only values at t − nTs , multiplying it by h(t − τ )

produces h(t − nTs ). Therefore the convolution will be:
∞
X
mδ (t) ∗ h(t) = m(nTs )h(t − nTs ) (2.15)
n=−∞

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Data Communication CIE3304 Azhar Abdulaziz

which equals the PAM transmitted signal in equation 2.11. In words, receiver
can detect PAM if it samples the received signal and convolve it with the
carrier rectangular pulses, that have the same frequency at the transmitter.
NOTE: Multiplying the sampled received signal by the carrier is a common
technique in demodulation.

PAM Receiver Design In fact, PAM signal s(t) could be obtained at the
receiver by convolving the received signal with the rectangular pulse train as
in equation 2.15, which is:
∞
X
s(t) = mδ (t) ∗ h(t) = m(nTs )h(t − nTs ) (2.16)
n=−∞

Now, take Fourier transform of the above equation, to convert convolution

to multiplication as the latter could be easily implemented using electronic
devices, which will lead to:

S(f ) = Mδ (f )H(f ) (2.17)

where Mδ (f ) is the Fourier transform of the sampled received PAM, which

is related to the continuous PAM as follows (using sampling theory 2.9):
∞
X
Mδ (f ) = fs M (f − kfs ) (2.18)
k=−∞

where fs is the sampling rate. Therefore:

∞
X
S(f ) = fs M (f − kfs )H(f ) (2.19)
k=−∞

In order to PAM signal back to its analog origin, all extra parts in the
previous equation should be removed to get pure S(f ). So, electronic devices
are used in PAM receiver to do the following:

1. Reconstruction: Pass the signal in 2.19 through a reconstruction

filter, which an LPF which has a characteristics shown in figure 2.8.

2. Equalization: Because flat-top sampling is the aspect of PAM,

amplitude distortion and a delay of T /2 is expected at the receiver
(aperture effect). Hence, the reconstructed signal is step 1 is passed
through an equalizer, which has to remove the flat-top sampling effect

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.8: Reconstruction filter ideal amplitude response

Figure 2.9: PAM receiver: A system that recovers message m(t) from PAM
signal.

by having a magnitude response that is reciprocal of the rectangular

samples H(f ). The PAM equalizer response is:
1 1 2π
= = (2.20)
H(f ) T sinc(f T ) sin(2πf )

Using those logical steps, received PAM signal can be back to the original
message m(t). The resultant PAM receiver could be as shown in 2.9.

2.5 Pulse Width Modulation

In PAM, the amplitude of the pulse is the variable parameter that represents
the amplitude of the message. The width (duration) of the pulse could also
be used to indicate the message voltage level for modulation. In pulse-width
modulation (PWM), the message signal samples are used to vary the width
of the pulse train signal. This form of modulation is also referred to as
pulse-duration modulation (PDM) or pulse-length modulation (PLM). The
modulating message signal can vary the time of occurrence of the rising edge,
the falling edge, or both edges of the pulse. Figure 2.10 depicts an example
of PWM where the modulating signal is sinusoidal.
In PWM, the carrier pulse train has a variable duration so that the mod-

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.10: Pulse-width modulation (a) Modulating message signal (b)

Carrier pulses (c) Modulated PWM signal.

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Data Communication CIE3304 Azhar Abdulaziz

ulated signal s(t) will be :

X
s(t) = ∞ḣ(t − nTs ) (2.21)
n=−∞

where ḣ is the variable-width PWM signal, which is:

t − T2
 
ḣ = rect wherea ∝ m(t) (2.22)
aT

where m(t) is the modulating signal (message).

Therefore, the sampling rate relies on the maximum expected input volt-
age, of the message signal, and has the following relationship with it:
1
Ts ∝ (2.23)
amax T
The maximum expected voltage in the message signal will degrade the
sampling rate. If the expected maximum voltage increases, the PWM sig-
nal width increases as well, and hence the transmitter power will also be
increased, which is the main advantage of the PWM.
The PWM receiver is simply a counter and voltage comparator. The
counter counts the clock pulses when the PWM is high and decode the mes-
sage as a proportion of those pulses. Low number of pulses on high PWM
signal means low voltage and vice versa.

2.6 Time Division Multiplexing

The sampling theorem provides the basis for transmitting the information
contained in a band-limited message signal m(t) as a sequence of samples
of taken uniformly at a rate that is usually slightly higher than the Nyquist
rate. An important feature of the sampling process is a conservation of time.
That is, the transmission of the message samples engages the communication
channel for only a fraction of the sampling interval on a periodic basis, and in
this way some of the time interval between adjacent samples is cleared for use
by other independent message sources on a time-shared basis. We thereby
obtain a time-division multiplex (TDM) system, which enables the joint uti-
lization of a common communication channel by a plurality of independent
message sources without mutual interference among them.
Suppose a TDM communication system shown in figure 2.11 has N mes-
sages to be transmitted. According to Nyquest theorem, the commutator
(or sampling switch) has to take a sample from each message at a rate offs .

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.11: Time division multiplexing (TDM) system.

The receiver will use a the same rate to decode the time division multiplexed
messages and separate them eventually. If the highest frequency among those
messages is W , then the sampling frequency fs of the commutator is:

fs ≥ 2W (2.24)
Hence the time space between successive samples of any message (signal)
is:
1
Ts = (2.25)
fs
1
∴ Ts ≤ (2.26)
2W
The sample time interval Ts contains one sample from each input message,
and each time interval is called a frame. For N messages, in each frame there
will be one sample of the N messages, which means one frame of interval Ts
will have N samples. Therefore:
Ts
Spacing between two samples = (2.27)
N
The signaling rate, which is the number of samples per second at the
TDM output, channel and the receiver input, will be:
1
TDM Signaling Rate =
Spacing between two samples
1
=
Ts /N (2.28)
N
=
Ts
= N fs

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Data Communication CIE3304 Azhar Abdulaziz

As it was stated before, the sampling frequency should be higher than

twice the maximum frequency, or fs ≥ 2W , therefore:

Signalling Rate ≥ 2N W (2.29)

2.7 Quantization
The sampled version of the analog signal contains voltage levels which has
to be scaled to predefined quantities in quantization phase. Each quantity
is given a certain binary code in the encoding process. After those two
processes finished, a binary sequence will represent the digital form of the
analog input. This sampling, quantization and encoding are the main parts
of any ADC.
In the encoding process, a sequence of bits are assigned to different quan-
tization values. Since there are a total of N = 2v quantization levels, v bits
are sufficient for the encoding process. In this way, we have v bits corre-
sponding to each sample; since the sampling rate is fs samples/sec, we will
have a bit rate of R = v × fs bits/sec.

2.7.1 Uniform Quantization

Basically, the quantizer will assign specific values for discrete samples of the
analog signal. The quantized levels are discrete values step size δ is the
difference between each two consecutive values. There are many types of
quantizer, while in this course only midriser type is covered.
For midriser, when the quantizer output is rising will be zero when the
input is zero, and it is up-rise from −δ/2 to δ/2 directly as shown in figure
2.12,. this is why it is called midriser quantizer.
It is important to understand that the x − axis of a continuous signal is
the time domain t, and it will be replaced by nTs in the discrete domain after
sampling. On the other hand, the ADC designer has to choose the number
of bits v to represent the digital form of the signal.
Let x(nTs ) is a continuous signal that its peak voltage ranges from −xmax
to xmax . The quantizer will map those voltages to 0 q 0 levels on the vertical
axis as shown in figure 2.12. Hence:

Total amplitude range = xmax − (−xmax )

(2.30)
= 2xmax

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.12: Transfer function for eight level midriser uniform quantizer
and its error ε function.

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Data Communication CIE3304 Azhar Abdulaziz

The range is divided into 0 q 0 levels by the quantizer, and the step size δ
is:

xmax − (−xmax )
δ=
q
(2.31)
2xmax
=
q
If samples xq (t) represent the quantization of the signal x(t), then the
quantization error is (t) is the difference between the analog value and the
quantized value at the same time t.
(t) = x(t) − xq (t) (2.32)
If the amplitude of x(t) is normalized to minimum and maximum values
of 1, then:
2
δ= (For normalized signals) (2.33)
q
Now, the maximum error is:
δ
max = (2.34)
2

2.7.2 Non-uniform quantization

In certain applications, it is preferable to use a variable separation between
the quantization levels. For example, the range of voltages covered by voice
signals, from the peaks of loud talk to the weak passages of weak talk, is on
the order of 1000 to 1. By using a nonuniform quantizer with the feature that
the step size increases as the separation from the origin of the inputoutput
amplitude characteristic is increased, the large end-step of the quantizer can
take care of possible excursions of the voice signal into the large amplitude
ranges that occur relatively infrequently. In other words, the weak passages
that need more protection are favored at the expense of the loud passages.
In this way, a nearly uniform percentage precision is achieved throughout the
greater part of the amplitude range of the input signal, with the result that
fewer steps are needed than would be the case if a uniform quantizer were
used. The use of a nonuniform quantizer is equivalent to passing the message
signal through a compressor and then applying the compressed signal to a
uniform quantizer. A particular form of compression law that is used in
practice is the so called µ-law is defined by:
ln(1 + µ|m|)
|v| = (2.35)
ln(1 + µ)

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.13: Quantization compression (a) µ-law (b) A-law.

where ln is the natural logarithm; m and v are respectively the normalized

input and output voltages, and µ is a positive constant compression param-
eter. The practical values of µ are from 1-255, while µ = 0 is for uniform
quantization as shown in figure 2.13 (a), .
The µ-law compression is commonly used in the USA, Canada and Japan.
In Europe, another compression algorithm, which is called A-law is used. The
A-law is defined as follows:
( A|m|
1+logA
0 ≤ |m| < A1
|v| = 1+log(a|m|) 1 (2.36)
1+logA A
≤ |m| ≤ 1

For the A-law compression, A is the compression parameter an it is typ-

ically less than 100 (specifically in Europe they use 87.6). When A = 1 the
quantizer will work as a uniform one.

2.8 Encoding
In scalar quantization, a natural way of encoding is to assign the values of
0 to N to different quantization levels starting from the lowest level to the
highest level in order of increasing level value.
This type of encoding is called natural binary coding or NBC for short.
Another approach to coding is to encode the quantized levels in a way that
adjacent level differ only in one bit. This type of coding is called Gray coding.

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Data Communication CIE3304 Azhar Abdulaziz

Table 2.1: Encoding the quantized analog voltages in ADC

Quantization Level Level Order NBC Code Gray Code

x̂1 0 000 000
x̂2 1 001 010
x̂3 2 010 011
x̂4 3 011 001
x̂5 4 100 101
x̂6 5 101 100
x̂7 6 110 110
x̂8 7 111 111

Ts

Analog signal g(t) C Quantizer Encoder Digital

Figure 2.14: The ADC block diagram which should have sample and hold
followed by a quantizer and an encoder.

2.9 The Analog to Digital Converter

The analog to digital converter shown in figure 2.14 consists of S/H followed
by a quantizer and a digital encoder. The digital output which comes out of
the encoder has 0 v 0 lines, where each line is a bit. The bit rate r is the number
of bits produced for each second and depends on the sampling interval Ts of
the S/H circuit. In this case:
v
r= = v × fs (2.37)
Ts

Example 2.3 A message signal that varies between -8 and 8 volt is processed
by a specific ADC. The S/H had 0.001 seconds sampling interval. What is
the maximum quantization error and the bit rate of a 3 bits encoder?
Solution: As number of bits v = 3, the number of quantization levels is:
q = 2v = 23 = 8

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Data Communication CIE3304 Azhar Abdulaziz

Now, the step size δ is:

Vmax − (−Vmax )
δ=
q
8 − (−8) 16
= = =2 volts.
8 8
The maximum quantization error is:
δ
max = =1 volts.
2
and the ADC bit rate r will be:
v 3
r= = = 3000 bits/seconds.
Ts 0.001

2.10 Pulse Code Modulation

Pulse Code Modulation (PCM) is the most basic form of digital pulse modu-
lation. In pulse-code modulation (PCM), a message signal is represented by
a sequence of coded pulses, which is accomplished by representing the signal
in discrete form in both time and amplitude, which is simply an ADC.
The basic operations performed in the transmitter of a PCM system are
sampling, quantization, and encoding, as shown in figure 2.15(a), the low-
pass filter prior to sampling is included merely to prevent aliasing of the
message signal. The quantizing and encoding operations are usually per-
formed in the same circuit, which is called an analog-to-digital converter.
The basic operations in the receiver are regeneration of impaired signals,
decoding, and reconstruction of the train of quantized samples, as shown
in figure 2.15(c). Regeneration also occurs at intermediate points along the
transmission path as necessary, as indicated in figure 2.15(b).

2.10.1 PCM Signal to Noise Ratio (SNR)

The PCM noise mainly comes from the quantization process, which is the
error  discussed earlier in section 2.7.1. Besides there is thermal noise which
comes from the electronic devices used in implementation. The signal-to-
noise ratio by definition is:
Signal Power
SN R = (2.38)
Noise Power

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Data Communication CIE3304 Azhar Abdulaziz

Figure 2.15: The basic PCM system (a) transmitter (b) channel (c) receiver.

If that devices thermal noise is negligible, the PCM noise is only the
quantization error . This error is randomly changing which makes it non-
parametric (random) signal. Assume that the quantization error  is uni-
formly distributed random signal, its power PQN (which is the noise power in
PCM) will be:

Z δ/2
1
PQN = 2 d
δ −δ/2
1  3 δ/2
=
 3 −δ/2
1  δ 3 /8 − (−δ 3 /8)  (2.39)
=
δ 3
1  δ 3 /4 
=
δ 3
δ2
∴ PQN =
12
For PCM of a uniform quantizer with N − levels, the step size is:
2xmax
δ= (2.40)
N
Substitute in equation 2.39, the noise power, which is the quantization
noise, will be:

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Data Communication CIE3304 Azhar Abdulaziz

4x2max /N 2
P QN =
12 (2.41)
x2
∴ P QN = max2
3N
Now, the quantization level N = 2v , where v is number of bits of the
PCM encoder. Hence:
N 2 = 22v = (22 )v = 4v (2.42)
Substitute in equation 2.41, the noise power will be:
x2max
P QN = (2.43)
3 × 4v
In the other hand, the signal power Px is the statistical average (mean)
of the squared samples of the random signal x. This assumption is suitable
for deterministic and non-deterministic signals as the information is random
in most cases. So:

Px = x¯2 (2.44)
Assume that x̂ is the normalized version of the information signal samples
x, which is defined as:
x
x̂ = (2.45)
xmax
Then the power of the normalize signal is:
x¯2
x̂¯2 = 2 (2.46)
xmax
Hence, the signal power in PCM will be related to the normalized version of
as follows:
x̄2 = x̂¯2 × x2max (2.47)
Now, the PCM signal-to-noise ratio (SQN R), which is merely the quan-
tization SNR is:
x̂¯2
SQN R =
PQN
x̂¯2 × x2max (2.48)
= 2
xmax /3 × 4v
∴ SQN R = 3 × 4 × x̂¯
v 2

CHAPTER 2. DISCRETE PULSE MODULATION 42

Chapter 3

Digital Transmission and

Modulation

3.1 Introduction
In general, the physics of human beings communication are usually captured
as analog signals in the electrical domain, as for the speech and visual com-
munication patterns. However, converting the signals into digital form will
introduce many advantages over transmitting their analog version, which are:

1. Performance: Many noise sources that are imposed to the signal

through the communication channel are analog. Hence, the analog
communication is less immune to noise effect as compared to the dig-
ital system. Even by adding analog repeaters before the receiver, like
amplifiers, the noise will be amplified as the same amount as the sig-
nal, which means that the signal-to-noise ratio is hardly improved. In
contrast, the digital repeaters are reshaping the weak bits (which have
square shapes) easily and distinctively from the analog random-shape
channel noise.

2. Reliability: The error control ability of the digital system enables

the communication system to expect whether the received message has
errors, and hence needed to be retransmitted again. This ability was
not easy to be implemented on the analog communication.

3. Efficiency: The digital communication system is more flexible for

the trade-off between bandwidth and signal-to-noise ratio (SNR). This
is inherited from the nature of binary system which samples only lim-
ited number of levels.

43
Data Communication CIE3304 Azhar Abdulaziz

Figure 3.1: Baseband binary data transmission.

(Remember that digitized signals are discretized in amplitude with lim-

ited levels. )

4. Security : Transmitting digital signals of limited levels opened the

door for adding more security information. Again, this is not easy to
be done with analog systems as it will cost more bandwidth.

5. System integration: The use of digital communications makes it

possible to integrate digitized analog signals (i.e., voice and video sig-
nals) with digital computer data, which is not possible with analog
communications.

3.2 Digital Baseband Transmission

The term baseband is used to designate the band of frequencies representing
the original signal delivered by a source of information. The source of infor-
mation, for example, could be a computer that produces a stream of binary
data made up of the symbols 0 and 1. The task of a digital communication
system is to transport the data stream from the source to its destination over
a channel and do so in a reliable manner.
Binary data could be transmitted basically or ’as is’, without modulating
it with a carrier frequency. The baseband transmission of a digital signal
means putting its binary on the channel directly. The message is then trans-
mitted over an electromagnetic media, like cables, in different ways.
As a binary sequence of 1s and 0s is produced, a line code is needed for
electrical representation of that binary sequence. There are several line codes
which are:

1. Onoff signaling, in which symbol 1 is represented by transmitting a

pulse of constant amplitude for the duration of the symbol, and symbol
0 is represented by switching off the pulse, as in Figure 3.2(a).

CHAPTER 3. DIGITAL TRANSMISSION AND MODULATION 44

Data Communication CIE3304 Azhar Abdulaziz

2. Non-return-to-zero (NRZ) signaling, in which symbols 1 and 0 are rep-

resented by pulses of equal positive and negative amplitudes, as illus-
trated in Figure 3.2(b).

3. Return-to-zero (RZ) signaling, in which symbol 1 is represented by a

positive rectangular pulse of half-symbol width, and symbol 0 is repre-
sented by transmitting no pulse, as illustrated in Figure 3.2(c).

4. Bipolar return-to-zero (BRZ) signaling, which uses three amplitude

levels as indicated in Figure 3.2(d). Specifically, positive and negative
pulses of equal amplitude are used alternately for symbol 1, and no
pulse is always used for symbol 0. A useful property of BRZ signaling
is that the power spectrum of the transmitted signal has no DC com-
ponent and relatively insignificant low-frequency components for the
case when symbols 1 and 0 occur with equal probability.

5. Split-phase (Manchester code), which is illustrated in Figure 3.2(e). In

this method of signaling, symbol 1 is represented by a positive pulse
followed by a negative pulse, with both pulses being of equal amplitude
and half-symbol width. For symbol 0, the polarities of these two pulses
are reversed. The Manchester code suppresses the dc component and
has relatively insignificant low-frequency components, regardless of the
signal statistics.

6. Differential encoding, in which the information is encoded in terms of

signal transitions, as illustrated in Figure 3.2(f). In the example of the
binary PCM signal shown in the figure, a transition is used to designate
symbol 0, whereas no transition is used to designate symbol 1. It is
apparent that a differentially encoded signal may be inverted without
affecting its interpretation. The original binary information is recovered
by comparing the polarity of adjacent symbols to establish whether or
not a transition has occurred. Note that differential encoding requires
the use of a reference bit, as indicated in Figure 3.2(f).

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Data Communication CIE3304 Azhar Abdulaziz

Figure 3.2: Line codes. (a) Onoff signaling. (b) Nonreturn-to-zero

signaling. (c) Return-to- zero signaling. (d) Bipolar return-to-zero
signaling. (e) Split-phase or Manchester encoding. (f) Differential encoding.

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3.3 Intersymbol Interference (ISI)

The transmission of digital data over a physical communication channel is
limited by two unavoidable factors:

1. Intersymbol interference (ISI), which arises due to imperfections in the

frequency response of the channel.

2. Channel noise, which refers to unwanted electric signals that arise at the
channel out- put due to random and unpredictable physical phenomena.

As the name implies, intersymbol interference refers to interference caused

by the time response of the channel spilling over from one symbol into ad-
jacent symbols. It is a common problem in digital communication that is
required to be considered seriously.
In order to understand why and how ISI occurs in digital communication,
assume ak is the encoded signal of the digital information bk as shown in figure
3.1, and let it be encoded on the line as simple as:
(
+1 for Symbol 1
ak = (3.1)
−1 for Symbol 0
The transmitted signal will go through a transmit filter to produce a
proper pulse shape of Tb duration and has a shape defined as g(t) and G(f )
in time and frequency domains. Then, the transmitted signal that will go
through channel is:
∞
X
s(t) = ak g(t − kTb ) (3.2)
k=−∞

The signal s(t) is transmitted across a linear communication channel,

which is described in the time by the impulse response h(t), and frequency
domains transfer function H(f ) respectively. Ignoring the effect of additive
channel noise, the channel output may be expressed as:

x(t) = s(t) ∗ h(t) (3.3)

The convolution in the last equation happens between the transmitted
signal and the communication channel. The receiver will see x(t) signal,
which is the transmitted signal affected by the channel only (as we ignored
additive noise for now). The receiver tries to invert the effect of the channel
and tries to retrieve the signal s(t) as it was transmitted.

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The received signal is filtered at the receiver using receive-filter, which

has a impulse response q(t) and transfer function Q(f ).

y(t) = x(t) ∗ q(t) (3.4)

The filter output y(t) is next sampled synchronously with the generator of
clock pulses in the transmitter. synchronization is commonly established by
extracting a clock or timing signal from the receive-filter output. Finally, the
sequence of samples thus obtained is used to reconstruct the original binary
data stream by means of a decision-making device.
Specifically, the amplitude of each sample is compared to a threshold. If
the threshold is exceeded, a decision is made in favor of symbol 1, say. If
the threshold is not exceeded, a decision is made in favor of symbol 0. If the
sample amplitude equals the threshold exactly, the symbol may be chosen as
0 or 1 through the flip of a fair coin without affecting overall performance.
For the case when symbols 0 and 1 are equiprobable, it is reasonable to set
the threshold at the zero amplitude level.
Now, using equations 3.2 and 3.4, the received signal is:
∞
X
y(t) = ak p(t − kTb ) (3.5)
k=−∞

Using linearity, the received pulse shape p(t) is affected by all linear sys-
tem on the communication path. Hence:

p(t) = g(t) ∗ h(t) ∗ q(t) (3.6)

or in frequency domain:

P (f ) = G(f )H(f )Q(f ) (3.7)

The received filtered signal y(t) is sampled at the receiver (for bit duration
Tb to detect bits) as follows:
∞
X
y(iTb ) = ak p[(i − k)Tb ] i = 0, ±1, ±2, ±3, . . . (3.8)
k=−∞

where y(t) is sampled at each t = iTb . For short let:

yi = y(iTb )
pi = p(iTb )

then

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∞
X
yi = ak pi−k i = 0, ±1, ±2, ±3, . . . (3.9)
k=−∞

It is important to note the following:

1. The index i refers to the instant at which the receive-filter output is
sampled in the receiver.

2. The index k refers to a symbol in the data stream produced by the

source of information at the transmitted input.
,
Referring to figure 3.1, yi = y(iTb )) is the input to the decision-making
device. At the instant i = K, the sampled received at this instant signal is
the exact transmitted bit. Let
p
p0 = p(0) = Eb (3.10)
where Eb is the transmitted energy per bit (symbol). Then at this specific
instance, where i = k:
y(iTb ) = ak p(0) (3.11)
Remember that ak is the symbol on the line encoded information, which
is either +1 or -1, while p(.) is the pulse shape that appears on the received
part.
In equation 3.9, the first term represents the transmitted binary symbol
ai , while the second term involving the combined effect of all other trans-
mitted binary symbols before and after ai represents a residual phenomenon
called the intersymbol interference (ISI).

p ∞
X
yi = Eb ai + ak pi−k + n(t) i = 0, ±1, ±2, ±3, . . .
| {z } |{z}
T ransmitted k = −∞ N oise

Binary k 6= i
| {z }
Symbol ISI

(3.12)
In the absence of ISI and noise (optimum case):
p
yi = Eb ai for alli (3.13)

Even in the absence of noise, intersymbol interference is troublesome be-

cause it has the effect of introducing deviations (i.e., errors) between the

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data sequence reconstructed at the receiver output and the original data se-
quence applied to the transmitter input. Hence, unless corrective measures
are taken, intersymbol interference could impose a limit on the attainable
rate of data trans- mission across the channel, which is below the physical
capability of the channel.
The ISI appeared at the receiver sampler as a result of the channel af-
fecting the pulse shape p(t). The channel bandwidth depends on the channel
type and regulations, and it acts like a low pass filter. Therefore, you either
increase the channel BW, which is not always possible and it costs more
money, or play with the channel shape. The first option is limited by the
channel physical characteristics or regulations, while second option means
changing the transmitter and the receiver design.
Given a channel transfer function H(f ), the designer has to determine
(specify) the transmit-pulse spectrum G(f ) and the receive filter transfer
function Q(f ) to satisfy the following requirements:
√
1. ISI reduced to zero (yi = Eb ai ∀i)
2. Transmission bandwidth does not exceed the channel bandwidth.

Example 3.1 Determine the first three samples at the receiver for a base-
band transmission system if the transmitted signal ak = {1, 0, 1} and the line
code is NRZ signaling. Assume that the additive noise is negligible and only
those symbols are affecting each other.
Solution:

p ∞
X
yi = Eb ai + ak pi−k + n(t) i = 0, ±1, ±2, ±3, . . .
| {z } |{z}
T ransmitted k = −∞ N oise

Binary k 6= i
| {z }
Symbol ISI

The receiver takes the first sample at instant zero i = 0 and it suppose to
take a sample of symbol zero k = 0 beside the ISI and additive noise. The
ISI part is the contribution of all other symbols that interfere symbol k. This
is true for all other instances.
This Now, for i = 0
p ∞
X
y 0 = E b a0 + ak p−k
k = −∞
k 6= 0

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Because it is assumed that only the given three symbols are affecting each
others, then:

p 2
X
y0 = Eb a0 + ak p−k
k=1
p
= Eb a0 + a1 p−1 + a2 p−2
Substituting the symbols values ak with their NRZ line codes:
p
y0 = Eb − p−1 + p−2

Hence, for instance i = 1

p 2
X
y1 = E b a1 + ak p−k
k=0,k6=1
p
= Eb a1 + a0 p1−0 + a2 p1−2
p
= Eb a1 + a0 p1 + a2 p−1
p
= − Eb + p1 + p−1
And for instance i = 2
p 2
X
y 2 = E b a2 + ak p−k
k=0
p
= Eb a2 + a0 p2−0 + a1 p2−1
p
= E b a2 + a0 p 2 + a1 p 1
p
= E b + p 2 + p1
Figure 3.3 depicts the received signal and how the channel distortion is
affecting the communication for example 3.1.

3.4 Pulse Shapes

To minimize the transmission bandwidth, the pulse shape in baseband com-
munication system has to have zero values at frequencies greater than the
bandwidth B0 . As the receiver samples the signal after each bit duration iTb ,
the Brick-Wall bandwidth should be:
1
B0 = (3.14)
2Tb

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Figure 3.3: Three symbols example at baseband receiver

Figure 3.4: Pulse Shape of Nyquest channel, or the optimum BW channel.

(a) pulse in time (b) the spectrum of it.

which also means that the bit rate is twice the bandwidth. The spectrum
has an optimum brick-wall shape, and it is defined as:
( √
Eb
2B
−B0 ≤ f ≤ B0
p(f ) = 0 (3.15)
0 Otherwise

This condition will prevent aliasing of the sampling process at the receiver.
Therefore, the baseband communication system which uses pulse shape as in
figure 3.4(b) is called Nyquest Channel
Figure 3.4(a) is the time domain pulse shape which is defined as:
p
p(t) = Eb sinc(2B0 t)
√
Eb sin(2πB0 t) (3.16)
=
2πB0 t

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The Nyquist channel defined by the overall pulse spectrum is the opti-
mum solution for zero intersymbol interference at the minimum transmission
bandwidth possible in a noise-free environment. However, it is difficult to re-
alize in real life, because it is very difficult to approximate in practice because
of the abrupt transitions at ±B0 [2].
Besides, as the Nyquest channel pulse in time domain is ∝ 1/|t|, a small
timing error will make the shape decay slowly. Hence, there will be no error
margin at the receiver sampler as it has to take samples at exact time, which
in turn is not guaranteed practically.
To solve the Nyquest channel problems, which are related to the pulse
shape p(t), a Raised Cosine shape was introduced. Figure 3.5 shows the
raised cosine signal spectrum and time response.
To ensure physical realizability of the overall pulse spectrum P (f ), we
need a solution that differs from the Nyquist channel in one important
respect: the modified P (f ) decreases toward zero gradually rather than
abruptly. In more specific terms, P (f ) is proposed to consist of two por-
tions:
• Flat portion, which occupies the frequency band 0 ≤ |f | ≤ f1 for some
parameter f1 .
• Roll-off portion, which occupies the frequency band f1 < |f | < 2B0 −f1 .
The parameter f1 is adjustable under the designers control. The flat
portion may thus retain part of the ideal brick-wall solution. As for the roll-
off portion, it provides for the gradual decrease of toward zero. The following
formula represents the raised-cosine pulse spectrum in mathematical terms:
 √E
 √2B0
 b
, 0 ≤ |f | < f1
E
 π(|f |−f1 ) 
P (f ) = b
{1 + cos 2(B0 −f1 ) } , f1 ≤ |f | < 2B0 − f1 (3.17)
 4B0

0 , 2B0 − f1 ≤ |f |
The frequency f1 and the Nyquest bandwidth B0 are related by the roll-off
parameter α:
f1
α=1− (3.18)
B0
For α = 0, that is, f1 = B0 we get the Nyquist channel discussed before
see figure 3.5. The raised-cosine pulse shape in time domain is defined as:
!
p cos(2παB0 t)
p(t) = Eb sinc(2B0 t) (3.19)
1 − 16α2 B02 t2

The function of equation 3.19 exhibits two interesting properties:

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Data Communication CIE3304 Azhar Abdulaziz

Figure 3.5: (a) Raised-cosine pulse spectrum for varying roll-off rates. (b)
Pulse response p (i.e., inverse Fourier transform of for varying roll-off rates
[2].

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√
1. At t = ±T b/2 = 1/(4B0 ), we have pt = 0.5 Eb ; that is, the pulse
width measured at half amplitude is exactly equal to the bit duration
Tb .

2. There are zero crossings at t = 3Tb /2, 5Tb /2, . . . , in addition to the
usual zero crossings at the sampling times t = Tb , 2Tb , . . . .

These two properties are particularly useful in the provision of a basis for
extracting a timing signal from the receive-filter output y(t), which is used
to synchronize the receiver to the transmitter.

3.5 Passband Data Transmission

There are two types of digital signal transmission:

1. Baseband transmission: It is a direct transmission of data over a short

distance using low pass channels, like coaxial cable, telephone line and
UTP cable.

2. Passband transmission: It is useful for long distance communication

systems and has to use band pass channels like wireless media and fiber
optics cables. The binary data modulate a relatively high frequency
carrier signal.

In passband transmission, the information source binary bits modulates a

carrier sinusoidal signal with a relatively high frequency. As in analog mod-
ulation schemes, digital information can modulate the carrier amplitude, fre-
quency and phase. Because the digital information as two states only, ON
and OFF, the digital modulation usually referred as ON/OFF keying. There
are different types of digital modulation (keying):

1. Amplitude Shift Keying (ASK): It is like AM in analog, where the bits

modulate the carrier amplitude.

2. Frequency Shift Keying (FSK) : It is like FM in analogue, the bits

modulate the carrier frequency.

3. Phase Shift Keying (PSK): it is more like PM in analog, the bits mod-
ulate the carrier phase.

There are two types of digital passband receivers:

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1. Coherent (synchronized) receivers: Detection using local oscillator that

generates a sinusoidal signal of the same frequency as that of the trans-
mitter. This receiver locks the phase, and hence the frequency, with
the received signal.

2. Non-coherent detector (Envelope detector): It doesn’t require local os-

cillator, but it has high error rate.

3.5.1 Amplitude Shift Keying (ASK)

As in analog communication systems, binary information can modulate the
amplitude of the carrier signal. This means that the carrier amplitude varies
with the information level changes. ASK is also known as ON-OFF signaling.
For binary data stream b(t), where:
( √
Eb for Symbol 1
b(t) = (3.20)
0 for Symbol 0

the transmitted ASK signal, s(t) is simply generated by multiplying b(t) by

the carrier signal c(t) which is:

c(t) = Ac cos(2πfc t) (3.21)

where Ac , is the carrier amplitude, Tb is the bit duration and fc is its fre-
quency. To make average symbol energy equal, the amplitude will be:
r
2
Ac = (3.22)
Tb
Now, the transmitted ASK signal s(t) will be:
( q
2Eb
Tb
cos(2πfc t) for Symbol 1
SASK (t) = (3.23)
0 for Symbol 0

The binary ASK receiver can be based on envelope detector, which is a

non-coherent receiver as shown figure 3.7.

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b(t) × SASK (t)

c(t)

Figure 3.6: ASK generation

Figure 3.7: Binary ASK non-coherent receiver.

3.5.2 Frequency Shift Keying

In the simplest form of frequency-shift keying known as binary frequency-
shift keying (BFSK), symbols 0 and 1 are distinguished from each other by
transmitting one of two sinusoidal waves that differ in frequency by a fixed
amount. A typical pair of sinusoidal waves is described by:
 q
2Eb

Tb
cos(2πf1 t) for Symbol 1
SBF SK (t) = q (3.24)
2Eb

Tb
cos(2πf2 t) for Symbol 0

Sundes BFSK is the simplest form of a family of digitally modulated sig-

nals known collectively as continuous-phase frequency-shift keying (CPFSK)
signals, which exhibit the following distinctive property: The modulated
wave maintains phase continuity at all transition points, even though at
those points in time the incoming binary data stream switches back and
forth between symbols 0 and 1.
In other words, the CPFSK signal is a continuous-wave modulated wave
like any other angle-modulated wave experienced in the analog world, despite
the fact that the modulating wave is itself discontinuous.
In Sundes BFSK, the overall change δf in the transmitted frequency from
symbol 0 to symbol 1, or vice versa, is equal to the bit rate of the incoming
data stream (1/Tb ). In another special form of CPFSK known as minimum
shift keying (MSK), the binary modulation process uses a different value for
the frequency excursion δf is half of the bit rate.
MSK modulated wave offers superior spectral properties to Sundes BFSK.

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Figure 3.8: (a) Binary sequence and its non-return-to-zero level-encoded

waveform. (b) Its corresponding BFSK signal.

Figure 3.9: BFSK non-coherent receiver.

The MSK transmitted signal has a BW of 1.5/Tb Hz, while the Sunde’s BFSK
consumes 3/Tb Hz.

3.5.3 Phase Shift Keying (PSK)

The information binary bits can also modulate the carrier phase in what is
known as binary phase shift keying (BPSK). In BPSK, the carrier signal has
specific phase for each symbol, phases φ1 and φ2 for symbol 1 and symbol 0
respectively. So that:
 q
2Eb
 cos(2πf0 t + φ1 ) for Symbol 1
SBP SK (t) = q Tb (3.25)
2Eb

Tb
cos(2πf0 t + φ2 ) for Symbol 0

where f0 is the carrier frequency, Tb is the bit duration and Eb is the bit
energy, and symbol energy in this case because BPSK uses 1 bit per symbol.

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√
π Es
0
I

√
Distance d = 2 Es

Figure 3.10: BPSK constellation in polar form to show that optimum phase
shift is π or 180o for φ2 . The axes in constellation are I for in-phase and Q
for quadrature phase shift (π/2 or 90o ).

The best phases choices happen when φ1 = 0 and φ2 = π, because the

distance between the two symbols is at its maximum. The constellation of
the BPSK symbols is shown in figure 3.10, where d is related to symbol Es :
p p
d = 2 Es = 2 Eb (3.26)
because Es = Eb in BPSK as each symbol has 1 bit.
Figure 3.11 shows that the phases are:
(
0 i=1 for Symbol 1
φi =
π i=2 for Symbol 0

BPSK is simply generated by multiplying the bipolar NRZ line-coded

binary bits with the carrier sinusoidal signal c(t) as shown in figure. So that
the transmitted BPSK SBP SK (t) is

BPSK Bandwidth
As fb = 1/Tb ,(Tb is the bit duration), represent the maximum frequency
binary signal in bipolar NRZ form, which is the baseband bandwidth. The
spectrum of BPSK signal is centered around the carrier frequency f0 as shown
in figure 3.13. Because the BPSK main lobe extends from f0 − fb to f0 + fb ,
the bandwidth is:

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Data Communication CIE3304 Azhar Abdulaziz

Figure 3.11: Binary phase shift keying signal in time domain.

b(t)
Information Bi-Polar NRZ × SBP SK (t)

c(t)

Figure 3.12: Binary Phase Shift Keying transmitter

BW = Highest Frequency - Lowest Frequency in the main lobe.

BW = f0 + fb − (f0 − fb )
(3.27)
∴ BW = 2fb

Figure 3.13: The spectrum of BPSK signal.

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Figure 3.14: BPSK synchronous (coherent) receiver.

BPSK Coherent Receiver Example

If the transmitted BPSK signal is:
√
s(t) = b(t) 2P cos(2πf0 t)
where P = Eb /Tb which is the transmitted signal power.
The signal undergoes phase distortion through the channel. The phase
change depends on the delay from the transmitter to the receiver. Therefore,
the received signal will be:
√
r(t) = b(t) 2P cos(2πf0 t + θ)
Figure 3.14 is an example of a coherent BPSK receiver, it has the following
steps:
1. Square Law Device: It is a non-linear electronic device like a diode, the
received signal will be squared so that it will be:
cos2 (2πf0 t + θ)
The squared cosine signal will be 1/2 + 1/2cos2(2πf0 + θ), which means
it has a DC component beside the cosine part.
2. Band pass filter (BPF) to remove DC.
3. Synchronous Demodulator: it is coherent multiplier that multiplies the
received signal by the recovered signal:
√
b(t) 2P cos(2πf0 t + θ) × cos(2πf0 + θ)
| {z } | {z }
Received Signal Recovered
√
= b(t) 2P cos2 (2πf0 t + θ)
√ 1 
= b(t) 2P 1 + cos2(2πf0 t + θ)
2
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Figure 3.15: Block diagrams for (a) DPSK transmitter and (b) DPSK
receiver; for the sampler, integer i = 0, ±1, ±2, . . . .

4. Bit synchronizer and integrator: It integrates the signal over a bit

period and the synchronizer takes care of the start and end times of a
bit.

• At the end of bit time Tb , the bit synchronizer closes switch S2 to

connect the integrator output to the decision device.
• The bit synchronizer opens switch S2 and closes S1 temporarily
to discharge the integrator capacitor. The integrator is ready to
integrate the next bit by then.

3.5.4 Differential Phase-Shift Keying (DPSK)

From the above discussion, we see that both amplitude-shift keying and
frequency-shift keying lend themselves naturally to noncoherent detection
whenever it is impractical to maintain carrier-phase synchronization of the
receiver to the transmitter. But in the case of phase-shift keying, we can-
not have noncoherent detection in the traditional sense because the term
noncoherent means having to do without carrier-phase information. To get
around this difficulty, we employ a pseudo PSK technique known as differen-
tial phase-shift keying (DPSK), which, in a loose sense, does permit the use
of noncoherent detection. DPSK

Example 3.2 Illustrate the DPSK generation and detection of the informa-
tion bk = {10010011}.
Solution: Starting with the binary data stream bk given in the first row
of Table 3.1 and using symbol 1 as the first reference bit, we may construct

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Table 3.1: DPSK generation and decoding example

the differentially encoded stream dk in row 3 of the table. The second row
is the delayed version of by one bit. Note that for each index k, the symbol
dk is the complement of the modulo-2 sum of dk−1 and bk . The fourth row
of Table 3.1 defines the phase (in radians) of the transmitted DPSK signal.
The last two rows of Table 3.1 pertain to the DPSK receiver. Row 5
of the table defines the polarity (positive or negative) of the low-pass filter
output in the receiver of Figure 3.15(b). The final row of the table defines
the binary data stream produced at the receiver output, which is identical
to the input binary data stream at the top of the table, as it should be in a
noise-free environment

3.6 Multi-level Digital Modulation Schemes

Up to this point, we have discussed binary ASK, FSK and PSK in which the
modulated symbols contains one bit only. So that two levels of voltages are
modulating the carrier signal.
The advantage of binary system is that a symbol of more than one bit
could be formed before modulating the information. In multi-level digital
,modulation scheme, more than two levels are transmitted as shown in figure
3.16. For example, when symbol has two bits, the digital-to-analog converter
(DAC) convert the four levels 00, 01,10,11 to analog and modulate each one
of them separately.
Symbol rate (also called baud rate D will be lower than bit rate which
is useful for maintain and control transmission bandwidth. Baud rate is:
R
D= Symbol/second (3.28)
l
where R is the bit rate (bits/sec) and l is the number of bits per symbol.
Note that number of levels M = 2l .
r
2E  
Si (t) = cos 2πfc t + 2π(i − 1)/M (3.29)
T

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Binary Serial-to- Digital-to-

Parallel l bits analog con-
R bits/sec output verts l bits
l bits

M = 2l Symbol rate
Multi-level D = Rl
digital signal

Transmitter
Modulatd
Output

Figure 3.16: Multi-level digital transmission system.

where i = 1, 2, 3, 4; E is the transmitted signal energy per symbol and T is

the symbol duration.
The following sections will discuss some multi-level modulation patterns
with some details.

3.6.1 Quadrature Phase-Shift Keying (QPSK)

In quadrature phase shift keying QPSK, M = 4 which means that each two
consequent bits are forming a unique symbol. As with BPSK, QPSK trans-
mitted information are carried by the phase of a sinusoidal carrier. QPSK1
is allowed to be represented into the following two forms:
r
2Es  
Si (t) = cos 2πfc t + 2π(i − 1)/4 (3.30)
T
or r
2Es  
Si (t) = cos 2πfc t + (2i − 1)π/4 (3.31)
T
where i = 1, 2, 3, 4; Es is the transmitted signal energy per symbol and
T = 2Tb is the symbol duration. Each one of the four equally spaced phase
values corresponds to a unique pair of bits called a dibit.
1
Hyken and Moher call it quadriphase-shift keying in Chapter 7 in [2], rather than
quadrature phase-shift keying, so do not be surprised.

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Figure 3.17: QPSK allowed symbol constellations.

The constellation is a polar representation which shows transmitted sym-

bols as two-dimensional vectors. The QPSK’s two permitted formulas 3.30
and 3.31 are shown in Figure 3.17(a) and 3.17(b) respectively.

Example 3.3 For example, we may choose the foregoing set of phase values
to represent the Gray encoded set of dibits: 10, 00, 01, and 11. In particular,
the phase of the sinusoidal carrier takes on one of four equally spaced values,
such as π/4, 3π/4, 5π/4 and 7π/4. Figure 3.18 shows the generation and
recognition of such QPSK system.

Table 3.2: Relationship Between Index i And Identity of Corresponding

Dibit, and Other Related Matters

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Figure 3.18: Block diagrams of (a) QPSK transmitter and (b) coherent
QPSK receiver; for the two synchronous samplers, integeri = 0, ±1, ±2, . . .

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QPSK Minimum Distance of Symbols

The minimum distance between constellation points follows Pythagoras the-
orem, so that:
p p
d2min = ( Es )2 + ( Es )2
d2min = 2Es (3.32)
p
∴ dmin = 2Es

As QPSK symbol has two bits, symbol energy Es is twice the bit energy
Eb , or Es = 2Eb , hence:
p p p
dmin = 2 × 2Es = 4Eb = 2 Eb (3.33)

The distance between symbols represents the noise immunity of the sys-
tem. The QPSK has√the same symbol distance of BPSK which is twice the
symbol amplitude 2 Eb .

QPSK Bandwidth
In QPSK signal, each symbol has two bits, hence the symbol duration is twice
the bit duration, orTs = 2Tb . The spectrum of QPSK is shown in figure 3.19,
where carrier frequency f0 cancels out. The BW is:
BW = Highest Frequency - Lowest Frequency in the main lobe.

1 1
BW = − (− )
Ts Ts
(3.34)
2
=
Ts
as Ts = 2Tb then
2 1
BW = = = fb (3.35)
2Tb Tb
where fb is the bit rate or the baseband bandwidth.

Example 3.4 In QPSK system the bit rate of a NRZ stream is 10 Mbps and
carrier frequency of 1 GHz. Find:
1. The transmission symbol rate D.

2. The BW requirement of the channel.

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Data Communication CIE3304 Azhar Abdulaziz

Figure 3.19: QPSK signal power spectral density.

Solution:

1. Symbol rate D is:

1
D=
Ts
while bit rate R is
1
R=
Tb
As each symbol in QPSK has two bits, then symbol duration

Ts = 2Tb

Then symbol rate as related to bit rate is

1
D=
2Tb
1 1
=
2 Tb
R
=
2
10 × 106
=
2
∴ D = 5MHz

2. The channel BW for such QPSK is the same as the baseband BW, or:

BW = fb = 10MHz

NOTE: In real life, QPSK is used in Wireless LAN (802.11), ZigBee

(802.15.4), 3G WCDMA (UMTS)

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3.6.2 M-Ary PSK

We have seen that BPSK transmission is one bit at a time, which makes
symbols of one bit only. Hence, there are only two symbols and when the
symbol is changed, the phase shift in BPSK is:
2π 2π
Phase shift in BPSK = = =π or 180o
no.of Symbols 2
In QPSK, two successive bits form a symbol, therefore there are four
levels and whenever the symbol is changed the phase will be shift is:
2π 2π π
Phase shift in QPSK = = = or 90o
no.of Symbols 4 2
This can be extended for l bits. If l successive bits are combined to form
M symbols (levels), so that M = 2l , then the phase shift for M − P SK is:
2π
Phase shift in M-PSK = (3.36)
M
Since there are M symbols, the method is called M-ary PSK. The trans-
mitted wave in M-ary PSK is represented by:
p
s(t) = 2Ps cos(2πfc t + φi ) (3.37)

where i = 1, 2, . . . , M − 1 and the symbol phase angle is given as:

π
φi = (2i + 1) (3.38)
M
Equation 3.37 could be expanded as:
p p
s(t) = 2Ps cos(φi )cos(2πfc t) − 2Ps sin(φi )sin(2πfc t) (3.39)

Let’s arrange the waveform in 3.39 into:

r r
p 2 p 2
s(t) = Ps Ts cos(φi )cos(2πfc t) − Ps Ts sin(φi )sin(2πfc t)
Ts Ts
p p
= Ps Ts cos(φi )cos(φ1 (t)) − Ps Ts sin(φi )sin(φ2 (t))

Here r
2
φ1 (t) = cos(2πfc t) (3.40)
Ts
and r
2
φ2 (t) = sin(2πfc t) (3.41)
Ts

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Data Communication CIE3304 Azhar Abdulaziz

Figure 3.20: Signal space diagram for M-PSK modulation.

The above two equations are orthonormal, which means they are orthog-
onal (perpendicular to each other) and have normalized (which means their
amplitude 1).
In signal space diagram, the two orthonormal carriers φ1 (t) and φ2 (t) form
the axes as shown in figure 3.20. The signal points s1 , s2 , . . . , si−1 are on the
circumference of the circle. Those signal points have equal space separated
by phase√shift of 2πM
. The distance of those signals from the center of the
circle is Ps Ts , where Ps is the power of the symbol and Ts is the symbol
duration. Besides:

Ps Ts = Es Which is symbol energy (3.42)

Thus, QPSK is special form of the M-ary PSK, in which M = 4.

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φ2 (t)

dmin
θ
√
Es φ1 (t)

Figure 3.21: M-PSK dmin derivation.

Distance between Symbols

As the phase shift between θ = 2π/M , this angle is divided by 2 to get the
following relationship: the minimum distance between symbols is defined as:

dmin /2
sin(θ/2) = √
Es
dmin /2
sin(2π/2M ) = √
Es
dmin /2 (3.43)
sin(π/M ) = √
Es
dmin p
∴ = Es sin(π/M )
2 p
dmin = 2 Es sin(π/M )

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PSD
Ps Ts
fs = symbol frequency
= T1s

f
−2fs −fs fs 2fs

Figure 3.22: M-ary PSK power spectral density.

M-ary PSK Bandwidth

The power spectral density (PSD) of the M-ary PSK is:
h sin(πf T ) i2
s
S(f ) = Ts Ps (3.44)
πf Ts
The M-ary PSK BW could be derived from its spectrum as shown 3.22. As
the main power lies on the main lobe, BW is:

BW = fs − (−fs )
= 2fs
2 h 1 1 i
= ∵ fs = =
Ts Ts Symbol Period (3.45)
2 h i
= ∵ Ts = lTb
lTb
2fb h 1i
= ∵ Tb =
l Tb
where l is the number of bits per symbol and it is related to the modulation
scheme or order M = 2l . The above equation shows that when l increases
the required BW is reduced.

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Data Communication CIE3304 Azhar Abdulaziz

Figure 3.23: 16-QAM constellation diagram.

3.6.3 Quadrature Amplitude Modulation (QAM)

In M-ary PSK, symbols have equal amplitudes and they differ only by their
phases. If the amplitude and the phase varies for each symbol, the noise im-
munity will be increased. Then the technique is called quadrature amplitude
shift keying (QASK) or simply quadrature amplitude modulation (QAM).

Minimum Distance
Figure 3.23 shows 16-QAM, in which the distance from the neighboring sym-
bols is d = 2a. Then the averge symbol energy Es associated with this scheme
is: (consider the first quadrant)
1
Es = [(a2 + a2 ) + (9a2 + a2 ) + (a2 + 9a2 ) + (9a2 + 9a2 )] (3.46)
4
it is divided by 4 as there are four quadrant.
Then: p
Es = 10a2 ⇒ a = 0.1Es (3.47)
Since d = 2a:
d p
= 0.1Es
2
(3.48)
p
∴ d = 2 0.1Es
p
= 0.4Es

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Data Communication CIE3304 Azhar Abdulaziz

The last equation is the distance between two symbols in 16-QAM. Be-
cause each symbol has 4 bits, the symbol energy is four times bit energy, or
Es = 4Eb . Hence:
p p
d = 4 × 0.4Eb ⇒ d = 1.6Eb (3.49)

Transmitter and Receiver of QAM

The signal in figure 3.23 is represented by:

s(t) = k1 aφ1 (t) + k2 aφ2 (t) (3.50)

where k1 and k2 are the amplitudes which take the values ±1 or ±3. φ1 (t)
and φ2 (t) are the orthogonal carriers having the values as follows:
r
2
φ1 (t) = cos(2πfc t) (3.51)
Ts
and r
2
φ2 (t) =sin(2πfc t) (3.52)
Ts
√
From equation 3.47 we know that a = 0.1Es , therefore:
r r
Es Es
s(t) = k1 0.2 cos(2πfc t) + k2 0.2 sin(2πfc t) (3.53)
Ts Ts
We know that Es = Ps Ts ,
Es
∴ = Ps
Ts
then the signal s(t) will be :
p p
s(t) = k1 0.2Ps cos(2πfc t) + k2 0.2Ps sin(2πfc t) (3.54)

Figure 3.24 shows the transmitter for 16-QAM, which has 4 bits per
symbol. Using Gray code, the digital values are divided into 8 odd levels and
8 even levels. Therefore, Ao (t) and Ae (t) takes four levels depending upon
the combination of two input bits.
At the receiver, as shown in figure 3.25, the received signal will be raised
to 4th power and then passed through a BPF around 4fc . This is where
the carrier frequency is recovered with its phase in order to synchronize the
receiver with the transmitter (and that is the coherency concept).

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Figure 3.24: QAM Transmitter

Figure 3.25: QAM Receiver

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QAM Bandwidth
The power spectral density (PSD) of QAM is the same as in the M-ary PSK,
which is: h sin(πf T ) i2
s
S(f ) = Ts Ps (3.55)
πf Ts
Again, as the main power lies on the main lobe, BW is:

BW = fs − (−fs )
= 2fs
2 h 1 1 i
= ∵ fs = =
Ts Ts Symbol Period (3.56)
2 h i
= ∵ Ts = lTb
lTb
2fb h 1i
= ∵ Tb =
l Tb
where l is the number of bits per symbol and it is related to the modulation
scheme or order M = 2l . The above equation shows that when l increases
the required BW is reduced.

CHAPTER 3. DIGITAL TRANSMISSION AND MODULATION 76

Bibliography

[1] John G Proakis, Masoud Salehi, Ning Zhou, and Xiaofeng Li. Commu-
nication systems engineering, volume 2. Prentice Hall New Jersey, 1994.

[2] Simon Haykin and Michael Moher. Introduction to Analog and Digital
Comunications. page 540, 2007.

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