Course Book Y3S1 EC 3611 Communication Engineering 1 Sliithands - Blogspot - Com PDF

DEPARTMENT OF ELECTRICAL & COMPUTER
ENGINEERING
FACULTY OF ENGINEERING
MODULE OUTLINE
Module Name Communication Engineering I
Module Code EC3611 Version No. 2015-1
Year/Level 3 Semester 1
Credit Points 03
Pre-requisites MA1300 Engineering Mathematics I
MA1310 Engineering Mathematics II
MA2300 Engineering Mathematics III
EC2111 Signals and Systems
Co-requisites None
Methods of Delivery Lectures (Face-to-face) 2 Hours/Week
Tutorials 1 Hour/Week
Labs 2 Hours/Fortnight
Course Web Site http://courseweb.sliit.lk
Date of Original Approval September 2013
Date of Next Review September 2015
MODULE DESCRIPTION
This module introduces fundamental concepts of analogue and digital modulation
Introduction techniques and their analysis.
At the end of the module student will be able to:

Learning
Outcomes LO1: Analyse basic analogue modulation schemes
LO2: Analyse basic digital modulation schemes
LO3: Explain basic digital transmission techniques
LO4: Analyse the performance of digital transmission systems in the presence of
noise
Assessment During the semester, there will be one mid-term and a final exam. The mid-term test
Criteria will be based on the tutorial work and the lecture material covered until the week
before it is held. The final examination will be a comprehensive exam based on the
lecture materials covered during the semester.
 Midterm Examination 20% LO1

 Laboratories 20% LO1-LO4
 Final Examination 60% LO1-LO4
TOTAL 100%
Module To pass this module students are required to achieve a minimum pass in both the
Requirement “Continuous Assessments” and the “End of the Semester Examination “components
and achieve an overall mark that would qualify for a “C” grade or above.
Learning Recommended Texts

Resources
 Bernard Sklar, Digital Communications: Fundamentals and Applications, 2nd
edition, Prentice hall, 2001.
 A. B. Carlson, P. B. Crilly, J. C. Rutledge, Communication Systems: An
introduction to Signals and Noise in Electrical Communication, 4th edition,
McGraw-Hill, 2002
 B. P. Lathi, Modern Digital and Analog Communication Systems, 3rd edition,
Oxford University Press, 1998.
 S. Haykin, Communication Systems, 4th edition, Wiley, 2001.
MODULE ADMINISTRATION PROCEDURE

Contact Information
Lecturer-in-
charge
Telephone E-mail
Location
Consultation
Time
CONTENTS OF THE MODULE
1. Linear Modulation
 Standard amplitude modulation (AM)
 Double sideband suppressed carrier (DSB-SC)
 Single sideband
 An introduction to vestige sideband
2. Exponential Modulation
 Frequency modulation
 Phase modulation
3. Coherent/Incoherent Demodulation and Carrier Recovery
 Coherent demodulation: Costas loop (DSB-SC)
 Incoherent demodulation: envelope detection (standard AM)
4. Introduction to Basic Digital Communication Systems

5. Formatting of Information Sources
 Character coding
 Sampling
 Quantization
 Pulse Code Modulation (PCM)
6. Baseband Modulation
 Waveform representation of binary digits
 PCM waveform types
 M-ary pulse modulation waveforms
7. Baseband Demodulation/Detection
 Signals and noise
 Detection of binary signals in Gaussian noise
 Error probability performance of binary signalling
Generic Information
Any type of plagiarism is not allowed.

Plagiarism: Academic honesty is crucial to a student’s credibility and self-esteem, and ultimately
reflects the values and morals of the Institute as whole. A student may work together with one or
a group of students discussing assignment content, identifying relevant references, and debating
issues relevant to the subject. Plagiarism occurs when the work of another person, or persons, is
used and presented as one’s own.
End of Module Outline

Unit Study Calendar
EC3621 – Communication Engineering I
Year 3, Semester 1: 06thFeb. – 14th May, 2018
Date Remarks
Assessment
Week Lecture Tutorial Laboratory
Due
Feb. 6 The detailed

delivery may
Introduction and
1 (M. Ding) None None vary, depending
Background Review
on various
factors
Feb. 13
(M. Ding) Linear modulation

2 Tutorial 1 None
(Part 1:DSB-SC)
Feb. 20 Assignment 1
to be
Linear modulation
3 (M. Ding) Tutorial 2 None distributed on
(Part 2: Standard AM)
or before Feb.
23, 2018
Feb. 27 No makeup. In
case of medical
Linear modulation Lab 1 is an in-
(M. Ding) excuses, please
4 (Part 3: SSB, VSB) Tutorial 3 Lab 1 (tentative) class
attend the next
assessment
available
session.
March 6
Angle modulation
5 (M. Ding) Tutorial 4 Lab 1 (tentative)
(FM/PM Part 1)
March 13 Assignment 1
due by
Angle modulation
6 (M. Ding) Tutorial 5 Lab 1 (tentative) 4:00pm,
(FM/PM Part 2)
Friday, Mar.
16, 2018
March 20
Angle modulation
7 (M. Ding) Tutorial 6 Lab 1 (tentative)
(FM/PM Part 3)
March 27
8 Soft week Mid-term

April 03 Assignment 2 Based on
to be contents of
Digital Communication
9 (C. None None distributed on Week 9 to
Systems
Weeraddana) or before Apr. Week 12
06, 2018
April 10
Formatting and Base Band
10 (C. Tutorial 7 Lab 2 (tentative)
(BB) Modulation
Weeraddana)
April 24
BB Demodulation and
11 (C. Tutorial 8 Lab 2 (tentative)
Detection (Part 1)
Weeraddana)
April 30- Rescheduling

May 04 of the lecture
BB Demodulation and
12 Tutorial 9 Lab 2 (tentative) is required due
Detection (Part 2)
(C. to the holiday
Weeraddana) on May 01
May 08 Assignment 2
Amplitude Shift Keying, due by
13 (C. Frequency Shift Keying, Tutorial 10 Lab 2 (tentative) 4:00pm,
Weeraddana) and Phase shift keying Friday, May
11, 2018
14
Notes:
 Leave the relevant cell blank if no tutorial or lab is schedules or no assignment is due.
 Add columns if necessary.
 Use one document per each module per each semester
Lecture 1: Introduction
Chathuranga Weeraddana, Minhua Ding
Feb. 6, 2018
Agenda
• Course information
• Introduction
• Required basics for analog communications
1
Course information
• Lecturers:
– Dr. Minhua Ding, Analog communications (first half)
– Dr. Chathuranga Weeraddana, Digital communications (second half)
• Course website: (Enrolment code: CE2018)
http://courseweb.sliit.lk/
assignments + Midterm
• Grading: 2 labs + 2 + Final

20% 20% 60%
If plagiarism is detected for assignments for individuals, then the weights of CA components may change
for those students.
2
Course information
• Lab, assignment and exam policies:

– No negotiation with the submission deadline
– Late penalty: 10% per day
– Excuses must be made before the deadline.
• Textbooks:
– B. P. Lathi, Modern Digital and Analog Communication Systems, 3rd
edition, Oxford University Press, 1998.
– S. Haykin, Communication Systems, 4th edition, Wiley, 2001.
– Bernard Sklar, Digital Communications: Fundamentals and
Applications, 2nd edition, Prentice hall, 2001.
3
Course syllabus
• Introduction to communication systems

• Analog systems: linear modulation
• Analog systems: exponential modulation (FM, PM)
• Digital communication systems
• Baseband (BB) modulation
• Background review (e.g., random variables)
• BB demodulation detection
• Pass band modulation (ASK, FSK, PSK)
• Bit error rate performance analysis
4
Communication systems
• Today communication systems are ubiquitous.

• Our concern: the fundamental principles of communications
5
Historical sketch
Early days: the development of communication technology followed that

of electrical technology.
• Discovery of electromagnetism by Oersted and Ampère in the 1820s

=⇒ Demonstration of telegraphy
by Joseph Henry (1832) and Samuel F. B. Morse (1838)
• Hertz’s verification of Maxwell’s postulation (1873) in 1880s, predicting

the wireless propagation of electromagnetic energy.
=⇒ the radio-telegraph experiments of Marconi and Popov (within 10
years)
• Invention of diode by Fleming in 1904 and of triode amplifier by de

Forest
=⇒ rapid development of long-distance radio and wired telephony
6
Historical sketch
Subsequently: the development of communication technology drove that

of electrical technology.
• The success of telephone, patented by Alexander Graham Bell in 1876

=⇒ stimulated innumerable fundamental advances in electrical
technology, e. g., the wave filter by G. A. Campbell in 1917
• The insatiable demand for communication
=⇒ stimulated fundamental studies of signal processing, the devices and
the underlying physics
7
Historical advances
• Telephone (late 19th century)

– Converting sound, typically the human voice, into electronic signals
suitable for transmission via cables or other transmission media over
long distances
– Replaying such signals simultaneously in audible form to its user
• Wireless telegraphy (late 19th and early 20th century)

– Transmission of electric telegraphy signals without wires (wirelessly)
– Morse code was transmitted by electromagnetic waves (initially called
Hertzian waves, discovered by Heinrich Hertz in 1886)
– At early 20th century, Guglielmo Marconi began investigating the
means to signal completely across the Atlantic in order to compete
with the transatlantic telegraph cables.
8
More recently
• Radio (early 20th century)

– sending audio signals by means of electromagnetic waves
– both amplitude modulation (AM) and frequency modulation (FM)
9
Today
• Public Switched Telephone Network (PSTN)

• Radio and TV broadcasting
• Computer networks
• Cellular networks
• Satellite systems
• Global Positioning System (GPS)
• ...
10
PSTN
• the aggregate of the world’s circuit-switched telephone networks

• operated by national, regional, or local telephony operators
• providing infrastructure and services for public telecommunication
• telephone lines, fiber optic cables, microwave links, cellular networks, communications
satellites and undersea telephone cables
• originally analog; now almost entirely digital
11
Radio and TV broadcast
• AM and FM Radio
• analog TV (e.g., NTSC, PAL)
• digital TV (e.g., DVB)
12
Computer networks
• LAN: computers/devices in a limited geographical area

• WAN: telecommunications/computer networks extended to a large
geographical area
• internet: a global system of interconnected computer networks
• packet switching (no dedicated channels)
13
Cellular networks
• a communication network where the last link is wireless

• distributed over land areas called cells
• one fixed-location transceiver (base station) at each cell
• frequency, time slots, codes are reused at spatially separated cells
14
Satellite systems
• satellites cover very large areas

• different orbit altitudes: GEOs (39,000 Km), LEOs (2,000 Km)
• satellite relays and amplifies signals via a transponder
• it creates a communication channel between a source transmitter and a
receiver at different locations on Earth
15
GPS
• A satellite navigation system

• Providing specially coded satellite signals that can be processed in a GPS
receiver, enabling the receiver to compute position, velocity and time
16
Basic communication process
• Source
Taking the original signal (sound, picture) and converting it into an
electrical waveform (referred to as baseband signal or message signal)
17
• Transmitter
Modifying the message signal for efficient transmission over the channel
18
• Channel: the physical medium for message delivery

such as wire, waveguide, coaxial cable, optical fiber, or radio link
Signal distortion due to channel imperfection
Noise and interference added at the channel output
19
• Receiver
undoing the signal modifications at the transmitter and the channel
producing an estimate of the original message signal
20
• Destination: user of the information

Converting electrical waveform to original signal
21
Analog and digital messages
Analog signals
values varies continuously
e.g., temperature readings
or wind speed
Digital signals
values taken from a finite set
Binary signals
binary values (digital special case)
The classification of signals can be made more specific considering both the time-scale and the value of
the signal. Please refer to the Background for analog communications (Part 1) for details.
22
Analog and digital systems
• Analog communication systems deal with analog signals in the baseband

• Digital communication systems deals with digital signals in the baseband
23
Analog communications
• Mainly modulation theory (starting from next week)

Here we are concerned only about the basics. More advanced digital
modulation schemes will be deferred to subsequent studies.
• Modulation is used on past and current communication systems.
Radio-frequency modulation is an essential part of wireless
communication systems!
• Background on Fourier series and Fourier transform are required. Other
mathematical tools will also be used in the due course.
24
Background 1
Minhua Ding
February 2017
Signals
• A signal is a real (or complex) valued function of one or more real

variables.
– a voice signal
– atmospheric pressure as a function of altitude
– exchange rate of USD to SL Rupee at the end of each trading day
Our focus: signals with a single independent variable – time t
• Physical signals have units.
• Signals can usually be measured, e.g., for a signal g(t)
– its value at a specific t, say t = 2 =⇒ g(2)
– its energy/power, ...
1
Classification of Signals
• continuous-time and discrete-time

• continuous-valued (analog) and discrete-valued (digital)
• deterministic and random
• periodic and aperiodic
2
3
Previous page
(a) analog, continuous-time

(b) digital, continuous-time
(c) analog, discrete-time
(d) digital, discrete-time
4
Periodic and Aperiodic Signals
• A signal g(t) is said to be periodic if for some positive constant T0 > 0,
g(t) = g(t + T0), for all t
The smallest value of T0 that satisfies the above is the period of g(t).
Example: sin(2πt), ej2πt
• A signal is said to be aperiodic if there is no such value of T0 that

satisfies the above.
5
Signal Energy and Power
• the instantaneous power of a signal g(t) is p(t) = |g(t)|2

• the energy dispatched during the interval (−T /2, T /2) is
T /2
EgT = |g(t)|2dt
−T /2
• the average power dispatched during the interval (−T /2, T /2) is
T /2
1
PgT = |g(t)|2dt
T −T /2
6
Signal Energy
• The energy of a signal g(t) is
∞
Eg = |g(t)|2dt
−∞
Practically meaningful only when it is finite

• Examples: a finite pulse with a finite duration; exponentially decaying
signals
7
Signal Power
• Signals such as periodic signals have infinite energy. A suitable measure

is the time average of the energy.
• In general, the power of a signal is defined as
T /2
1
Pg = lim |g(t)|2dt
T →∞ T −T /2
This limit can be 0, e.g., for some signals of finite energy.
8
Example: Signal Power
• Calculating the power of g(t) = C cos(ω0t + θ)

Answer: C 2/2
• Calculating the power of
g(t) = C1 cos(ω1t + θ1) + C2 cos(ω2t + θ2), ω1 = ω2
Answer: 12 (C12 + C22)
9
Energy Signals and Power Signals
• A signal with finite energy is an energy signal.

(Finite energy, zero power)
• A signal with finite power is a power signal.

(Finite power, infinite energy)
10
Signal Operations: Time Shifting
g(t + T ): advance; g(t − T ): delay
11
Signal Operations: Time Scaling
g(at), a > 1: compression in time; g(at), 0 < a < 1: expansion in time
12
Signal Operations: Time Reversal
g(−t)
13
Unit Impulse Function
• Dirac delta function δ(t): an infinitely narrow pulse with area 1,
unbounded at t = 0

δ(t) = 0, t = 0
∞
−∞
δ(t)dt = 1
14
Property of Unit Impulse Function
• g(t)δ(t) = g(0)δ(t) (assuming g(t) continuous at t = 0)

• Sampling (or sifting) property: (assuming g(t) continuous at t = T )
∞
g(t)δ(t − T )dt = g(T )
−∞
• Convolution (for regular continuous function g(t))

∞ ∞
g(t) ∗ δ(t) = δ(τ )g(t − τ )dτ = δ(τ )g(t)dτ
−∞ −∞
∞
= g(t) δ(τ )dτ = g(t)
−∞
15
Unit Step Function
• Heaviside unit step function u(t):

1 t>0
u(t) =
0 t<0
t du(t)
• We can verify that −∞
δ(τ )dτ = u(t) and thus δ(t) = dt
16
Fourier Series of Periodic Signals
• Periodic signals can be expressed1 as a sum of sinusoids whose frequencies

are integer multiples of the fundamental frequency f0 = T10 . (T0 is the
period.)
∞
g(t) =a0 + [an cos(2πnf0 t) + bn sin(2πnf0 t)]
n=1
where the coefficients can be calculated as

t1 +T0 t1 +T0
1 2
a0 = g(t)dt, an = g(t) cos(2πnf0 t)dt
T0 t1 T0 t1
t1 +T0
2
bn = g(t) sin(2πnf0 t)dt
T0 t1
1
Note: The equivalence here is often in terms of the total energy of the signal.
17
• Alternatively:
∞

g(t) = C0 + Cn cos(2πnf0 t + θn),
n=1
where
C0 =a0

Cn = a2n + b2n
−1 −bn
θn = tan
an
18
• Fourier series using complex exponentials
∞

g(t) = Dnej2πnf0t
n=−∞
where

1
Dn = g(t)e−j2πnf0tdt
T0 T0
19
Example: Fourier Series of Periodic Square Wave (1/5)
This example is taken from [Lathi and Ding, 4th edition, P57, Example 2.8]
20
A periodic square wave of period 2π is defined as follows. Over the

interval [−π, π]

1 |t| < π2
w(t) =
0 π2 < |t| < π
We want to express it as:

∞
w(t) = a0 + n=1 [an cos(2πnf0 t) + bn sin(2πnf0 t)]
21
1 2π
• Here T0 = 2π, f0 = T0 (ω0 = 2πf0 = T0 )
T0 /2
1 1
a0 = 1 · dt =
T0 −T0 /2 2
T0 /4
2 1 π/2
an = cos(2πnf0 t)dt = cos(nt)dt
T0 −T0 /4 π −π/2
⎧
⎪
⎨ 0 n even
2 nπ 2
= sin = nπ n = 1, 5, 9, 13, . . .
nπ 2 ⎪
⎩ 2
− nπ n = 3, 7, 11, 15, . . .

2 T0/4
bn = sin(2πnf0 t)dt = 0
T0 −T0/4
22
23
1 2π
Here T0 = 2π, f0 = T0 (ω0 = 2πf0 = T0 )
∞

w(t) =a0 + [an cos(2πnf0 t) + bn sin(2πnf0t)]
n=1

1 2 1
= + cos(2πf0t) − cos(2π · 3f0t) + . . .
2 π 3

1 2 1
= + cos t − cos(3t) + . . .
2 π 3
24
Background 2
Minhua Ding
February 2017
From Fourier Series to Fourier Transform
• Consider g(t), periodic with period T0. Recall its Fourier series using
complex exponentials:
∞
∞
1
g(t) = Dnej2πnf0t = g(t)e−j2πnf0tdt ej2πnf0t
n=−∞ n=−∞
T0 T0
1
• Let T0 → ∞. Then f0 = T0 → Δf (or df ), nf0 → f
∞

g(t) → g(t)e−j2πnf0tdt ej2πnf0tΔf
n=−∞ T0 →∞
∞ ∞
→ g(t)e−j2πf t dt ej2πf tdf
−∞ −∞
G(f )
1
From Fourier Series to Fourier Transform
Periodic square wave, when T0 → ∞
0.6 0.3
0.4 0.2
0.2 0.1
0 0
−0.2 −0.1
0 0
0.15 0.06
0.1 0.04
0.05 0.02
0 0
−0.05 −0.02
0 0
Note: all the plots are for Dn versus n in the square wave case.
2
Fourier Transform
Let g(t) denote a signal (real or complex).

• The Fourier transform of g(t) is denoted F[g(t)]
∞
G(f ) = F[g(t)] = g(t)e−j2πf tdt
−∞
• The inverse Fourier transform is given by

∞
−1
g(t) = F [G(f )] = G(f )ej2πf tdf
−∞
∞
Not all signals have Fourier transform. A sufficient condition is given as follows. If −∞ |g(t)|dt < ∞,
then G(f ) exists. Note that this condition is sufficient ∞ but not necessary for the existence of Fourier
2
transform. In addition, if g(t) has finite energy, i.e., −∞ |g(t)| dt < ∞, then G(f ) exists for “most”
frequencies.
3
Fourier transform of a rectangular pulse
Here
1
|t| < τ2
g(t) = τ
0 otherwise
with area 1.
ͳȀ߬
t
Ͳʏ/2 0 ʏ/2
4
Fourier transform of a rectangular pulse
τ /2
1 −j2πf t sin(πτ f )
G(f ) = e dt = = sinc(τ f )
−τ /2 τ πτ f
sin(πx)
Here sinc(x) πx .
1
Ͳ2/ʏ Ͳ1/ʏ 0 1/ʏ 2/ʏ

0
5
Signal Bandwidth
• Signal Bandwidth: the difference between the highest (significant)

frequency to the lowest (significant) frequency in the signal spectrum.
• Example: A signal with the following spectrum is said to have a bandwidth
of B.
f
ͲB 0 B
• The bandwidth of the rectangular pulse (see previous 2 pages) is

approximately 1/τ . The narrower the pulse, the wider the spectrum.
6
Fourier transform of δ(t)
Now we know that for the rectangular pulse:

1
|t| < τ2
g(t) = τ
0 otherwise
τ /2
1 −j2πf t sin(πτ f )
G(f ) = e dt = = sinc(τ f )
−τ /2 τ πτ f
Consider the limiting case: δ(t) = limτ →0 g(t),
F[δ(t)] = lim F[g(t)] = lim sinc(τ f ) = 1

τ →0 τ →0
Therefore,

F[δ(t)] = 1

7
Fourier transform: Time shifting and frequency shifting
• Time shifting
F [g(t)] = G(f )
F [g(t − t0)] = G(f )e−j2πf t0
• Frequency shifting
F [g(t)] = G(f )

F g(t)e j2πf0 t
= G(f − f0)
8
Fourier transform of sinusoids
• It can be shown that F [1] = δ(f ). Using
1 jθ −jθ
1 jθ −jθ

cos θ = e +e , sin θ = e −e
2 2j
and the frequency shifting property, we have

j2πf t
e 0 + e−j2πf0 t
F [cos(2πf0 t)] =F
2
1
= [δ(f − f0) + δ(f + f0)]
2
j2πf t −j2πf0 t

e 0 −e
F [sin(2πf0 t)] =F
2j
1
= [δ(f − f0) − δ(f + f0)]
2j
9
Modulation theorem (1/2)
One of the basic modulation schemes is simply given by1
m(t) cos(2πfct)
If F[m(t)] = M (f ), then
1
F [m(t) cos(2πfct)] = [M (f − fc) + M (f + fc)]
2
Comment:
• fc is called carrier frequency.

• We can also use sin(2πfct) for modulation.
1
This is called double sideband suppressed carrier (DSB-SC) modulation. We will discuss it in detail.
10
A basic modulation scheme (2/2)
• Let m(t) satisfy F[m(t)] = M (f ) = 0, |f | > B and let fc B.
Then the spectrum of m(t) cos(2πfct) is restricted to fc − B ≤ |f | ≤

fc + B.
Its bandwidth is thus given by
(fc + B) − (fc − B) = 2B
• In practice, if m(t) is not band-limited, we have to filter them.
11
Fourier transform: Time-domain Convolution
• Time-domain convolution, frequency-domain multiplication

The convolution in time is given by
∞
g1(t) ∗ g2(t) = g1(τ )g2 (t − τ )dτ
−∞
If F [g1(t)] = G1(f ), F [g2(t)] = G2 (f ), then

F [g1(t) ∗ g2(t)] = G1(f )G2(f ) convolution theorem

12
Fourier transform: Time-domain multiplication
• Time-domain multiplication, frequency-domain convolution

If F [g1(t)] = G1(f ), F [g2(t)] = G2 (f ), then

∞
F [g1(t)g2(t)] = G1(f ) ∗ G2(f ) = −∞
G1(α)G2(f − α)dα

13
Systems
• A system has an input signal and an output signal.
input output
signal signal
system
The output is related to the input through system operation.

• Examples: a hi-fi system, a camera
14
Linear and Time-Invariant (LTI) Systems
• Let yi(t) be the output signal (response) given the input signal xi(t),
i = 1, 2. A system is linear if
(1) Given the input x1(t) + x2(t), the output is y1(t) + y2(t) [additivity]
(2) Given the input ax1(t), the output is ay1(t) [scaling or homogeneity]
Let y(t) be the output signal (response) given the input signal x(t).
A system is time-invariant if given the input x(t − t0), the output is
y(t − t0)
• We will mostly deal with LTI systems such as filters.
15
LTI Systems
• Impulse response of an LTI system h(t): the output of the system given
δ(t) as the input
ߜሺ‫ݐ‬ሻ ݄ሺ‫ݐ‬ሻ
LTIsystem
݃ሺ‫ݐ‬ሻ ‫ݕ‬ሺ‫ݐ‬ሻ ൌ ݃ሺ‫ݐ‬ሻ ‫݄ כ‬ሺ‫ݐ‬ሻ
• Fundamental fact: The response/output y(t) of an LTI system given a

continuous but otherwise arbitrary input g(t) is given by

∞
y(t) = g(t) ∗ h(t) = −∞
g(τ )h(t − τ )dτ

16
Transfer function of LTI Systems

∞
y(t) = g(t) ∗ h(t) = −∞
g(τ )h(t − τ )dτ

• In communication system analysis, we often deal with the spectrum.

• Let g(t) ←→ G(f ), h(t) ←→ H(f ), y(t) ←→ H(f ). Using convolution
theorem, we have
Y (f ) = H(f )G(f ).
• H(f ) is referred to as the system transfer function.
We use the notation g(t) ←→ G(f ) to denote F [g(t)] = G(f )
17
Distortionless systems
• Desired: a replica of the original signal from message sender to message

receiver
• Distortionless if y(t) = kg(t − td), k > 0 (scaling + time delay only); in
frequency domain:
Y (f ) = kG(f )e−j2πf td
Y (f )
H(f ) = = ke−j2πf td
G(f )
It follows that

|H(f )| = k, θh(f ) = −2πtdf

18
Ideal filters
ȁ‫ ܪ‬ሺ݂ሻȁ
ͳ
f
Ͳ
െ݂௨ െ݂௟ ݂௟ ݂௨
‫ ܤ‬ൌ ݂௨ െ ݂௟
ͳ ሺ݂௨ ൌ ‫ܤ‬ǡ ݂௟ ൌ Ͳ)

f
െ‫ܤ‬ 0 ‫ܤ‬
ͳ ሺ݂௨ ൌ λ)
െ݂௟ Ͳ ݂௟ f
(Upper) ideal band-pass filter (BPF); (Middle) ideal low-pass filter (LPF);
(Lower) ideal high-pass filter (HPF)
19
Ideal LPF
• g(t): bandlimited to B, G(f ) = 0, |f | > B. The ideal low-pass filter

gives y(t) = g(t − td), i.e.,
H(f ) =|H(f )|e−j2πf td
|H(f )| =1, |f | < B
←→ F[|H(f )|] =2Bsinc(2Bt)
hLP(t) =F[H(f )] = F[|H(f )|e−j2πf td ] = 2Bsinc (2B(t − td))
Comment: non-causal (what is a causal system?)
20
Ideal LPF
21
Example: Ideal low-pass Filters
Before LPF
2
−2
−8 −6 −4 −2 0 2 4 6 8
t
After LPF
1
cos(0.4π t)
−1
−8 −6 −4 −2 0 2 4 6 8
t
Upper: g(t) = cos(0.4πt) + cos(1.8πt). Using an ideal LPF filter
H(f ) = 1, |f | < 12 ←→ h(t) = sinπtπt , we get y(t) = cos(0.4πt) (lower)
22
Practical low-pass filters
Butterworth filters: |H(f )| = √ 1
1+(f /B)2n
Butterworth n=1
1
n=2
n=3
n=4
0.8 n=8
n=16
n=∞
|H(f)|
0.6
0.4
0.2
0
0 1 2
f/B
23
Butterworth (tradeoff between amplitude response and phase response)
24
1
j2πf C α 1
H(f ) = 1 = , α=
R + j2πf C
α + jf 2πRC
1
|H(f )| = ≈ 1, f
α
1+ (f /α)2
θh(f ) = − tan−1(f /α) = −f /α, f
α
Note: first-order Butterworth!!
25
0.8
0.6
|H(f)|
0.4
0.2
0
0 1 2 3 4 5
f
26
Low-pass filtering the DSB-SC signal
m(t) = cos 2πfmt, s(t) = m(t) cos 2πfct. Low-pass filtering the DSB-
SC signal:
ͳ
ሾ‫ܯ‬ሺ݂ െ ݂௖ ሻ ൅ ‫ܯ‬ሺ݂ ൅ ݂௖ ሻሿ
ʹ LPF2
LPF1
݂
െ݂௖ െ ݂௠ െ݂௖ ൅ ݂௠ ݂௖ െ ݂௠ ݂௖ ൅ ݂௠
27
Lecture 2: DSB-SC amplitude modulation
Minhua Ding
Feb. 20, 2018

Agenda
• Introduction to modulation
• Amplitude modulation
– Double-sideband suppressed-carrier (DSB-SC) amplitude modulation
(today)
– Standard amplitude modulation (standard AM) (next lecture)
1
Modulation
Modulation: a process that causes a shift in the range of signal

frequencies
Why? For example,
• Wireless communication requires effective radiation (antenna size ∝

wavelength of the radiated signal) and uses modulation to shift low
frequencies of baseband signals to a higher frequency
Voice signal: 300-3400 Hz; GSM carrier frequency: around 900MHz
• Frequency-division multiplexing (FDM)

Dividing the available spectrum into a number of frequency bands
Allocating different bands for different users/devices.
2
Modulation
• The idea of modulation: to shift the spectrum of a signal by using a

carrier. Consider a sinusoidal carrier with frequency fc:
A cos(ωct + φ) =A cos(2πfc t + φ)
ωc =2πfc
• The message signal m(t) (e.g., telephone voice signal) modifies the
amplitude, frequency, or phase of a sinusoid carrier.
• The modulated signal
x(t) = A(t) cos(2πfc t + φ(t))
3
Basic modulation types
The modulated signal
x(t) = A(t) cos( ωct + φ(t) )

generalized angle
• Amplitude modulation (AM): A(t) proportional to m(t)

• Angle modulation:
– Frequency modulation (FM): ωc (angular frequency) can be varied to
be a function of t, i.e., ωc(t), and ωc(t) is proportional to m(t).1
– Phase modulation (PM): φ(t) proportional to m(t)
• Composite modulation: single-sideband (SSB) and vestigial-sideband
(VSB) modulation
1
This implies the instantaneous frequency fc ∝ m(t), since ωc = 2πfc .
4
Convention
The modulating signal m(t) is generally band-limited (to B Hz), i.e.,
m(t) ←→ M (f ), and M (f ) = 0, for |f | > B
If the signal is not band-limited, we need to low-pass filter it first.
Let fc be the carrier frequency. In practice, one typically chooses
fc B.
5
Amplitude modulation
In amplitude modulation, the carrier amplitude varies linearly2 with the

message signal m(t), i.e.,
A(t) = k · m(t) + c
The resulting modulated signal can be written as
x(t) = [km(t) + c] cos(2πfc t + φ)
where k, c, φ are constants.

To simplify mathematical presentation, we often set φ = 0.
m(t) is referred to as the (base-band) message signal or the modulating
signal.
2
In fact, when c = 0, the relation between A(t) and m(t) is affine.
6
Double-sideband suppressed-carrier (DSB-SC)
modulation
• The simplest form of amplitude modulation is when k = 1, c = 0, i.e.,
x(t) = m(t) cos(2πfct) = m(t) cos ωct
– This is known as the DSB-SC modulation. It is also referred to as the

product modulation.
– Certainly, the amplitude of the carrier can be chosen to be any proper
positive value Ac(Ac > 0). Here Ac = 1.
7
A product modulator for DSB-SC
m(t) m(t) cos2ʌfct

modulating signal modulated signal
cos2ʌfct
(Carrier)
x(t) = m(t) cos ωct = m(t) cos(2πfct)
8
DSB-SC modulated signal (1)
A DSB-SC modulated signal looks like this.
9
DSB-SC modulated signal (2)
0.5 0.5
m(t)cos(ωc t)
m(t)
PR
0 0
−0.5 −0.5
0 0.5 1 0 0.5 1
t t
0.5 0.5
m(t)cos(ωc t)
PR PR
m(t)
0 0
−0.5 −0.5
−0.5 0 0.5 −0.5 0 0.5
t t
10
The waveforms of the modulated signals are those of sinusoidal waves
with time-varying amplitude.
At the points with the red mark PR, phase reversals of 180◦ can be
observed. This happens when the message signal m(t) changes its sign
(polarity).
The phenomenon of phase reversal is typical for DSB-SC. This is also

known as envelope distortion.
11
DSB-SC modulated signal
The envelope of a signal E(t) cos ωct is given by E(t) if E(t) ≥ 0.
In our case

m(t) m(t) > 0
the envelope of m(t) cos ωct = = |m(t)|
−m(t) m(t) < 0
In DSB-SC, the envelope carries the information of |m(t)|, instead of m(t).

This explains the envelope distortion.
By knowing |m(t)|, we may not be able to know exactly what m(t) is.
12
Spectrum of a specific DSB-SC modulated signal
• Let m(t) = Am cos(2πfmt), where Am and fm are constant amplitude

and frequency, respectively. The DSB-SC modulated signal is given by
x(t) =m(t) cos(2πfct)

=Am cos(2πfmt) cos(2πfct)
Am
= [cos(2π(fc − fm)t) + cos(2π(fc + fm)t)]
2
– The above shows that the DSB-SC signal x(t) consists of 2 frequency
components with the same amplitude: fc − fm, fc + fm.
– The component with frequency fc − fm: the lower side frequency; The
component with frequency fc + fm: the upper side frequency
– The carrier (with frequency fc) is NOT in the modulated output. This
is why this is called double-sideband suppressed-carrier (DSB-SC).
13
The Fourier transform (frequency spectrum) of x(t) is given by
Am
X(f ) = [δ(f − (fc − fm)) + δ(f + (fc − fm))
4
+ δ(f − (fc + fm)) + δ(f + (fc + fm))]
14
Spectrum of a GENERAL DSB-SC modulated signal
In general, instead of a single sinusoid, the message (modulating) signal
m(t) has a frequency spectrum M (f ). In this case, the frequency spectrum
of the modulated signal x(t), denoted X(f ), will have two sidebands, as
shown here.
‫ ܯ‬ሺ݂ ሻ
ʹ‫ܣ‬
െ‫ܤ‬ ‫ܤ‬ f
ͳ
ܺሺ݂ሻ ൌ ሾ‫ܯ‬ሺ݂ െ ݂௖ ሻ ൅ ‫ܯ‬ሺ݂ ൅ ݂௖ ሻሿ
ʹ
USB
LSB
‫ܣ‬
f
െሺ݂௖ ൅ ‫ )ܤ‬െ݂௖ െሺ݂௖ െ ‫ܤ‬ሻ ݂௖ െ ‫ܤ‬ ݂௖ ݂௖ ൅ ‫ܤ‬
15
Spectrum of a GENERAL DSB-SC modulated signal
• The general DSB-SC modulated signal is given by x(t) =

m(t) cos(2πfct), and its Fourier transform is given by
1
X(f ) = [M (f + fc) + M (f − fc)] .
2
• LSB: lower sideband; USB: upper sideband
• For real-valued message signals, the LSB and USB form a mirror image
pair.
• It is observed from the figure on previous page that the bandwidth of the
DSB-SC signal is given by
DSB-SC signal bandwidth = (fc + B) − (fc − B) = 2B
where B is the bandwidth of the message (modulating) signal, and

fc B in general.
16
Demodulation of a DSB-SC (product modulated) signal
‫ݔ‬ሺ‫ݐ‬ሻ ൌ ݉ሺ‫ݐ‬ሻ ሺʹߨ݂௖ ‫ݐ‬ሻ ݁ሺ‫ݐ‬ሻ ‫ݒ‬௢ ሺ‫ݐ‬ሻ

LPF
DSB_SC signal
‫ݒ‬௅ ሺ‫ݐ‬ሻ (Local carrier)
• The base-band message signal m(t) can be recovered from a DSB-SC

signal x(t) by multiplying x(t) with a local carrier and then low-pass
filtering the product. This recovery process is called demodulation.
• The local carrier vL(t) is generated by a local oscillator in the receiver
vL(t) = cos [2π(fc + Δf )t + φ]
where Δf is a small frequency difference and φ is the phase (with respect

to that of cos(2πfc t)).
17
The DSB-SC modulated signal x(t) is demodulated by multiplying it

with vL(t). The output of the multiplier is
e(t) =x(t)vL (t) = m(t) cos(2πfct) · cos [2π(fc + Δf )t + φ]

1
= m(t) {cos(2π · Δf · t + φ) + cos [2π(2fc + Δf )t + φ]}
2
The required base-band signal m(t) is recovered by low-pass filtering e(t),
so that the second component shown above (centered at 2fc) is filtered
out. The filtered output is given by
1
vo(t) = m(t) cos(2π · Δf · t + φ)
2
Ideally, for zero distortion in the recovered output, we required that vo(t) ∝
m(t). This condition can be satisfied when (2π · Δf · t + φ) = 0.
18
• When φ = 0, vo(t) = 12 m(t) cos(2π Δf t). In this case, the recovered

output consists if the desired information signal m(t) beating with a low
frequency component Δf .
As a result, vo (t) suffers from periodic deep fades which led to
unacceptable distortion.
• When Δf = 0, vo(t) = 12 m(t) cos φ. The amplitude of the recovered
output is affected by the phase of the local carrier. When φ = 90◦, the
output is zero!
Ideally, φ needs to be zero for maximum amplitude. In this case, the
local oscillator is said to be phase locked to the transmit carrier.
19
Coherent demodulation (Δf = 0, φ = 0)
• When Δf = 0 and φ = 0, the local carrier vL(t) = cos(2πfct), which is

perfect synchronized with the carrier used at the transmitter.
• In this case, the demodulation is called synchronous or coherent
demodulation, which means that local carrier has the same frequency
and phase as the transmit carrier.
• Here
1 1
e(t) = x(t)vL(t) =m(t) cos2(2πfct) = m(t) + m(t) cos(4πfc t)
2 2
The filtered output is given by vo (t) = 12 m(t) (NO distortion).
An illustration is given in the next page.
20
Coherent demodulation (Δf = 0, φ = 0)
The desirable low-pass filter should be flat over the signal bandwidth.
21
DSB-SC demodulation using Costas receiver
The Costas receiver has a phase-locked loop used in practical
synchronous/coherent receiver for demodulating DSB-SC signal waves.
IͲchannel
૚
Product LowͲpass ૛ ࡭ࢉ ࢓ሺ࢚ሻ‫ܛܗ܋‬ሺ૛ૈοࢌ࢚ ൅ ࣘሻ
Demodulated
modulator filter
‫ݔ‬ூ ሺ‫ݐ‬ሻ signal
‫ܛܗ܋‬ሺ૛࣊ሺࢌࢉ ൅ οࢌሻ࢚ ൅ ࣘሻ Voltagecontrolled Phase

oscillator(VCO) discriminator
DSBͲSCsignal
࡭ࢉ ࢓ሺ࢚ሻ‫ܛܗ܋‬ሺ૛࣊ࢌࢉ ࢚ሻ
െૢ૙‫ ܗ‬phase
shifter
‫ܖܑܛ‬ሺ૛࣊ሺࢌࢉ ൅ οࢌሻ࢚ ൅ ࣘሻ
Product LowͲpass
modulator filter
૚
‫ݔ‬ொ ሺ‫ݐ‬ሻ ࡭ ࢓ሺ࢚ሻ‫ܖܑܛ‬ሺ૛ૈοࢌ࢚ ൅ ࣘሻ
૛ ࢉ
QͲchannel
22
• The Costas receiver has two multipliers supplied with the same incoming
DSB-SC signal, but with individual local oscillator signals of the same
frequency but having 90◦ difference in phase.
xI (t) =Acm(t) cos(2πfct) cos(2π(fc + Δf )t + φ)

xQ(t) =Acm(t) cos(2πfct) sin(2π(fc + Δf )t + φ)
The demodulator frequency fc = fc + Δf deviates slightly from the

transmit carrier frequency fc, and there is a phase difference φ.
• Recall the trigonometric identities:
1
cos α cos β = [cos(α + β) + cos(α − β)]
2
1
cos α sin β = [sin(α + β) − sin(α − β)]
2
23
It can be shown that
Ac
xI (t) = m(t) [cos(2π(2fc + Δf )t + φ) + cos(2πΔf t + φ)]
2
Ac
xQ(t) = m(t) [sin(2π(2fc + Δf )t + φ) + sin(2πΔf t + φ)]
2
These two outputs from the multipliers are low-pass filtered, so that only
the low-frequency parts remain.
Ac
zI (t) = m(t) cos(2πΔf t + φ)
2
Ac
zQ(t) = m(t) sin(2πΔf t + φ)
2
The upper branch is called the in-phase channel (I-channel) and the lower
branch is called the quadrature channel (Q-channel).
24
• The I-channel and the Q-channel outputs are applied to a phase

discriminator (i.e., a multiplier followed by a low-pass filter), to produce
a dc control signal for automatically correcting the phase errors in the
local oscillator (in the form of a VCO).
• For proper operation, the loop has to achieve rapid lock-up (Δf = 0, φ =
0) (carrier recovery).
• The desired output is recovered from the output of the I-channel, under
the condition that the local oscillator is of the same frequency and phase
as the transmit carrier, i.e., Δf = 0, φ = 0.
• Let φ(t) = 2πΔf t + φ. Suppose the local oscillator drifts from its proper
value by a small angle φ(t). The I-channel will be essentially the same,
while the Q-channel output will be proportional to sin φ(t) ≈ φ(t) for
small values of φ(t).
25
A discussion of the signal power
• Calculate the power of
g(t) = C cos(ω0t + θ), ω0 = 0.
Here θ is an arbitrary constant.
26
T /2 T /2
1 2 1
Pg = lim g (t)dt = lim C 2 cos2(ω0t + θ)dt
T →∞ T −T /2 T →∞ T −T /2
T /2
1 C2
= lim [1 + cos(2ω0t + 2θ)] dt
T →∞ T −T /2 2
2 T /2 2 T /2
C C
= lim 1dt + lim cos(2ω0t + 2θ)dt
T →∞ 2T −T /2 T →∞ 2T −T /2

C 2 /2 finite
→0
=C 2/2
27
Signal Power: Example 2
Sum of 2 sinusoids with different frequencies
g(t) = C1 cos(ω1t + θ1) + C2 cos(ω2t + θ2)

ω1 = ω2, ω1 = 0, ω2 = 0
Here θ1, θ2 are two arbitrary constants.
28
g(t) = C1 cos(ω1t + θ1) + C2 cos(ω2t + θ2), ω1 = ω2, ω1 = 0, ω2 = 0
T /2 T /2
1 2 1
Pg = lim g (t)dt = lim C12 cos2(ω1t + θ1)dt
T →∞ T −T /2 T →∞ T −T /2
T /2
1
+ lim C22 cos2(ω2t + θ2)dt
T →∞ T −T /2
T /2
2C1C2
+ lim cos(ω1t + θ1) cos(ω2t + θ2)dt
T →∞ T
−T /2
finite
→0
C12 C22
= +
2 2
29
Sum of 2 Sinusoids (same frequency, different phases)
Calculate the power of the following signal:
g(t) = C1 cos(ω1t + θ1) + C2 cos(ω1t + θ2), ω1 = ω2, ω1 = 0.
Assume that θ1 = θ2 in general.
30
Assume that θ1 = θ2 in general. It can be shown that
C12 + C22 + 2C1C2 cos(θ1 − θ2)

pg = .
2
31
Use the identity: cos(α + β) = cos α cos β − sin α sin β
g(t) =C1 cos(ω1 t + θ1) + C2 cos(ω1 t + θ2)
=[C1 cos(ω1t) cos θ1 − C1 sin(ω1t) sin θ1]
+ [C2 cos(ω1 t) cos θ2 − C2 sin(ω1 t) sin θ2]
= cos(ω1t) [C1 cos θ1 + C2 cos θ2]

E
− sin(ω1t) [C1 sin θ1 + C2 sin θ2]

F

= E 2 + F 2 cos(ω1t + φ)
F
where φ is the 4-quadrant inverse of tan E . The last line shows that g(t)
is a simple sinusoid with a fixed amplitude, a single frequency and a fixed
phase. Its power can be calculated as
E2 + F 2 C12 + C22 + 2C1 C2 cos(θ1 − θ2)
pg = = .
2 2
32
Lecture 3: Standard AM
Minhua Ding
Feb. 20, 2018

Agenda
• Amplitude modulation
– DSB-SC (last lecture)
– Standard AM (today)
1
• A basic problem associated with DSB-SC modulation is the need for
coherent demodulation, which means the local oscillator at the receiver
must be of the same frequency and phase as the transmit carrier.
This can be achieved by using a phase-locked loop such as that in Costas
receiver. But this leads to higher receiver complexity and cost.
• A method to avoid the need for carrier recovery is to add a carrier
component into the modulated signal.
Consider the following amplitude modulated signal
xAM(t) =Ac[1 + kam(t)] cos(2πfc t)

= Ac cos(2πfct) + Ackam(t) cos(2πfc t)

the desired carrier
in correct phase
where Ac, fc are the carrier amplitude and frequency, respectively, m(t)
is the baseband message signal, and ka > 0 is a constant (amplitude
sensitivity).
2
Standard amplitude modulation (Standard AM)
When |kam(t)| < 1 for all t, the envelope of xAM (t) has the same shape
as the message signal m(t) (no phase reversal). Thus, one may recover the
message signal from the envelope of the amplitude modulated signal using
an envelope detector. This form of AM with |kam(t)| < 1 for all t is called
the standard AM.
3
Standard AM
• It is the oldest modulation, widely used in AM radio broadcasting.

• The standard AM signal can be demodulated using the envelope detector,
which is a simple and low-cost device.
• It requires |kam(t)| < 1 for all t:
xAM(t) =Ac [1 + kam(t)] cos(2πfc t)

>0
• In addition, as in DSB-SC, fc B is typically required and assumed,

where B is the baseband signal bandwidth.
4
When |kam(t)| > 1 for some t, the envelope of xAM (t) no longer
resemble exactly the shape of the message signal m(t). Whenever 1+kam(t)
crosses zero, the carrier wave undergoes 180◦ phase reversals.
The condition that |kam(t)| > 1 is referred to as over-modulation. In

this case, the message signal m(t) cannot be recovered using an envelope
detector without distortion. But m(t) can be recovered using coherent
detection (as in DSB-SC).
5
Standard AM signal – Frequency spectrum (1/2)
• The standard AM signal:
xAM(t) = Ac[1 + kam(t)] cos(2πfct)
• The Fourier transform of xAM(t) is
Ac
XAM(f ) = [δ(f + fc) + δ(f − fc)]
2
Acka
+ [M (f + fc) + M (f − fc)]
2
How to get this?
6
Standard AM signal – Frequency spectrum (2/2)
‫ ܯ‬ሺ݂ ሻ
f
െ‫ܤ‬ Ͳ ‫ܤ‬
‫ܣ‬௖ ‫ܣ‬௖
ߜሺ݂ ൅ ݂௖ ሻ ߜሺ݂ െ ݂௖ ሻ
ʹ ʹ ‫ܣ‬௖ ݇௔
‫ܣ‬௖ ݇௔ ‫ܯ‬ሺ݂ െ ݂௖ ሻ
‫ܯ‬ሺ݂ ൅ ݂௖ ሻ ʹ
ʹ
f
െ݂௖ െ ‫ ܤ‬െ݂௖ െ݂௖ ൅ ‫Ͳ ܤ‬ ݂௖ െ ‫ܤ‬ ݂௖ ݂௖ ൅ ‫ܤ‬
The bandwidth (BW) of the standard AM signal:
=(fc + B) − (fc − B)
=2B
7
Example: Single-tone AM
• Let m(t) = cos ωmt. This is referred to as a single-tone signal with
frequency fm. Here
ϕAM(t) = Ac(1 + μm(t)) cos ωct = Ac(1 + μ cos ωmt) cos ωct
where Ac is the carrier amplitude, and μ > 0 is the termed the modulation
index. Using single-tone standard AM , the maximum value for μ is
μ = 1.
The peak/maximum value of the envelope of ϕAM(t) is Amax = Ac(1 +

μ), while its minimum is Amin = Ac(1 − μ). Therefore,
1 Amax − Amin Amax − Amin
Ac = (Amax + Amin), μ= =
2 2Ac Amax + Amin
The above can be used to obtain the value of modulation index μ directly
from the wave.
8
m(t)
2
−2
0 0.5 1 1.5 2
t
single−tone standard AM, μ=0.5
2
0
−2
0 0.5 1 1.5 2
t
single−tone standard AM, μ=1
2
0
−2
0 0.5 1 1.5 2
t
9
Single-tone AM signal spectrum
Ac
ϕAM (f ) = [δ(f − fc ) + δ(f + fc )]
2
Ac μ
+ [δ(f − (fc + fm)) + δ(f + (fc + fm ))]
4
Ac μ
+ [δ(f − (fc − fm )) + δ(f + (fc − fm))]
4
10
Switching modulator
• Same as DSB-SC signals, the standard AM signal can be generated using

a multiplier.
• In practice, the switching characteristic of a diode is often used for
generating an amplitude modulated signal wave. The modulator of this
type is termed the switching modulator . Here it is assumed that a diode
acts as an ideal switch.
• Consider the following voltage signal vi (t), which is the sum of the
baseband message (modulating) signal and the sinusoidal carrier with
amplitude Ac and frequency fc.
vi(t) = Ac cos(2πfct) + m(t)
The above signal is applied to a diode circuit shown on next page.
11
Switching modulator
Assume that Ac |m(t)|. The output voltage signal vo (t) is of the

form (assuming the diode is an ideal switch)

vi(t), Ac cos(2πfct) > 0
vo(t) =
0, Ac cos(2πfct) < 0
=[Ac cos(2πfct) + m(t)]wT (t)
12
Switching modulator
Continue from previous page:
vo(t) =[Ac cos(2πfct) + m(t)]wT (t)

1, |t| < T4
wT (t) = (in one period)
0, 4 < |t| < 2
T T
Here T = 1/fc. The Fourier series representation of wT (t):

1 2 1 1
wT (t) = + cos(ωct) − cos(3ωct) + cos(5ωct) − . . .
2 π 3 5
∞
1 2 (−1)n−1
= + cos [2π(2n − 1)fct]
2 π n=1 2n − 1
13
Switching modulator
It can be shown that, by using the Fourier series representation of wT (t)
in vo (t), in the output voltage signal vo(t),
• the desired standard AM signal is

Ac 2 Ac 4
cos(2πfct) + m(t) cos(2πfct) = 1+ m(t) cos(2πfct).
2 π 2 πAc
4
Here the amplitude sensitivity ka = πA c
, which can be adjusted by
varying Ac. However, Ac must be sufficiently large (Ac |m(t)) to
make sure that the diode is controlled by the sign of the carrier wave.
• the undesired frequency components are the even harmonics of the carrier
frequency (fc) (e.g., 2fc, 4fc, . . .), and the signal spectral centered at the
odd harmonics of fc (e.g., 3fc, 5fc, . . .). These components are cleaned
up using a band-pass filter (BPF) centered at fc with bandwidth 2B,
where B is the bandwidth of m(t). ([Note that fc B, useful here!])
14
Standard AM demodulation
• Conceptually, we can always use the coherent DSB-SC demodulator.
This is not commonly used in practice for standard AM, because this
does not serve its purpose =⇒ need to utilize the carrier inserted to the
modulated signal.
• An envelope detector is commonly-used in AM radio receivers, which
features low cost, simplicity and effectiveness.
• Recall that for standard AM signals
xAM(t) =Ac [1 + kam(t)] cos(2πfc t)

>0
Successful envelope detection requires:

– fc B: B is the bandwidth of m(t), i.e., m(t) varies slowly with
respective to the carrier
– |kam(t)| < 1 for all t (no phase reversal/envelope distortion).
15
Envelope detector for standard AM signals
1. When the input signal is positive, the diode is forward biased and the
capacitor C charges quickly to the peak value of the input.
2. When the input drops below the voltage across the capacitor C, the
diode is reversely biased. The capacitor C discharges slowly via the
resistor Ro until the next positive cycle.
3. When the input signal rises above the voltage across the capacitor C,
the diode conducts again and the process repeats itself.
16
Envelope detector
• The function of envelope detector relies on the choice of the time

constant of the RC circuit. Let the diode resistance be Rd when it
conducts and ∞ when it does not. Let the internal resistance of the
voltage source be Ri.
– For rapid charging of C when the diode conducts, the charging time
constant τ1 = (Ri + Rd)C 1/fc. In particular, it is required that
C must be charged fully in 1/4 cycle of the carrier. In practice, the
capacitor is considered to be fully charged in 5τ1. Thus.
1 1
5τ1 =⇒ C can be chosen as Farads
4fc 20fc(Ri + Rd)
17
Envelope detector
– When the diode is not conducting, the discharge constant τ2 = RoC

must be large enough to allow a slow discharge of C (on Ro) between
the positive cycles of the carrier. On the other hand, τ2 must be small
enough to follow the maximum changing rate of m(t) (i.e., its highest
frequency B).
1 1
τ2 = RoC .
fc B
The discharge should be complete in 1/2 of frequency component B.
In practice, the capacitor is considered fully discharged in 5τ2. Thus,
1 1
5τ2 =⇒ Ro can be chosen as Ohms
2B 10CB
• Typically the signal after the envelope detector is further low-pass filtered.
18
Envelope detector
In the above figure, RC = RoC.
• The output is the voltage signal taken from the resistor Ro.
• Typically the signal after the envelope detector is further low-pass filtered,
and the existing dc component will be blocked.
19
Envelope detector
The effect of the parameters in the RC circuit. In the above, RC = RoC.
20
Power efficiency of standard AM signals
Standard AM signal:
xAM(t) =Ac[1 + kam(t)] cos ωct

= Ac cos
ωct + Ackam(t) cos ωct

carrier sidebands
• The carrier power: Pc = 12 A2c

• The sideband signal power is denoted as Ps.
• The power efficiency
useful power Ps
η= = × 100%
total power Pc + Ps
21
Single-tone standard AM signals
Single-tone standard AM modulation with m(t) = cos ωmt and

modulation index μ:
ϕAM(t) = Ac[1 + μ cos ωmt] cos ωct
A2c
• The carrier component: Ac cos ωct (Power: 2 )
• The sideband signal is μAc cos ωmt cos ωct, which contains two frequency
μ2 A2c
components. Its power can be calculated as Ps = 4 (from two
sideband frequency components).
(μAc )2
Ps 4 μ2
ηTM = × 100% = A2c
=
Pc + Ps + 4(μAc )2 2 + μ2
2
22
Power efficiency of standard single-tone AM signals
The maximum value μ can take is μ = 1. Thus,

2
max μ
ηTM =
= 33.3%
2 + μ μ=1
2
• Choosing modulation index μ: a tradeoff between power efficiency and

demodulation methods
• Typical broadcast AM stations choose μ close to 1. Input message

signals are controlled using automatic gain control (AGC).
23
Standard AM vs. DSB-SC
BW Demodulation Power efficiency
DSB-SC 2B only coherent 100% (No waste)
Standard AM 2B non-coherent enabled 0<η<1

e.g. envelope detection
(“BW” denotes bandwdith; B is the message (baseband) signal

bandwidth.)
Resources in communication systems:
• Power (“precious”)
• Hardware complexity/cost
• Bandwidth (more “precious”! How to save the bandwidth?)
24
Lecture 4: SSB and VSB
Minhua Ding
Feb. 27, 2018

Agenda
• Linear modulation
– DSB-SC
– Standard AM
– Single Sideband (SSB) (today)
– Vestigial Sideband (VSB) (today)
1
Single-sideband (SSB) modulation
• DSB-SC or standard AM signals: two sidebands (LSB, USB)
• For real signals, the LSB and USB contain the same information
(symmetric). It is possible to remove one sideband for transmission,
which leads to single-sideband (SSB) modulation.
• SSB modulation reduces the bandwidth usage by half.
• SSB was designed for the minimal use of the transmission bandwidth, e.g.,
in radio, where saving transmission bandwidth is of great importance.
• Later it will be shown that, in SSB modulated signals, both the amplitude
and the phase of the carrier are varying according to the baseband
message signal.
2
SSB transmits a bandpass filtered version of the DSB-SC signal.
3
• In practice, the highly selective filter is realized using crystal resonators.
• However, to ease the requirement for the filter, the message signal must
have a gap in its spectrum centered around the origin.
• Voice signals (300 − 3400 Hz) naturally have a gap in spectrum about
600 Hz wide, which provides a transition bandwidth of 600 Hz between
the LSB and USB.
4
Time domain Representation of SSB signals (1/2)
Let the carrier be Ac cos ωct. The upper-sideband (USB) signal is

obtained by filtering the DSB-SC signal Acm(t) cos ωct with an ideal
bandpass filter

1 fc ≤ |f | ≤ fc + B
HUSB (f ) =
0 otherwise
The resulting time-domain USB signal is given by

Ac
xUSB (t) = [m(t) cos ωct − mh(t) sin ωct]
2
where mh(t) is the Hilbert transform of m(t), i.e.,

1 ∞ m(α) 1
mh(t) = dα = m(t) ∗ .
π −∞ t − α πt
5
Time domain Representation of SSB signals (2/2)
• The Hilbert transform of m(t) can be viewed as the message signal

passing through a Hilber transformer, which is a linear time-invariant
1
(LIT) system with impulse response πt .
• The transfer function of the Hilbert transformer is

1 −j = 1 · e−jπ/2 f > 0
H(f ) = F( ) = −jsgn f =
πt +j = 1 · ejπ/2 f <0
The Hilbert transformer is an ideal phase shifter.

For example, it can be shown that the Hilbert transform of cos(ωct) is
given by π
cos ωct − = sin ωct.
2
π
The Hilbert transform of sin(ωct) is given by sin ωct − 2 = − cos ωct.
6
Derivation of the time-domain USB signal (1/2)
Assuming m(t) has the Fourier transform M (f ).
m(t) ←→M (f )
mh(t) ←→Mh(f ) = F[mh(t)] = −jsgn(f )M (f )
Let Mp(f ) be the positive frequency components of m(t), which turns out
to be:
1 M (f ), f > 0
Mp(f ) = F (m(t) + jmh(t)) =
2 0, f <0
The negative frequency portion of m(t), i.e., Mn(f ), is given by

1 0, f >0
Mn(f ) = F (m(t) − jmh(t)) =
2 M (f ), f <0
7
Derivation of the time-domain USB signal (2/2)
The USB spectrum can be written as
Ac Ac
XUSB(f ) = Mp(f − fc) + Mn(f + fc)
2 2
. The inverse Fourier transform of the above is given by

Ac 1
xUSB (t) = (m(t) + jmh(t)) ej2πfct
2 2

Ac 1
+ (m(t) − jmh(t)) e−j2πfct
2 2
Ac Ac
= m(t) cos(2πfct) − mh(t) sin(2πfct)
2 2
8
(Summary) Time domain Representation of SSB signals
Let the carrier be Ac cos ωct, and let the filters be ideal bandpass filters.
• The resulting time-domain USB signal is given by

Ac
xUSB(t) = [m(t) cos ωct − mh(t) sin ωct]
2
Ac
mh(t)
= [m(t)]2 + [mh(t)]2 cos(ωct + φ(t)), where tan−1(φ(t)) =
2 m(t)
• The time-domain LSB signal is similarly given by

Ac
xLSB(t) = [m(t) cos ωct + mh(t) sin ωct]
2
Ac
= [m(t)]2 + [mh(t)]2 cos(ωct − φ(t))

2
Both amplitudes and phases are varying with m(t) (but still in a linear
fashion)! This explains why SSB is a composite modulation scheme.
9
Generation of SSB signals (Modulation)
• Selective filtering method: DSB-SC modulator + a sharp cutoff filter
• Phase-shift method (shown in the figure below)
A method of SSB signal generation (suggested by the LSB/USB signal

expression) without the use of highly selective filters.
10
• Adding the upper and lower branches, an LSB signal is obtained.
Subtracting the upper branch by the lower branch gives the USB signal.
• The −90◦ (or −π/2) phase shift of m(t) in the lower branch is obtained
using the Hilbert transformer. In practice, this cannot be implemented
exactly.
• The phase shift is typically performed using an all-pass phase shift
network. Disadvantage: it is difficult to implement a wideband −90◦
phase shift network to cover the entire bandwidth of the modulating
signal.
11
Demodulation of SSB (Method 1)
• Method 1: using a local carrier.

The local carrier is given by
vL(t) = AL cos[2π(fc + Δf )t + φ(t)]
Ac
while the USB signal is given by xUSB (t) = 2 [m(t) cos ωct − mh(t) sin ωct].
12
It can be shown that
e(t) =vL(t) · xUSB(t)

AcAL
= {m(t) cos[2πΔf t + φ(t)] + mh(t) sin[2πΔf t + φ(t)]}
4
AcAL
+ m(t) cos[2π(2fc + Δf )t + φ(t)]
4
AcAL
− mh(t) sin[2π(2fc + Δf )t + φ(t)]
4
The low-pass filter removes the frequency components around 2fc.
AcAL
vo(t) = {m(t) cos[2πΔf t + φ(t)] + mh(t) sin[2πΔf t + φ(t)]}
4
If Δf = 0, φ(t) = 0, vo(t) = Ac4AL m(t). This is coherent demodulation,

which requires the local carrier to be phase-locked to the carrier from
transmitter.
13
Ac AL
If Δf = 0, φ(t) = 0, vo (t) = 4 {m(t) cos[φ(t)] + mh(t) sin[φ(t)]}
• The first term is a time-varying attenuation of the message signal m(t).

This also happens in DSB-SC demodulation with a local carrier.
• The second term gives crosstalk and generates phase distortion.
Ac AL
If Δf = 0, φ(t) = 0, vo (t) = 4 {m(t) cos(2πΔf t) + mh(t) sin(2πΔf t)}
• The first term: the message signal m(t) beating with a low frequency
component (same as in DSB-SC demodulation with a local carrier)
• The second term gives crosstalk and generates phase distortion.
The phase distortion may be tolerated in voice communications, as human

ears are not sensitive to phase distortion. However, to transmit music, video
or data (which are sensitive to phase distortion), coherent demodulation is
mandatory.
14
Demodulation of SSB (Method 2)
• Method 2: carrier insertion at the transmitter and envelope detection at
the receiver
The transmitted signal is given by

1 1
xUSB (t) + K cos(ωct) = Acm(t) + K cos ωct − Acmh(t) sin ωct
2 2
=R(t) cos(ωct + ϕ(t))
2 2
1 Acmh(t)
R(t) = Acm(t) + K +
2 2

Acmh(t)/2
ϕ(t) = tan−1 1
2 Ac m(t) + K
1 2 2
Under the condition 2 Acm(t) + K [Acmh(t)/2] , the low-pass
filtered envelope detector output is 12 Acm(t) + K. (Any problem here?)
15
Vestigial-sideband (VSB) modulation
Motivation:
• SSB AM is bandwidth efficient. However, it is not easy to generate SSB

signals. To use highly selective filters, the message signal should not
contain low-frequency or dc component. A wideband phase shifter can
only be realized approximately.
• To overcome the drawback of SSB, a bit of bandwidth efficiency
is sacrificed by employing a sideband filter which keeps a part of
the unwanted sideband adjacent to the carrier. This is called
asymmetric sideband modulation, which is also known as vestigial
sideband modulation (VSB).
• The VSB signals are easier to generate (at the cost of a small increase
(typically 25%) of bandwidth).
16
VSB signal spectrum
Here the respective time signals are: (a) m(t), (b) ϕDSB-SC(t), (c)
ϕUSB(t) and (d) ϕVSB(t).
17
VSB modulation and demodulation
• VSB systems can be implemented by generating a DSB-SC or standard

AM signals and passing the signals through a sideband filter.
• Depending on whether a carrier tone is added at the transmitter, standard

AM or DSB-SC demodulation can be used at the receiver.
VSB in practice (TV broadcasting)
• Suitable for signals that have a strong low-frequency component such as

video signals; used in standard analog TV broadcasting.
• In TV broadcast, a large carrier component is transmitted. The message

signal is recovered using a simple envelope detector (same as the standard
AM in radio broadcasting).
18
VSB modulation
• Here the spectrum of the transmitted VSB signal is the filtered version
of the DSB-SC signal.
ΦVSB(f ) = [M (f + fc) + M (f − fc)] Hi(f )
Hi(f ): the VSB shaping filter at the transmitter
19
VSB demodulation
• Here in this system a synchronous demodulation is used, since no carrier

is added at the transmitter: e(t) = 2ϕVSB(t) cos ωct
[with spectrum: ΦVSB(f + fc) + ΦVSB (f − fc)]
• Spectrum of the output signal after the low-pass filter
M (f ) [Hi(f + fc) + Hi(f − fc)] Ho(f )
1
Choose Ho (f ) = Hi(f +fc)+H i(f −fc )
, |f | ≤ B, to recover M (f ) (m(t)!)
• The filter Ho(f ) at the receiver is also called an equalizer.
20
Relation between the filters in the transmitter and receiver of a VSB system
21
VSB demodulation
Refer to Lathi’s book, Chapter 4, Section 4.9, for further reading of the
TV broadcastng system.
22
Lecture 5: Angle modulation (Part I)
Minhua Ding
March 6, 2018
Overview of angle modulation
• Phase modulation (PM) and frequency modulation (FM)

• FM/PM bandwidth analysis
• FM signal generation
• FM detection
• Discussions
1
Agenda for today
• Phase modulation (PM) and frequency modulation (FM)

• FM/PM bandwidth analysis
• Narrow-band FM (NBFM) and narrow-band PM (NBPM)
2
Generalized angle
• A generalized angle-modulated sinusoid
x(t) = Ac cos(ωct + φ(t)) = Ac cos θ(t) = Ac Re(ejθ(t))

θ(t)
– θ(t): generalized angle containing message information m(t)

∗ explaining the name: angle or exponential modulation
∗ nonlinear relationship between m(t) and x(t)
– φ(t): the time-varying phase
• For a regular sinusoid signal, the angle is θ(t) = ωct + θ0, where ωc, θ0
are constants.
Recall: Euler’s formula: ejθ = cos θ + j sin θ
3
Instantaneous frequency
• Instantaneous frequency and instantaneous angular frequency:

dθ(t) dφ(t)
ωi(t) = = ωc +
dt dt
1 dθ(t) 1 dφ(t)
fi(t) = = fc +
2π dt 2π dt
Correspondingly:
t t
θ(t) = ωi(α)dα = 2π fi(α)dα
−∞ −∞
4
Phase modulation (PM)
In PM, the phase φ(t) is varied linearly with m(t): φ(t) = kpm(t)
θ(t) = ωct + kpm(t)
where kp > 0 is a phase modulation constant and ωc is also a constant.

Since m(t) is a voltage signal, the unit for kp is radians/volt.
The resulting PM wave is:
xPM(t) = Ac cos[ωct + kpm(t)]
The instantaneous angular frequency in PM:

dθ(t) dm(t)
ωi(t) = = ωc + kp
dt dt
In general, in PM: θ(t) = ωc t + kp m(t) + φ0 . Here φ0 = 0 without loss of generality. This is
similarly assumed for frequency modulation (next page).
5
Frequency modulation (FM)
In FM, the instantaneous angular frequency ωi(t) is varied linearly with
m(t):
ωi(t) = ωc + kf m(t)
where ωc is a constant and kf > 0 is a frequency modulation constant in

radians/second/volt. The instantaneous angle is: (???)
t t
θ(t) = [ωc + kf m(α)]dα = ωct + kf m(α)dα
−∞ −∞
The resulting FM wave is:

t
xFM (t) = Ac cos θ(t) = Ac cos ωct + kf m(α)dα
−∞
Since ωi (t) and fi(t) differ only in 2π , in FM, fi (t) is also linear in m(t): hence the name FM. We
t
shall also assume −∞ m(α)dα do not diverge when t → ∞.
6
PM and FM waves
7
Relationship between PM and FM
ωi(t) φ(t) θ(t)

PM ωc + kp dm(t)
dt kpm(t) ωct + kpm(t)
t t
FM ωc + kf m(t) kf −∞
m(α)dα ωct + kf −∞
m(α)dα
• the amplitude of PM and FM waves is constant

• the average power is 12 A2c
• both PM and FM have time-varying phase and frequency
• modulated signal is not trivially showing the message information
• zero crossings are not periodic as in standard AM
8
Relationship between PM and FM
xPM (t) =Ac cos[ωct + kpm(t)]
t
xFM (t) =Ac cos ωct + kf m(α)dα
−∞
dm(t)
• Phase modulation using m(t) ⇐⇒ Freq. modulation using
t dt
• Freq. modulation using m(t) ⇐⇒ Phase modulation using −∞
m(α)dα

න ݉ሺߙሻ݀ߙ ‫ݔ‬ிெ ሺ‫ݐ‬ሻ
݉ሺ‫ݐ‬ሻ Phase
ю modulator

݀݉ሺ‫ݐ‬ሻ
݉ሺ‫ݐ‬ሻ ݀ ‫ݔ‬௉ெ ሺ‫ݐ‬ሻ
݀‫ݐ‬ Frequency
݀‫ݐ‬ modulator

9
Why using angle modulation?
• Superior to amplitude modulation in terms of discrimination against noise

(at a cost of bandwidth)
• Immune to channel and circuit non-linearity
Note:
• Being a constant-envelope signal, the average power of an angle-

modulated wave is not affected by the modulating base-band signal.
• Same convention:
– message signal m(t) (bandlimited to B Hz)
– carrier frequency
fc B
10
Bandwidth analysis of angle-modulated signals
Assumptions
∞
• Assumption: m(t) has no dc component, meaning −∞
m(α)dα = 0
t
Then let a(t) = −∞ m(α)dα, we have
m(t) ←→ M (f )
M (f )
a(t) ←→ A(f ) =
j2πf
The second equation is based on the Fourier transform property related

to integration.
• Based on the convention and assumption, a(t) is also bandlimited to B.
11
Bandwidth analysis of FM signals
Rewrite the FM signal using Euler’s formula:
x(t)] = Ac cos [ωct + φ(t)]

x(t) = Re[

(t) =Acej[ωct+φ(t)] = Acejφ(t) · ejωct
where x
x2 xn
Now using the Maclaurin series e = 1 + x +
x
2! + ... + n! + ...

1 2 j n n

(t) =Ac 1 + jφ(t) − φ (t) + . . . + φ (t) + . . . ejωct
x
2! n!

1 2
x(t) =Ac cos ωct − φ(t) sin ωct − φ (t) cos ωct + . . .
2!
Implications?
12
From previous page:

1 2
x(t) =Ac cos ωct − φ(t) sin ωct − φ (t) cos ωct + . . .
2!
• φn(t) has bandwidth nB (why?) and n can be arbitrarily large

This implies that angle-modulated signals have infinite band-width,
though the modulating message signal is band-limited to B Hz!
φn(t)
• n! → 0 ⇒ Significant signal power lies in a finite bandwidth
In practice, angle-modulated waves are classified into 2 categories:
• Narrow-band
• Wideband
13
Narrow-band angle-modulated signals
If |φ(t)| 1, then only the first 2 terms of the infinite series are
significant.
x(t) ≈Ac [cos ωct − φ(t) sin ωct]
• The above expression is similar to that of the amplitude modulation.

Therefore, the band-width of a narrow-band angle-modulated wave is
approximately equal to that of an AM wave.
For example, if φ(t) is bandlimited to B Hz, the bandwidth of a narrow-

band angle-modulated wave is approximately 2B Hz.
14
Narrow-band FM (NBFM) signals
t
For FM, φ(t) = kf −∞ m(α)dα = kf a(t). For the case φ(t) 1,
t
either kf or −∞ m(α)dα is small, or both of these two items are small. In
this case,
xFM(t) ≈Ac [cos ωct − kf a(t) sin ωct] .
If m(t) = Am cos(ωmt), i.e., a single-tone sinusoid signal, then
xFM(t) ≈ Ac [cos ωct − β sin(ωmt) sin ωct]
where
kf Am
β= .
ωm
15
The narrow-band FM expression (highlighted in red on the previous
page) gives rise to the following NBFM signal generation.
This is known as Armstrong’s method of FM signal generation.
16
Narrow-band PM (NBPM) signals
For PM, φ(t) = kpm(t). For narrow-band PM, |φ(t)| = kp|m(t)| 1, where
either kp or |m(t)| or both are very small. xPM(t) ≈ Ac [cos ωct − kpm(t) sin ωct] .
17
NBFM and NBPM signals: a summary
• The NBFM and NBPM signals consist of an unmodulated carrier and

sidebands centered at fc.
• The band-width of NBFM and NBPM signals is approximately 2B, where
B is the band-width of m(t).
• Different from double-sideband AM (DSB-AM) signals, the sidebands of
NBFM and NBPM are phase shifted by 90◦ with respect to the carrier,
while the sidebands of DSB-AM are in phase with the carrier.
• For NBFM or NBPM, the envelope of the modulated wave is constant,
and the instantaneous frequency is time-varying. For DSB-AM waves,
the instantaneous frequency is constant, and the envelope varies with
time.
18
Distortion using Armstrong’s method
The general angle-modulated signal x(t) = Ac cos(ωct + φ(t)) has a
constant envelope. Just for simplicity of subsequent presentation, we
assume that Ac = 1 for the rest of this lecture. In practice, however,
narrow-band angle-modulated signals are generated using the following
approximation (based on the assumption that |φ(t)| 1)
x(t) ≈ x1(t) = cos(ωct) − φ(t) sin(ωct) (|φ(t)| 1)

x1(t) = 1 + φ2(t) cos(ωct + Δ(t))

envelope phase
where Δ(t) = tan−1 φ(t)
Clearly, Armstrong’s method or the narrow-band approximation induces
• amplitude distortion (This is eliminated using a hard limiter.)

• frequency/phase distortion
19
Phase distortion in NBPM

x1(t) = 1 + φ2(t) cos(ωct + Δ(t))
Ideally, the phase Δ(t) of x1 (t) should be equal to φ(t) for zero phase
distortion. However, using Armstrong’s method,
3 5 7
−1 φ (t) φ (t) φ (t)
Δ(t) = tan φ(t) = φ(t) − + − + . . ., (|φ(t)| 1)
3 5 7
distortion items
For NBPM, the phase distortion is not severe since |φ(t)| 1, and
Δ(t) = tan−1 φ(t) ≈ φ(t) = kpm(t).
20
Frequency distortion in NBFM

x1(t) = 1 + φ2(t) cos(ωct + Δ(t))
For FM, φ(t) = kf a(t), dφ(t)

dt = kf m(t). For NBFM, one needs to check the
instantaneous (angular) frequency ωi(t). The ideal ωi(t) = ωc + dφ(t)
dt =
ωc + kf m(t). Using the narrow-band approximation,
dΔ(t) d tan−1 φ(t) kf m(t)

ω1,i(t) = ωc + = ωc + = ωc + .
dt dt 1 + φ2(t)
1
Using the series expansion 1+x = 1 − x + x2 − x3 + . . ., for |x| < 1, for
narrow-band FM signal instantaneous angular frequency
ω1,i(t) =ωc + kf m(t)[1 − φ2(t) + φ4(t) − . . .]
≈ωc + kf m(t)[1 − φ2(t)].
There can be considerable distortion from the component m(t)φ2(t).
21
Lecture 6: Angle modulation (Part II)
Minhua Ding
March 13, 2018

Agenda for today
• Wideband FM/PM signal bandwidth analysis and estimation

• Angle-modulated signal generation
1
Wideband angle-modulated signals
In previous lecture, the NBFM/NBPM bandwidth is shown to be

approximately 2B, where B is the bandwidth of the message signal m(t).
It is difficult to derive spectrum of a general (wideband) angle-modulated

signal.
In this case, a typical method is to study the simple but non-trivial case
where m(t) is a single-tone sinusoid
m(t) = Am cos(ωmt).
In practice, tone modulation (with one, two or three tones) is often employed
for tuning and checking specifications of equipment.
2
Wide-band FM (WBFM) using single-tone signals
Recall: A general (wide-band) FM wave is expressed as

t
xFM(t) = Ac cos(ωct + φ(t)), where φ(t) = kf m(α)dα = kf a(t)
∞
a(t)
Here let the modulating (baseband message)

t signal m(t) be given as
k A sin ω t
m(t) = Am cos(ωmt). Then φ(t) = kf m(α)dα = f mωm m .
∞
a(t)
Note that kf Am represents the maximum deviation from angular carrier

frequency ωc. The modulation index (or deviation ratio) is defined as
peak frequency deviation Δf Δω kf Am

β= = = .
tone frequency fm ωm ωm
3
WBFM using single-tone signals
• From previous page, FM tone-modulated signals can be written as
xFM(t) =Ac cos[ωct + β sin(ωmt)] = Re {

xFM(t)}
FM(t) = Acejωct · ejβ sin ωmt
x
• ejβ sin ωmt is periodic [fundamental period ω2πm = f1m ]. Its Fourier series is
given by ∞

ejβ sin ωmt = Jn(β)ejnωmt
n=−∞
π
1
where Jn(β) = ejβ sin xe−jnxdx
2π −π
Note that Jn(β) is the Bessel function of the first kind and nth order.
4
∞
From previous page, FM(t)
x = Ac n=−∞ Jn (β)e
jnωm t jωc t
e .
Correspondingly,
∞

xFM (t) = Ac Jn(β) cos [(ωc + nωm)t] .
n=−∞
• Frequency components: fc, fc ± fm, fc ± 2fm, . . . , fc ± nfm, . . .

• The amplitude of the n-th sideband at f = fc ± nfm is given by Jn(β).
• A plot of Jn(β) as a function of β for various values of n is shown on
the next page.
5
1
n=0
n=1
n=2
n=3
0.5 n=4
J (β)
n
−0.5
0 5 10 15
β
When β 1, the tone FM signal has a large carrier component and only a few
sideband frequencies of relative large amplitude. This is the NBFM case.
When β is large (WBFM), the carrier component becomes smaller, and many sideband
frequencies appear with considerable amplitudes.
6
β=0.2
1
0
β=1
1
0
β=5
1
0
β=10
1
A plot of line spectral: |Jn(β)| vs. n for different values of β
7
1
β=1
β=2
β=3
0.5 β=4
β=5
β=6
Jn(β)
−0.5
0 2 4 6 8 10
n
The above shows Jn(β) as a function of n. |Jn(β)| becomes negligible

for n > β + 2. This result suggests that the bandwidth of a tone-modulated
FM signal can be estimated using n = β + 2 sideband components. The
bandwidth of a tone-modulated FM signal is estimated as BWtone-FM ≈
2nfm = 2fm(β + 2) = 2(Δf + 2fm) Hz.
8
General WBFM signal bandwdith
The result on previous page suggests that the bandwidth of a general

WBFM signal can also be estimated using n = β + 2 sideband components.
Here β is the deviation ratio for the highest frequency component (B

Hz) of the modulating signal m(t)
Δf Δω
β= = ,
B 2πB
where Δf is the maximum deviation from fc in the instantaneous frequency.
Therefore, the bandwidth of a general WBFM signal is estimated as
BWWBFM ≈ 2nB = 2B(β + 2) = 2(Δf + 2B) Hz.
9
In practice, a commonly used rule for estimating WBFM bandwidth is
known as Carson’s rule:
BWCarson = 2B(β + 1) = 2(Δf + B) Hz.
A good result based on experiments is
BW = 2B(β + k) Hz, 1 ≤ k ≤ 2.
10
Wideband PM (WBPM) signal bandwidth
Recall that for PM, φ(t) = kpm(t), and the instantaneous angular
frequency is given by ωi(t) = ωc + dφ(t)
dt = ωc + k p
dm(t)
dt .
When m(t) = Am cos(ωmt), ωi(t) = ωc − Amkpωm sin(ωmt), where

ΔωPM = Amkpωm = 2πΔfPM is the peak angular frequency deviation, a
function of both the amplitude and frequency of the tone signal.
In general, m(t) has the highest frequency B Hz, and its the WBPM
bandwidth is estimated as
BWWBPM ≈ 2(ΔfPM + 2B) Hz,
where ΔfPM is the peak frequency deviation. Note that the WBPM signal
bandwidth depends on both the amplitude and frequency of the message
signal m(t).
11
Generation of FM/PM signals
• Key observation: Instantaneous phase or frequency changes linearly with

the modulating signal
• Devices: sensitive to frequency variations in a linear fashion
Two methods:
• Indirect method
• Direct method
12
Armstrong’s indirect method for FM generation
Two steps:
Step 1: Generate an NBFM signal

Step 2: Convert the an NBFM signal to a WBFM signal using frequency
multipliers/translator. An example is shown below.
Localoscillator
NBFMsignal
Localoscillator WBFMsignal ݂௖ᇱ
݂௖ଵ ݂௖ଵ ǡ ο݂ଵ ݂௖ଶ ൌ ݂݊௖ଵ ǡ ο݂ଶ ൌ ݊ο݂ଵ
݉ሺ‫ݐ‬ሻ ‫ݔ‬ௐ஻ிெ ሺ‫ݐ‬ሻ

NBFM Frequency BandͲpass Frequency
generator Multiplier filter translator
13
Frequency multipliers and frequency translation
Let mp denotes the maximum value of |m(t)|.
• Frequency multiplication
kf kf mp m(t)
fi1(t) =fc1 + m(t) = fc1 + ·
2π 2π
mp
Δf1
nkf mp m(t)
fi2(t) =nfi1(t) = nfc1 + ·
2π
mp
nΔf1
• Frequency translation
Why? nfc1 can be much higher than what we need.
Solution: use a frequency translator/converter to lower the carrier
frequency
14
Armstrong’s Indirect Method for FM Generation
In the above, consider A = Ac.
• Major advantage: frequency stability, easy to implement

• Disadvantage: amplitude/frequency distortion (discussed earlier)
15
Direct generation of FM signals
• Direct FM requires simply a voltage-controlled oscillator (VCO)

The oscillation frequency varies linearly with the control voltage
ωi(t) = ωc + kf m(t)
• The VCO can be realized as an LC oscillator where either the inductance

L or capacitance C is varied by an external voltage signal m(t).
• Typically, the capacitance is varied. This is realized using a varactor
diode.
• Major disadvantage: poor carrier frequency stability
16
• In Hartley or Colpitt oscillators: ωc = √1
LC0
• If C(t) = C0 − km(t),
−1/2
1 1 km(t)
ωi(t) =
=√ 1−
LC(t) LC0 C0

1 km(t) k|m(t)|
≈√ 1+ , if 1
LC0 2C0 C0
In the above, the binomial approximation (1 + x)n ≈ 1 + nx, |x| 1,

is used with n = −1/2. Thus,
1 kωc
ωi(t) ≈ ωc + kf m(t), ωc = √ , kf =
LC0 2C0
kf mp ΔC
• ΔC = kmp =⇒ Δf = 2π = 2C0 · fc, where mp is the maximum of
|m(t)|.
17
Example: An oscillator operating at 100 MHz has a 80 pF capacitor in

its tuning circuit. What should be the total capacitance deviation, ΔC,
of the varactor diode for the FM modulator to achieve a peak frequency
deviation of 75 kHz?
2Δf C0 2 × 75 × 103
ΔC = = × 80 pF = 0.12 pF
fc 108
18
Lecture 7: Angle modulation (Part III)
Minhua Ding
March 20, 2018

Agenda
• Demodulating angle-modulated signals

• Advantages of angle-modulated signals
• Pre-emphasis and de-emphasis filters in FM systems
1
Recap: (Last week) WBFM bandwidth analysis using
single-tone FM
For a single-tone FM signal,
xFM(t) =Ac cos(ωct + β sin(ωmt))

∞

=Ac Jn(β) cos [(ωc + nωm)t] .
n=−∞
• |Jn(β)| very small when n > β + 2; take n from 0 to β + 2, as well as

from −(β + 2) to −1 (why?); together it gives BW estimate 2(β + 2)fm.
• Important note for the why part: J−n(β) = (−1)nJn(β), for all n.
2
Some clarifications
About kf and kp:
• kf : radians/s/volt;
• kp: radians/volt
Here we assume that m(t) is a voltage (message) signal.
3
Demodulating angle-modulated signals
4
Demodulating angle-modulated signals: the big picture
For example, for FM, the message resides in the instantaneous frequency:
ωi = ωc + kf m(t)
One can extract the ωi using a slope detecting device with a transfer
function |H(f )| = a · 2πf + b (or |H(ω)| = aω + b).
For example, an ideal differentiator has such a transfer function.
5
Changes in the instantaneous frequency of the FM wave (i.e., the
message) must be reflected in the output voltage eo at the end of
demodulation.
To realize this function, one needs a sequence of device or a network

(of device), as shown later. For example, one may need a differentiator
together with other device.
6
Demodulation of angle-modulated waves
Two methods will be discussed here
• FM-to-AM conversion (for FM signal detection)

• PLL-based signal detection (for both FM and PM) (PLL: phase-locked
loop)
7
Demodulating angle-modulated signals (Method 1:
FM-to-AM conversion)
8
An angle-modulated wave has a constant envelope. However, when going
through the transmission media, the waves undergo various distortions (e.g.,
multi-path fading in wireless communications). As a result, the received
signal no longer has a constant envelope.
In general, the received angle-modulated signal r(t) can be written as
r(t) = A(t) cos θ(t) = A(t) cos(ωct + φ(t)).

θ(t)
Before demodulation, it is desirable to eliminate the envelope variation by a

hard-limiter.
The hard-limiter works as follows:

+1, cos θ(t) > 0
vo (t) = sgn[r(t)] =
−1, cos θ(t) < 0
9
‫ݒ‬௢
‫ݎ‬ሺ‫ݐ‬ሻ Hard ‫ݒ‬௢ ሺ‫ݐ‬ሻ BandͲpass ‫ݔ‬ሺ‫ݐ‬ሻ +1
‫ݎ‬
limiter filter
Ͳ1
‫ݎ‬ሺ‫ݐ‬ሻ
vo
t
vo
+1
‫װ‬
ߨ ͵ߨ ͷߨ

ʹ ʹ ʹ
Ͳ1
vo as a function of θ is a periodic square wave with its Fourier series:

4 1 1
vo (θ) = cos θ − cos 3θ + cos 5θ + . . .
π 3 5
10
• θ(t) = ωct + φ(t) =⇒ vo as a function of t

4 1
vo(t) = vo(θ(t)) = cos[ωct + φ(t)] − cos 3[ωct + φ(t)]
π 3

1
+ cos 5[ωct + φ(t)] + . . .
5
• After the bandpass filter (centered at fc) (see the first figure on the
previous page)
4
x(t) = cos[ωct + φ(t)].
π
Comments: Certainly, one can adjust the amplitude to the desired level.
From now on, we simply assume that after the hard-limiter and band-pass
filter, the angle-modulated signal is
x(t) =Ac cos[ωct + φ(t)].
11
FM-to-AM conversion
݀‫ݔ‬ிெ ሺ‫ݐ‬ሻ
‫ݔ‬ிெ ሺ‫ݐ‬ሻ
݀‫ݐ‬ ‫ܣ‬௖ ሾ߱௖ ൅ ݇௙ ݉ሺ‫ݐ‬ሻሿ
HardͲlimiter
݀ Envelope DC ‫ܣ‬௖ ݇௙ ݉ሺ‫ݐ‬ሻ
Followedby
݀‫ݐ‬ detector blocking
aBPF
• This system is for FM signal detection.

t
• Here φ(t) = kf a(t), a(t) = −∞ m(α)dα. xFM(t) = Ac cos[ωct + kf a(t)]
• After the ideal differentiator,
dxFM(t)
=Ac[ωc + kf m(t)] sin[ωct + kf a(t) + π]
dt
12
FM-to-AM conversion
݀‫ݔ‬ிெ ሺ‫ݐ‬ሻ
‫ݔ‬ிெ ሺ‫ݐ‬ሻ
݀‫ݐ‬ ‫ܣ‬௖ ሾ߱௖ ൅ ݇௙ ݉ሺ‫ݐ‬ሻሿ
HardͲlimiter
݀ Envelope DC ‫ܣ‬௖ ݇௙ ݉ሺ‫ݐ‬ሻ
Followedby
݀‫ݐ‬ detector blocking
aBPF
• In FM, typically Δω = kf mp < ωc, where mp is the maximum value of

|m(t)|.
• Δω = kf mp < ωc =⇒ ωc + kf m(t) > 0, for all t.
dxFM(t)
=Ac[ωc + kf m(t)] sin[ωct + kf a(t) + π]
dt
>0
This is important, since for subsequent envelope detection, the envelope

has to be positive.
13
Demodulating angle-modulated signals (Method 2:
PLL-based detection)
14
Detecting angle-modulated signals using PLL
‫ݏ‬ሺ‫ݐ‬ሻ LowͲpass ݁ሺ‫ݐ‬ሻ Output
Input LPF2
filter
(LPF)
‫ݒ‬௅ ሺ‫ݐ‬ሻ
VCO
• A phase-locked loop (PLL) can be used for demodulating angle-

modulated signals.
• The VCO is the abbreviation of voltage-controlled oscillator. When the
input is zero or the control signal to the VCO is 0, i.e., e(t) = 0, the
free-tuning (angular) frequency of the VCO is ωL.
Let the input signal be an angle-modulated signal
x(t) = Ac cos(ωct + φ(t)).
15
‫ݏ‬ሺ‫ݐ‬ሻ LowͲpass ݁ሺ‫ݐ‬ሻ Output
Input LPF2
filter
(LPF)
‫ݒ‬௅ ሺ‫ݐ‬ሻ
VCO
• The instantaneous frequency of the output signal from the VCO is

ωi(t) = ωL + KLe(t), where KL is a loop gain constant.
• The output signal from the VCO is then
t
vL(t) =B sin(ωLt + KL e(τ )dτ )
−∞
t
=B sin(ωct + (ωL − ωc)t + KL e(τ )dτ )
−∞
φ1 (t)
=B sin(ωct + φ1(t))
16
s(t) = x(t)vL(t) = Ac cos(ωct + φ(t)) · B sin(ωct + φ1(t))

AcB
= {sin[2ωct + φ(t) + φ1(t)] + sin[φ1(t) − φ(t)]}
2
After the low-pass filter (often referred to as a loop filter), an error signal
AcB
e(t) = sin[φ1(t) − φ(t)]
2
is obtained and used to control the frequency of the VCO towards the
locked status. When the PLL is locked, the value of (φ1(t) − φ(t)) (phase
difference) is small enough, i.e., φ1(t) ≈ φ(t).
17
• For FM input signals, under the condition of phase-lock,
dφ1(t) dφ(t) dφ(t)

φ1(t) ≈ φ(t) =⇒ ≈ =⇒ (ωL − ωc) + KLe(t) =
dt dt dt
kf m(t) (ωL − ωc)
=⇒e(t) = −
KL KL
Therefore, at the status of phase-lock, e(t) contains the message signal
m(t).
• For PM input signals, under the condition of phase-lock,
dm(t) kp dm(t) (ωL − ωc)

(ωL − ωc) + KLe(t) = kp =⇒ e(t) = dt
− .
dt KL KL
An integrator is needed to obtain m(t) from e(t).
18
• The LPF2 in the diagram is optional. In practice, e(t) is often subject to

further filtering and amplification.
• Main advantage of using PLL: Better detection performance than other
methods when the signal-to-noise ratio (SNR) is low
19
Advantages of angle modulation
20
Advantages of using angle modulation
• Compared with amplitude modulation, the output signal-to-noise ratio

(SNR) for FM is much better. This is achieved at the expense of a larger
transmission bandwidth of the FM modulated signal.
In fact, in general, there is a trade-off between the receiver SNR
performance and the band-width.
• Compared with amplitude modulated signals, angle-modulated signals
are less susceptible to non-linearity. In another word, they are less
affected by the non-linear device (such as the high power amplifier) in a
communication link.
• Angle-modulated signals are capable of suppressing weak interference
signals.
We will explain the last two items in detail.
21
Angle modulation: Less susceptible to non-linearity (1/2)
Let a non-linear device with input and output relationship given by

y(t) = a0 + a1x(t) + a2x2(t) + . . . + anxn(t). Note that the higher order
terms starting from x2(t) up to xn(t) represent the non-linear items.
• If the input to the non-linear device is the angle-modulated signal, i.e.,

x(t) = Ac cos [ωct + φ(t)], then the output has the following form:
y(t) =c0 + c1 cos [ωct + φ(t)] + c2 cos [2(ωct + φ(t))]

+ . . . + cn cos [n(ωct + φ(t))] ,
where c0, c1, . . . , cn are coefficients that depend only on a0, a1, . . . , an.
All we want is the amplified x(t), i.e., c1 cos [ωct + φ(t)]. The advantage
here is that all the unwanted items can be cleared by using a bandpass
filter with center frequency fc and with a proper bandwidth (e.g., given
by the BW estimate).
22
Angle modulation: Less susceptible to non-linearity (2/2)
In the same setting as for the discussion on the previous page, consider
now the case for an amplitude-modulated signal, e.g., the DSB-SC signal
as the input.
• Let the input to the non-linear device be the DSB-SC signal, i.e.,
x(t) = m(t) cos(ωct). If the device is specifically given by the input-
output relation y(t) = ax(t) + bx3(t), then
y(t) =am(t) cos ωct + bm3(t) cos3(ωct)

3b b
= am(t) + m3(t) cos(ωct) + m3(t) cos(3ωct)
4 4
There are two distortion terms: 3b

4 m3
(t) cos(ωc t) and b 3
4 m (t) cos(3ωct).
While 4b m3(t) cos(3ωct) can be band-pass filtered, 3b 4 m3
(t) cos(ωct)
cannot be filtered.
23
Advantages of FM
Supressing weak interferences
Suppose interference is I cos(ωc + ω)t. The received signal r(t) is
r(t) = Ac cos ωct + I cos(ωc + ω)t
= (Ac + I cos ωt) cos ωct − I sin ωt sin ωct
= Er (t) cos [ωct + φd(t)]
−1 I sin ωt
where φd(t) = tan
Ac + I cos ωt
Weak interference (I Ac) =⇒ φd(t) ≈ AIc sin ωt. The output yd(t) of an
ideal FM demodulator with input r(t) gives
Iω
yd(t) = cos ωt
Ac
(inversely proportional to Ac)

24
Dealing with noise amplification in FM
demodulation: Pre-emphasis and de-emphasis filters
25
FM and noise
White noise: random thermal motion of electrons in a resistor creates a

voltage at terminals. The power spectral density of noise is flat over a wide
range.
On the other hand, for some device in the FM demodulator, the transfer
function satisfies
|H(ω)| = aω + b.
This device (filter) increases noise at higher frequencies.
26
FM and noise
• Preemphasis filter (Hp(ω)):

boosting the weaker high frequency components of m(t)
• Deemphasis filter (Hd(ω)):

Deemphasizing (attenuating) the high frequency component and
restoring m(t)
Ideally, in the frequency band of our interest
Hp(ω) · Hd(ω) = 1
27
FM and noise
28
FM and Noise
• Is preemphasis and deemphasis filtering a perfect solution?
The peak amplitude of the signal entering the FM demodulator is

changed, potentially causing an increase in FM transmission bandwidth.
But in practice this bandwidth increase is very small.
29
Communication Engineering I
EC-3612
Chathuranga Weeraddana
03 April 2018
1/22
Lecture 8: Digital Communication Systems
1/22
A Digital Communication System (DCS)
2/22
Why Digital ?
I regeneration is possible
I transmission lines/circuits have non-ideal frequency transfer
function
I unwanted electrical noise or other interfering signals
I regenerative repeaters → original pulse is reborn
3/22
Why Digital ?
I less subject to distortions and interference

I digital circuits → just two states
I analog circuits → Not two states, infinite possibilities
I once analog signal distorted → distortion cannot be removed

by amplification
I regenerative repeaters → original pulse is reborn

I more reliable and at the same time less expensive circuits
I more flexible operation, e.g., microprocessors, digital

switching, VLSI
4/22
Why Digital ?
I combining digital signals using time-division multiplexing is

simpler than combining analog signals using frequency-division
multiplexing
I different types of traffic (e.g., data, telephone, television) can

be treated identically, A Bit is a Bit
I techniques, for compete against interference, jamming,

privacy, etc., can be integrated
5/22
Cost Associated with Digital ?
I very signal processing intensive
I significant share of resources to the task of

Synchronization
I signal-to-noise ration below a threshold → quality drops

abruptly
6/22
7/22
Digital Communication System
I information source
I speech
I music
I moving pictures
I computer data
I not necessarily binary
8/22
8/22
I formatting + source encoding
I insure that message is compatible with digital processing
I when data compression (source encoding), in addition to

formatting is used → source coding
I i.e., efficient use of the channel
I e.g., Huffman codes, Jpeg, Mpeg
I message symbols (mi , i = 1, 2, . . . , M ) constitute a finite

alphabet
I bit streams
9/22
I formatting
I character coding
I sampling
I quantization
I pulse code modulation
I source encoding
I predictive coding
I block coding
I variable length coding
I synthesis/analysis coding
I lossless compression
I lossy compression
10/22
10/22
I channel encoder
I errors due to noise, fading of signals, interference
I add redundant bits to message symbols to compete with errors
I i.e., message symbols are transformed to channel symbols;
mi −→ ui , i = 1, . . . M
I bit streams
11/22
I channel coding (waveform)
I M-ary signalling
I antipodal
I orthogonal
I Trellis-coded modulation
I channel coding (structured sequences)
I block coding
I convolutional coding
I turbo coding
12/22
12/22
I pulse modulation (baseband signalling)

I channel symbols are transformed to baseband waveforms, i.e.,
ui −→ gi (t), i = 1, . . . M
I base band → spectrum ≈ 0−few MHz
I include filtering for minimizing transmission bandwidth
I when applied to binary symbols, result is called

pulse-code-modulation (PCM) waveform
I when applied to non-binary symbols, result is called M-ary

pulse-modulation waveform
13/22
I pulse modulation (applied to binary symbols), i.e., PCM
waveforms
I NRZ (non-return-to-zero)
I RZ (return-to-zero)
I phase encoded
I multilevel binary
I pulse modulation (applied to non-binary symbols), i.e., M-ary

pulse modulation waveforms
I pulse-amplitude-modulation (PAM)
I pulse-position-modulation (PPM)
I pulse-duration-modulation (PDM)
14/22
14/22
I bandpass modulation (bandpass signalling)

I used whenever the transmission medium will not support
prorogation of pulse-like waveform
I baseband waveforms are transformed to bandpass waveforms,

i.e.,
gi (t) −→ si (t), i = 1, . . . M
I baseband wave gi (t) is frequency translated by a carrier
wave, whose frequency is much larger than the spectral
content of gi (t)
15/22
I bandpass signalling (coherent)
I PSK (phase shift keying)

I FSK (frequency shift keying)
I ASK (amplitude shift keying)
I continuous phase modulation
I hybrid versions
I bandpass signalling (noncoherent)
I DFSK (differential phase shift keying)

I FSK (frequency shift keying)
I ASK (amplitude shift keying)
I continuous phase modulation
I hybrid versions
16/22
16/22
I channel
I channel characteristics can be described by channel impulse
response hc (t)
I at various points along the signal rout, additive random

noise distort the signal
I i.e., the received signal r(t) is a corrupted version of si (t),

mathematically
r(t) = si (t) ∗ hc (t) + n(t), i = 1, . . . M
I here ∗ denotes the convolution operation
17/22
17/22
I demodulation and sampling
I demodulator provides frequency down conversion for r(t)
I there can be filters to eliminate unwanted high frequency

components
I there can be filters to shape the received wave forms for better
performance gains
I there can be equalizers to remove or diminish any signal

distortions caused by non-ideal hc (t)
I after these, processes, r(t) is restored to an baseband pulse

z(t) in preparation for sample
I sampling process transforms z(t) to a sample z(T )
18/22
18/22
I detection
I decides what could be the channel symbol transmitted
I the sampled value z(T ) is used to make and estimate ûi of the
channel symbol ui
19/22
19/22
I channel decode
I invoke error control coding mechanisms (together with its the
redundant bits) to detect errors in ûi
I if error control coding mechanisms has means for correct the

detected errors, correct those and remove redundant bits to
yield the corresponding message symbol m̂i
I if error control coding has No means for correct the detected

errors, invoke an ARQ (automatic-repeat-on-request)
mechanism so that the corrupted message symbol is
retransmitted
20/22
20/22
I synchronization
I involve estimation of both time and frequency
I coherent systems → sync the frequency reference with the

carrier in both frequency and phase
I noncoherent systems → sync the frequency reference with the

carrier in frequency
I time synchronization mean symbol or bit synchronization
I detector needs to know when to start and end the process of

symbol or bit detection
I frame synchronization to reconstruct the message symbols
21/22
22/22
EC-3612
10 April 2018
1/41
Lecture 9: Formatting
1/41
Last Week
2/41
Formatting
2/41
Formatting
3/41
Formatting
I insure that message is compatible with digital processing
I transmit formatting: source information → digital symbols
I when data compression (source encoding), in addition to

formatting is used → source coding
I e.g, character coding, sampling, quantization, PCM
4/41
Formatting-Transmission of BB Signals
5/41
Formatting-Textual Data
5/41
Formatting-Textual Data
I also known as character coding
I e.g.,
I ASCII: American Standard Code for Information Interchange

(p. 59, Sklar)
I EBCDIC: Extended Binary Coded Decimal Interchange Code
I character coding transform text into binary digits
6/41
Messages, Characters and Symbols
I textual messages consist of characters
I characters are encoded into bits
I everything is now a stream of bits
I group of bits can be combined to form new symbols from a

finite symbol set of M = 2k symbols
I M -ary system
I M = 2: binary system; M = 4: quaternary system
I k, M are design parameters of the digital communication

system
7/41
Messages, Characters and Symbols
I M = 2: binary system
I symbols and the bits are the same
I pulse modulator chooses one of the two different waveforms to

represent binary symbols
I M -ary system
I symbols and the bits are not the same
I pulse modulator chooses one of the M different waveforms to

represent M -ary symbols
8/41
Messages, Characters and Symbols,
Example
9/41
Messages, Characters and Symbols,
Example
10/41
Formatting Analog Information
10/41
Formatting Analog Information
I analog information cannot be character encoded
I the process of transforming analog information into digital

form is sampling
11/41
Sampling
I the sampling process can be implemented in several ways
I e.g., sample-and-hold
I the analog waveform can be approximately retrieved from

sampled data
I low-pass filter
I how closely a filtered sample data approximate the original

waveform?
12/41
Sampling Theorem
I a bandlimited signal having no spectral components above

fm Hz can be determined uniquely by values sampled at
uniform intervals of
1
Ts ≤ s
2fm
I the statement is also called the uniform sampling theorem
I 1/Ts = fs is called the sampling rate
I Nyquist criterion: fs ≥ 2fm ; a sufficient condition for recovery
I fs = 2fm is called the Nyquist rate
13/41
Sampling Approaches
I impulse sampling
I natural sampling
I sample-and-hold
14/41
Impulse Sampling
I consider an analog waveform x(t) with FT X(f )
I sampling of x(t) is the product xs (t) = x(t)xδ (t), where

∞
X
xδ (t) = δ(t − nTs )
n=−∞
I find spectrum Xs (f ) of xs (t)
Xs (f ) = F{xs (t)} = F{x(t)xδ (t)} (1)

= X(f ) ∗ Xδ (f ) (2)
∞
" #
1 X
= X(f ) ∗ δ(f − nfs ) (3)
Ts n=−∞
∞
1 X
= X(f − nfs ) (4)
Ts n=−∞
15/41
Impulse Sampling
16/41
Impulse Sampling
I fs = 2fm :
I analog waveform is recovered with a filter with infinitely steep

sides
I fs > 2fm :
I replications move farther apart in frequency
I easier to perform filtering
I fs < 2fm :
I replications overlap
I the phenomenon is called aliasing
17/41
Impulse Sampling
18/41
Natural Sampling
I technical details: not discussed in this lecture
I spectrum:
∞
X
Xs (f ) = cn X(f − nfs ) (5)
n=−∞
19/41
Natural Sampling
20/41
Sample-And-Hold Operation
I technical details: not discussed in this lecture
I spectrum:
∞
1 X
Xs (f ) = P (f ) X(f − nfs ) (6)
Ts n=−∞
where P is a sinc function
21/41
Sample-And-Hold Operation
I resulting spectrum is similar in appearance to the natural

sampling
I significant attenuation of the higher frequency replicates
I desired baseband experiences a nonuniform spectral gain
I fixed by postfiltering, where inverse P (f ) operation is

performed over the pastband
22/41
Aliasing
23/41
Aliasing
24/41
Antialiasing Filters
I prefiltering
25/41
Antialiasing Filters
I postfiltering
26/41
Recall.. The Formatting Block
27/41
Pulse Code Modulation (PCM)
I sampling is followed by quantization
I PCM: the base band signals obtained from the quantized

natural-sampled data by encoding each quantized sample into
a digital word
I source info. is sampled and quantized to one of L levels
I then each quantized sample is digitally encoded to into an

`-bit codeword, where ` = log2 L
28/41
29/41
I effects of increasing/decreasing quantization levels ?
I applications, where the delay is not permissible
I applications, where the delay is permissible
30/41
I PCM word size (bits per analog sample)?
I requirement: |e| ≤ pVpp
q Vpp Vpp
I |emax | = = ≈
2 2(L − 1) 2L
Vpp 1
I ≤ pVpp , i.e., L ≥
2L 2p
1
I since 2` = L, we have ` ≥ log2 bits
2p
31/41
I do not be confuse with the difference PCM and PCM

waveform
I PCM: represents a bit sequence
I PCM waveform: a particular waveform conveyance of that

sequence
32/41
Uniform and Nonuniform Quantization
I relevance to speach
I very low speech volumes

predominate
I large amplitude values are

relatively rare
I equally spaced
quantization → noise is
the same for all signal
magnitudes
33/41
I relevance to speach
I i.e., signal-to-noise (SNR)

is worse for low-level
signals than high-level
signals
I how to fix this SNR
imbalance?
I nonuniform
quantization
34/41
I fine quantization of week signals
I course quantization of strong signals
I nonuniform quantization → quantization noise can be made

proportional to signal size
35/41
I noise
I the noise for predominant weak signals is reduced
I the noise for infrequent strong signals is increased
I nonuniform quantization → can make SNR constant for for all

signals within the input range
I overall SNR is reduced
36/41
37/41
Nonuniform Quantization
38/41
Companding
I the process of compression and expansion is called

companding
I early PCM systems → smooth logarithmic compression
I today
I North America: µ−law compression characteristics
I Europe: A−law compression characteristics
39/41
µ−law Companding
loge [1 + µ(|x|/xmax )]
y = ymax sgn x
loge (1 + µ)
I µ is a positive constant
40/41
A−law Companding

A(|x|/xmax ) |x| 1
 ymax sgn x 0< ≤


y= 1 + log e A xmax A
1 + loge [A(|x|/xmax )] 1 |x|
 ymax sgn x < ≤1.


1 + loge A A xmax
I A is a positive constant
41/41
EC-3612
24 April 2018
1/24
Lecture 10: Baseband Modulation
1/24
Last Lecture
I formatting textual data
I messages, characters, symbols
I formatting analog information
I sampling
I aliasing, antialiasing filter
I quantization (linear, nonlinear)
I encoding (PCM)
2/24
Baseband Transmission
2/24
Waveforms
I binary digits are just abstractions - a way to describe message

information
I we need something physical to carry the digits
I “pulse modulate” block: abstract data is converted into

physical waveforms
3/24
Waveform Representation of Binary Digits
4/24
Waveform Representation of Binary Digits
I at the Rx → presence or absence of a pulse
I maximize the likelihood of correct decision → increase pulse

energy
I so make the pulse width as wide as possible
5/24
Waveform Representation
I pulse modulation applied to binary symbols → PCM waveform
I pulse modulation applied to nonbinary symbols → M-ary

6/24
PCM Waveform Types (Line Codes)
I nonreturn-to-zero (NRZ)
I return-to-zero (RZ)
I phase encoded
I multilevel binary
read pages 87-88 of Sklar
7/24
Why So Many PCM Waveform Types ?
I the difference in performance
I e.g.,
I DC components
I self-clocking
I error detection
I bandwidth compression
I differential encoding
I noise immunity
8/24
I DC component
I eliminate dc energy
I dc balanced signals
I less vulnerable to systems which are less sensitive for very low
frequencies
I e.g., magnetic recording systems, systems with transformer

coupling
9/24
I self-clocking
I synchronization is required in any digital system
I some PCM schemes have inherent synchronizing → enable

clock recovery
I e.g., phase encoding → bi-φ-L or Manchester codes
10/24
I error detection
I capability to detect errors without introducing additional error

detection capabilities
I e.g., duobinary encoding
11/24
I bandwidth compression (spectral attributes)
I increase the efficiency of bandwidth utilization
I information Tx rate per unit bandwidth is increased
12/24
I effect on polarity changes
I some waveforms’s polarity can be inverted without affecting

the data detection
I e.g., differential encoding
13/24
I noise immunity
I probability of bit error vs SNR curves
I e.g., NRZ outperforms unipolar RZ
14/24
Spectral Attributes of PCM Waveforms
I power spectral density (W/Hz) versus normalized bandwidth

(W T )
I W T time-bandwidth product, where T is the pulse duration
I W T = W/Rs , where Rs is the pulse rate
I units: hertz/(pulse/sec) = Hz/(bs−1 )
I W T describes how efficiently the Tx bandwidth is being

utilized
15/24
16/24
I if a waveform type needs less than 1Hz/(bs−1 ) → relatively

bandwidth efficient, e.g.,
I delay modulation
I duobinary encoding
I if a waveform type needs more than 1Hz/(bs−1 ) → relatively

bandwidth inefficient, e.g.,
I bi-φ-level (Manchester)
17/24
I bandwidth efficiency R/W
I units: bs−1 /Hz
I how much data throughput can be transmitted per Hertz
18/24
Waveform Representation
I recall
I pulse modulation applied to binary symbols → PCM waveform
I pulse modulation applied to nonbinary symbols → M-ary

19/24
M-ary Pulse-Modulation Waveform
I 3 basic ways to modulate nonbinary symbols
I pulse-amplitude modulation (PAM)
I pulse-position modulation (PPM)
I pulse-duration modulation (PDM) or PWM
I information samples are modulated without quantization →

analog pulse modulation
I information samples are quantized, yielding symbols from an

M-ary alphabet, and then modulated → (digital) M-ary pulse
modulation
20/24
M-ary PAM
I one of M allowable amplitude levels are assigned to each M
possible symbol values
21/24
M-ary PPM and PDM
I modulation is effected by delaying/advancing a pulse

occurrence by an amount that corresponds to the symbol
value
I modulation is effected by varying the pulse width by an

amount that corresponds to the symbol value
I in both cases the amplitude is kept constant
22/24
M-ary PAM Versus PCM
I PCM need more bandwidth (say R bits per second)
I M = 2k −PAM has a rate of R/k symbols per second
I i.e., a reduction in the number of symbols Tx’ed per second
I M −ary PAM as opposed to binary PCM reduces the required

bandwidth
I this reduction is achieved at the cost of higher energy for

equivalent detection performance
I equal average power in both cases → PCM needs k−times

the bandwidth of M −ary PAM
23/24
Example
I quantization levels and multilevel signalling (page 93, Sklar)
24/24
EC-3612
01 May 2018
1/30
Lecture 11: Baseband Demodulation/ Detection
1/30
Last Lecture
I Baseband transmission
I Waveform representation
I Baseband transmission
I PCM waveforms
I M-ary pulse-modulation waveform
I Spectral attributes of PCM waveforms
2/30
Baseband Demodulation/ Detection
3/30
Why Demodulator and Detector ?
I Tx waveforms are in pulse-like forms
I however, Rx waveforms are NOT in pulse-like forms
I one pulse may occupy many symbol intervals
I intersymbol interference (ISI)
I the goal of the demodulator/detector is to recover baseband

pulses with the best possible SNR, free of ISI
4/30
Signals and Noise
4/30
Error-Performance Degradation
I demodulator/detector → retrieve bits from the received

waveform as error free as possible
I two primary reasons for error-performance degradation
I effect of filtering, channel, and receiver (ISI)
I noise (thermal noise, cannot be eliminated)
I noise amplitudes are Gaussian
I two sided psd Gn (f ) = N0 /2, white
I additive, white, gaussian → AWGN
5/30
Demodulation and Detection
I binary baseband systems: within T , one of the two

waveforms, s1 (t) and s2 (t), i.e.,

s1 (t) 0 ≤ t ≤ T for binary 1
si (t) =
s2 (t) 0 ≤ t ≤ T for binary 0
I the Rx signal is
r(t) = si (t) ∗ hc (t) + n(t) i = 1, 2
I if no degradation from channel, the Rx signal is
r(t) = si (t) + n(t) i = 1, 2
6/30
7/30
I Rx filter → recover a baseband pulse with the best possible
SNR, free of ISI
I e.g., matched filter, correlator
I equalizing filter → channel induced ISI
I sampler → end of each symbol duration T , sample value z(T )

is determined
I z(T ) ∞ energy of the Rx symbol
I z(T ) ∞ 1/energy of the noise
8/30
I the o/p of step 1 yields z(T ) = ai (T ) + n0 (T ) i = 1, 2
I ai (T ) : desired signal component
I n0 (T ) : noise component
I for simplify the notation, we write
z = ai + n0
9/30
I pdf of Gaussian noise n0
n20

1
p(n0 ) = p exp − 2
2πσ02 2σ0
I thus, the conditional pdfs p(z|s1 ) and p(z|s2 ) given by
(z − a1 )2

1
p(z|s1 ) = p exp −
2πσ02 2σ02
(z − a2 )2

1
p(z|s2 ) = p exp −
2πσ02 2σ02
10/30
11/30
I in step 2, detection is performed by choosing the hypothesis
that results from the threshold measurements, i.e.,
H1
z(T ) R γ
H2
I H1 and H2 are the two possible hypotheses
I H1 : Tx Symbol = s1
I H2 : Tx Symbol = s2
I hypothesis H1 is chosen if z(T ) > γ
I hypothesis H2 is chosen if z(T ) < γ
12/30
The Basic SNR Parameter for DCSs
I analog communications systems → signal-to-noise power
(SNR = S/N )
I digital communications systems → Eb /N0
I Eb is the bit energy, Eb = (signal power)(bit time) = STb
I N0 is noise psd, N0 = (noise power)/(bandwidth) = N/W
I thus we have

Eb STb S/Rb S W
= = =
N0 N/W N/W N Rb
13/30
The Basic SNR Parameter for DCSs
I bit error probability (PB )

Vs Eb /N0
I waterfall-like shape
I higher the Eb /N0 ,

steeper the curve
I having a small
improvement in
Eb /N0 yields a huge
performance gain in
terms of PB
14/30
Detection of Binary Signals in Gaussian Noise
14/30
Minimum Probability of Error Criterion
I recall ..the decision making criterion:
H1
z(T ) R γ
H2
I how to choose γ ?
I such that the probability of error PB is minimized (min. error

criterion)
15/30
Minimum Probability of Error Criterion
I the resulting rule for minimizing PB states the following

decision making criterion:
p(z|H1 ) H1 P (H2 )
R
p(z|H2 ) H2 P (H1 )
I P (Hi ) is the prior probabilities that si , i = 1, 2 is transmitted
16/30
Maximum Likelihood (ML) Rx Structure
I if equiprobable symbols transmitted the decision making
criterion
p(z|H1 ) H1 P (H2 )
R =1
p(z|H2 ) H2 P (H1 )
becomes
H1
a1 + a2
z(T ) R = γ0
H2
2
I a1 : signal component of z(T ) when s1 (t) is transmitted
I a2 : signal component of z(T ) when s2 (t) is transmitted
17/30
ML Rx Structure, Error Probability
I probability of error: sum of the probabilities of all the ways
that an error can occur
I with previous decision criterion, we have
PB = P (decide H2 , H1 ) + P (decide H1 , H2 )
= P (decide H2 |H1 )P (s1 ) + P (decide H1 |H2 )P (H2 )
I i.e.,
I if s1 (t) is transmitted and hypothesis H2 is chosen then an

error results OR
I if s2 (t) is transmitted and hypothesis H1 is chosen then an

error results
18/30
ML Rx Structure, Error Probability
I recall: equi-probable symbols for ML, and therefore
Z ∞
PB = p(z|s2 )dz
γ0 =(a1 +a2 )/2
Z ∞
(z − a2 )2

1
= exp − dz
2σ02
p
γ0 =(a1 +a2 )/2 2πσ02
Z ∞ 2
1 u
= √ exp − du
(a1 −a2 )/2σ0 2π 2

a1 − a2
=Q
2σ0
I recall a1 > a2
I Q(·) is the complementary error function
19/30
The Matched Filter
I a liner filter designed to provide the maximum S/N for a
given Tx symbol waveform
I recall..
I a known time-limited signal s(t) plus white noise n(t) is the

input to a LTI filter, followed by a sampler
I at t = T the sampler output z(T ) with the signal component

a(T ) and noise component n0 (T ) = n0 with variance σ02
I ratio of instantaneous signal power to average noise power at

t = T is
a2 (T )

S
=
N T σ02
20/30
The Matched Filter
I note that a(t) = s(t) ∗ h(t), where h(t) is the Rx filter

response to be determined to maximize (S/N )T
I from inverse Fourier transform

Z ∞
a(t) = H(f )S(f )ej2πf t df
−∞
I output noise power

Z ∞
N0
σ02 = |H(f )|2 df
2 −∞
21/30
The Matched Filter
I by substituting above two identities in (S/N )T , we get
Z ∞
2
j2πf T

H(f )S(f )e df
S
−∞
=
N0 ∞
Z
N T |H(f )|2 df
2 −∞
Z ∞ Z ∞
2
|H(f )| df |S(f )ej2πf T |2 df
−∞ −∞
≤
N0 ∞
Z
|H(f )|2 df
2 −∞
Z ∞
2 2E
= |S(f )|2 df =
N0 −∞ N0
22/30
The Matched Filter
I the equality of the above is achieved when
H(f ) = H0 (f ) = kS ∗ (f )e−j2πf T
I take inverse FT of H0 (f ), i.e.,

−1 ks(T − t) 0≤t≤T
h0 (t) = F {H0 (f )} =
0 otherwise
I i.e., maximum (S/N )T is when the Rx filter is the mirror

image of the signal s(t) delayed by symbol duration T
23/30
The Matched Filter
summary..
I given a time-limited pulse (signal) s(t) in white noise n(t)
I then s(t) is fed into a LTI filter h(t)
I o/p signal component is a(t) and the noise component is n0 (t)
I i.e., s(t) ∗ h(t) = a(t); n(t) ∗ h(t) = n0 (t)
I variance (power) of noise is σ02
I the ratio a2 (T )/σ02 is maximized when the filter h(t) is

matched to s(t)
I i.e., when h(t) = s(T − t)
24/30
The Matched Filter
I correlation realization of the matched filter
I section 3.2.3 of sklar (reading assignment)
RT
I z(T ) = 0 r(τ )s(τ )dτ
25/30
Optimizing Error Performance
I recall..
I AWGN channel
I minimize PB
I optimumdecision threshold
in step 2 is already found →
a1 − a2
PB = Q
2σ0
I in step 1, we need a filter to maximize (a1 − a2 )/(2σ0 )
I this is achieved by maximizing (a1 − a2 )2 /σ02
I i.e., by matching the filter to s1 (t) − s2 (t)
26/30
The Required Matched Filter
I required matched filter and correlator
27/30
I suppose the filter is matched to difference signal s1 (t) − s2 (t)
I maximizing the o/p SNR as above → match filter provides a

maximum distance between two candidate outputs
28/30
I error performance of binary signaling (pages 131-133)
I unipolar signaling
I bipolar signaling
29/30
I bit error performance of unipolar and bipolar signaling
30/30
EC-3612
08 May 2018
1/32
Lecture 12: Bandpass Modulation
1/32
Last Lecture
I why demodulator and detector
I error-performance degradation
I demodulation and detection
I the basic SNR parameter for DCSs
I detection of binary signals in gaussian noise
I maximum likelihood (ML) Rx structure
I the matched filter
2/32
Bandpass Modulation
3/32
Why Modulate ?
I digital symbols are transformed into waveforms that are
compatible with the characteristics of the channel
I baseband modulation → waveforms are usually shaped pulses
I bandpass modulation → shaped pulses modulate a sinusoid

called a carrier
I carrier is converted to an EM field for propagation
I antenna size ∞ λ, i.e., smaller the frequency, larger the

antenna size
I so need to use high frequencies to have small (practical)
antenna sizes
I frequency-division multiplexing
I place signals in desired frequency bands

4/32
Digital Bandpass Modulation Techniques
I bandpass modulation is the process by which an information is
converted to a sinusoidal waveform
I three features of the sinusoid (of duration T ) are used when

embedding digital information into it
I amplitude
I frequency
I phase
I i.e., bandpass modulation → the process whereby the

amplitude, frequency, or phase of an RF carrier, or a
combination of them, is varied in accordance with the
information to be Tx
5/32
The General form of Carrier Wave
I s(t) = A(t) cos[ω0 t + φ(t)]
I ω0 = radian frequency (rad/s)
I ω0 = 2πf0 , where f0 = frequency (Hz)
I φ(t) = phase (rad)
6/32
Bandpass Modulation
I coherent systems
I Rx exploits the knowledge of carrier’s phase
I phase estimation is required
I Rx has prototypes of all possible arriving signals
I during demodulation, Rx correlates the incoming signal with

each of its prototypes
I noncoherent systems
I Rx does not utilize the carrier’s phase information
I phase estimation is not required
I price paid is increased probability of error
7/32
Bandpass Modulation
coherent systems noncoherent systems
I phase shift keying (PSK) I differential phase shift keying

(DPSK)
I frequency shift keying (FSK)
I frequency shift keying (FSK)
I amplitude shift keying (ASK)
I amplitude shift keying (ASK)
I continuous phase modulation (CPM)
I continuous phase modulation (CPM)
I hybrids
I hybrids
8/32
Vector View of Signals and Noise
I define a N -dimensional orthogonal space
I a space characterized by N linearly independent (basis)

functions {ψj (t)}
I any function is generated by linearly combining basis functions
I the following condition holds

Z T
Kj j=k
ψj (t)ψk (t) =
0 0 otherwise .
I Kj 6= 0 for all j = 1, . . . , N → orthogonal space
I Kj = 1 for all j = 1, . . . , N → orthonormal space
9/32
I orthogonal
I each ψj (t) is
independent of the
other
I each ψj (t) is Not

interfering others,
when detecting
I geometrically ψj (t) is
perpendicular to
others
10/32
I consider a arbitrary finite set {si (t)} (i = 1, . . . , M ) of

waveforms
I suppose, they are physically realizable
I suppose they are time-limited, i.e., of duration T
I the above set can be expressed as a linear combination of

N (≤ M ) orthogonal waveforms ψ1 (t), . . . , ψN (t), i.e.,
I si (t) = ai1 ψ1 (t) + ai2 ψ2 (t) + · · · + aiN ψN (t)
1
RT
I aij = Kj si (t)ψj (t)
0
11/32
I then the vector of the waveform si (t) is
I si = (ai1 , ai2 , . . . , aiN )
I then the vector of the waveform n(t) (noise with psd= N0 /2)
is
I n = (n1 , n2 , . . . , nN )
RT
I nj = (1/Kj ) n(t)ψj (t)dt
0
I n is a zero mean Gaussian with ni (i = 1, . . . , N ) are

independent
I variance of the correlator j o/p σ 2 = N0 /2 for a set of

orthonormal functions ψj (t)
12/32
I example 3.1, page 114 of Sklar
13/32
Digital Modulations
13/32
Phase Shift Keying (PSK)
I early deep-space programs, military and commercial

communications syetems
I the general analytic expression

r
2E
si (t) = cos [ω0 t + φi ] , 0 ≤ t ≤ T, i = 1, . . . , M
T
I φi will have M discrete values
2πi
I φi = , i = 1, . . . , M
M
14/32
I e.g., M = 2, called BPSK
I two waveforms
r r
2E 2E
s1 (t) = cos [ω0 t] , s2 (t) = cos [ω0 t + π] , 0≤t≤T
T T
15/32
I waveform → vectors or phasors on a polar plot
I vector length → signal amplitude
I vector direction → signal phase relative to other signals in the

set
16/32
Frequency Shift Keying (FSK)

r
2E
si (t) = cos [ωi t + φ)] , 0 ≤ t ≤ T, i = 1, . . . , M
T
I ωi will have M discrete values
I phase φ is an arbitrary constant
17/32
Frequency Shift Keying (FSK)
I e.g., M = 3, 3FSK
I three waveforms
I 3 mutually perpendicular axes → represent sinusoids with

different frequencies (not all FSK signalling is orthogonal)
I note the continuous phase above, i.e., continuous-phase FSK

(CPFSK)
I general MFSK can have abrupt changes

18/32
Amplitude Shift Keying (ASK)

r
2Ei
si (t) = cos [ω0 t + φ)] , 0 ≤ t ≤ T, i = 1, . . . , M
T
q
I 2Ei
T will have M discrete values
I phase φ is an arbitrary constant
19/32
Amplitude Shift Keying (ASK)
I e.g., M = 2, binary ASK (on-off keying)

q
I two waveforms with amplitudes 2E
T and 0
I not heavily used recently in DCSs, though it was
20/32
Amplitude Phase Keying (APK)

r
2Ei
si (t) = cos [ω0 t + φi )] , 0 ≤ t ≤ T, i = 1, . . . , M
T
q
I 2Ei
T and φi are combined to have M discrete values
21/32
Amplitude Phase Keying (APK)
I e.g., M = 8
I eight waveforms
22/32
Waveform Amplitude
I different representations..
r
2E
s(t) = cos [ωt]
T
= A cos [ωt]
√
= 2Arms cos [ωt]
√
= 2P cos [ωt]
23/32
Coherent Systems
23/32
Binary Phase Shift Keying (BPSK)
I two waveforms
r
2E
s1 (t) = cos(ω0 t + φ), 0≤t≤T
T
r
2E
s2 (t) = cos(ω0 t + φ + π)
T
r
2E
=− cos(ω0 t + φ), 0 ≤ t ≤ T
T
n(t) = zero-mean white Gaussian random process
E = signal energy per bit
T = symbol duration
24/32
Vector View of BPSK
I use the basis function

r
2
ψ1 (t) = cos ω0 t
T
I i.e.,
√ √
s1 (t) = a11 ψ1 (t) = Eψ1 (t) −→ s1 = E
√ √
s2 (t) = a21 ψ1 (t) = − Eψ1 (t) −→ s2 = − E
25/32
Vector View of BPSK
I how to realize the o/p of vectors
26/32
Vector View of MPSK
I e.g., four waveforms
r
2E 2πi
si (t) = cos ω0 t +
T 4
r
2E πi
= cos ω0 t + , 0 ≤ t ≤ T, i = 1, 2, 3, 4
T 2
27/32
Vector View of MPSK
I use the basis functions
r
2
ψ1 (t) = cos ω0 t
T
r
2
ψ2 (t) = sin ω0 t
T
I i.e., for 4PSK

√ √
s1 (t) = 0ψ1 (t) + Eψ2 (t) −→ s1 = (0, E)
√ √
s2 (t) = − Eψ1 (t) + 0ψ2 (t) −→ s2 = (− E, 0)
√ √
s3 (t) = 0ψ1 (t) − Eψ2 (t) −→ s3 = (0, − E)
√ √
s4 (t) = Eψ1 (t) + 0ψ2 (t) −→ s4 = ( E, 0)
28/32
Vector View of MPSK
I decision region for 4PSK
29/32
Vector View of FSK
I general waveform
r
2E
si (t) = cos (ωi t) , 0 ≤ t ≤ T, i = 1, . . . , M
T
I ωi+1 − ωi = nπ/T
I use the basis function

r
2
ψj (t) = cos ωj t, j = 1, . . . , N
T
30/32
Vector View of FSK
Z T
aij = si (t)ψj (t)
0
Z Tr r
2E 2
= cos(ωi t) cos(ωj t), 0 ≤ t ≤ T, i = 1, . . . , M
0 T T
√
E i=j
=
0 otherwise .
√ √
s1 (t) = Eψ1 (t) + 0ψ2 (t) + · · · + 0ψM (t) −→ s1 = ( E, 0, 0, . . . , 0)
√ √
s2 (t) = 0ψ1 (t) + Eψ2 (t) + · · · + 0ψM (t) −→ s2 = (0, E, 0, . . . , 0)
.. ......
. · ...
√ √
sM (t) = 0ψ1 (t) + 0ψ2 (t) + · · · + EψM (t) −→ sM = (0, 0, 0, . . . , E)
31/32
Coherent Detection
I Rx exploits the knowledge of carrier’s phase
I phase estimation is required
I Rx has prototypes of all possible arriving signals
I during demodulation, Rx correlates the incoming signal with

each of its prototypes
I the previous correlation receivers can be used for coherent

detection of any digital waveforms
I such a correlation detector is also called maximum likelihood

detector
32/32
Tutorial 1
Minhua Ding
Feb. 13, 2017

g(t) = C cos(ω0t + θ), ω0 = 0.
1

ω1 = ω2, ω1 = 0, ω2 = 0
2
3
Tutorial 2
Minhua Ding
Feb. 27, 2018

Question 1
A standard AM signal is given by
xAM(t) = A[1 + M cos(2πfmt)] cos(2πfc t)
where m(t) = cos(2πfmt) is the message signal, M is the modulation

index; A is the carrier amplitude, and fc fm. Show that a synchronous
or coherent detector can demodulate xAM(t) regardless of the value of M .
1
Question 2
Consider a square-law device with input vi(t) and output vo (t) such that
vo (t) = 2vi2 (t).
If the input voltage signal is given by
vi(t) = A cos(2πfct) + m(t),
where m(t) has a uniform spectrum up to the frequency B. Find the output
vo(t) and sketch the spectrum of the output signal.
2
Question 3
The power efficiency of the standard AM signal is defined. Here the

standard AM signal is given by
xAM(t) = A[1 + μ cos(2πfmt)] cos(2πfc t)
where μ is the modulation index.
(a) Find the power efficiency for μ = 0.5.

(b) Show that the maximum power efficiency is 0.33.
3
Tutorial 3
Minhua Ding
March 6, 2018
Question 1
Show that the Hilbert transform of cos(ωct) is sin(ωct).
1
Question 2
DSB-SC, Standard AM and SSB are different analog modulation

schemes. Discuss their advantages and disadvantages.
2
Question 3
A phase-shift modulator is use to generate an SSB signal x(t) , given by
x(t) = m(t) cos(ωct) − mh(t) sin(ωct)
where m(t) is the message signal with a uniformly distributed spectral

density up to the frequency ωm , and mh(t) is the Hilbert transform of
the message signal. The carrier frequency ωc ωm. Express x(t) in the
frequency domain to show that it is an upper-sideband SSB signal. Sketch
the amplitude spectrum of x(t).
3
Question 4
Consider the SSB wave
x(t) = m(t) cos(2πfct) − mh(t) sin(2πfct)
where fc is the carrier frequency, where m(t) is the message signal, and
mh(t) is its Hilbert transform. This SSB wave is applied to the following
square-law device
y(t) = x2(t).
Show that the output y(t) contains a frequency component at twice the
carrier frequency but it has a time-varying phase, which makes it impractical
to recover the carrier using squaring.
4
Tutorial 4
Minhua Ding
March 6, 2018
Question 1
Given the following angle-modulated signals:
1. s1(t) = 6 cos(400πt + 4πt2 )

2. s2(t) = cos(1500πt) cos[3 sin(50πt)] − sin(1500πt) sin[3 sin(50πt)]
For the above two signals, determine
• the average power

• the carrier frequency, and
• the instantaneous frequency.
1
Question 1: solution
1. s1(t) = 6 cos(400πt + 4πt2 )

For s1(t)
• the average power = 62/2 = 18 (watts)

• the carrier frequency = 200 (Hz)
1 1 dθ(t) 1 d[400πt + 4πt2 ]

fi (t) = ωi(t) = = = 200 + 4t
2π 2π dt 2π dt
2
1. s1(t) = 6 cos(400πt + 4πt2 )

For s2(t): s2(t) = cos[1500πt + 3 sin(50πt)]
• the average power = 12/2 = 0.5 (watts)

• the carrier frequency = 750 (Hz)
1 1 dθ(t)
fi(t) = ωi(t) =
2π 2π dt
1 d[1500πt + 3 sin(50πt)]
= = 750 + 75 cos(50πt)
2π dt
3
Question 2
If s2(t) is frequency-modulated and the modulator constant kf is 100

Hz/volt,
s2(t) = cos(1500πt) cos[3 sin(50πt)] − sin(1500πt) sin[3 sin(50πt)],
determine
• the maximum and minimum instantaneous frequencies

• the amplitude and frequency of the modulating sinusoidal signal
• the modulation index of the FM signal.
4
Question 2 (answer)
If s2(t) is frequency-modulated and the modulator constant kf is 100

Hz/volt,
s2(t) = cos(1500πt) cos[3 sin(50πt)] − sin(1500πt) sin[3 sin(50πt)],
determine
• the maximum and minimum instantaneous frequencies (fi,max = 825 Hz;

fi,min = 675 Hz)
• the amplitude and frequency of the modulating sinusoidal signal
(Amplitude Am = 0.75 volt; frequency fm = 25 Hz)
• the modulation index of the FM signal. (β = 3)
5
Tutorial 5
Minhua Ding
March 13, 2018

Question 1
A sinusoidal carrier is frequency modulated by a 4 kHz sine wave with a

3 volt peak-to- peak amplitude. The resulting FM signal has a maximum
instantaneous frequency of 107.218 MHz and minimum instantaneous
frequency of 107.196 MHz.
• What is the carrier frequency of the FM signal?

• Calculate the maximum frequency deviation of the FM signal.
• What is the modulator constant expressed in kHz/volt?
• Determine the modulation index of the FM signal.
1
A sinusoidal carrier is frequency modulated by a 4 kHz sine wave with a

3 volt peak-to- peak amplitude. The resulting FM signal has a maximum
instantaneous frequency of 107.218 MHz and minimum instantaneous
frequency of 107.196 MHz.
• What is the carrier frequency of the FM signal? (107.207MHz)

• Calculate the maximum frequency deviation of the FM signal. (11k Hz)
• What is the modulator constant expressed in kHz/volt? (11/1.5 kHz/volt
= 7.333 kHz/volt)
• Determine the modulation index of the FM signal. (β = 2.75)
2
Question 2
Consider frequency modulation (FM) using a sinusoid, i.e., FM tone

modulation. Let m(t) = 2 cos 2000πt. Assuming the carrier frequency is fc
(or carrier angular frequency ωc). Let kf = 2000π radians/s/volt.
1. m(t) = 2 cos(2000πt) = 2 cos(2πBt). Find the value of B here.

2. Let mp be the maximum value of m(t). Find the value of mp here.
k m
3. The frequency deviation of an FM signal is given by Δf = f2π p . Find
Δf here.
4. The deviation ratio (or modulation index in this case) is defined as
β = ΔfB . Find the value of the deviation radio β.
5. In the previous steps, we have obtained Δf and B. According to Carson’s
rule, the bandwidth of an FM signal is estimated as BFM ≈ 2(Δf + B).
Find BFM here for the above FM signal.
3
Tutorial 6
Minhua Ding
March 20, 2018

Question 1
In an FM modulation system, the modulating (baseband) signal m(t) =

2 cos(8000πt) and the carrier wave is given by c(t) = Ac cos(2πfct). Denote
the resulting FM signal as s(t). It is known that the FM wave s(t) has an
instantaneous frequency range given by [105.5, 105.65] MHz.
1. Find the carrier frequency fc.

2. Calculate the frequency deviation Δf (in Hz) of the FM signal.
3. Write down the FM signal, assuming that the modulator constant kf is
given in radians/s/volt.
t
s(t) = Ac cos 2πfct + kf m(α)dα
0
4. Find the expression for the instantaneous frequency.

5. Find the modulator constant kf in radians/s/volt.
6. Find the modulation index.
1
Question 2
Consider a narrowband FM signal approximately defined by s(t) =

Ac cos(2πfct) − βAc sin(2πfc t) sin(2πfmt).
a. Determine the envelope of this modulated signal.

b. What is the ratio of the maximum to the minimum value of this envelope?
Plot this ratio versus β, assuming that β is restricted to the interval
between 0 ≤ β ≤ 0.3.
c. Determine the average power of the narrow-band FM signal.
2
Solution
3
Question 1
1. fc = 105.575 MHz.
2. The frequency deviation Δf = 75 kHz = 75 × 103 Hz

sin(8000πt)
3. s(t) = Ac cos 2πfct + kf 4000π
4. Find the expression for the instantaneous frequency.
kf
fi(t) = fc + cos(8000πt)
π
5. kf = 75π × 103 radians/s/volt.

6. The modulation index β = 75
4 = 18.75.
4
Question 2
Consider a narrowband FM signal approximately defined by s(t) =

Ac cos(2πfct) − βAc sin(2πfc t) sin(2πfmt).

a. s(t) = Ac 1 + (β sin(2πfmt))2 cos(2πfct + φ(t)), where tan φ(t) =
β sin(2πfmt). The envelope is given by

Ac 1 + (β sin(2πfmt))2.
b. What is the ratio of the maximum to the minimum value of this envelope?
Plot this ratio versus β, assuming that β is restricted to the interval
between 0 ≤ β ≤ 0.3.

Ac 1 + (β)2
ratio = = 1 + β 2.
Ac
5
c. Determine the average power of the narrow-band FM signal.
s(t) =Ac cos(2πfct) − βAc sin(2πfct) sin(2πfmt)

1 1
=Ac cos(2πfct) − βAc [cos(2π(fc − fm)t] + βAc[cos(2π(fc + fm)t]
2 2
Clearly, s(t) is a sum of three cosine waves with 3 different frequencies.

The power of s(t) is the sum of individual powers of the 3 cosine waves
of different frequencies.
A2c (βAc/2)2 (βAc/2)2 A2c

P = + + = (2 + β 2)
2 2 2 4
In the above, we have used
1
sin α sin β = [cos(α − β) − cos(α + β)]
2
6
Tutorial 8 & 9
1. The information in an analog waveform, with maximum frequency fm = 3kHz, is
to be transmitted over an M -ary PAM system, where the number of pulse levels
is M = 16. The quantization distortion is specified not to exceed ±1% of the
peak-to-peak analog signal.
(a) What is the minimum number of bits/sample, or bits/PCM word that should
be used in digitizing the analog waveform?
(b) What is the minimum required sampling rate, and what is the resulting bit
transmission rate?
(c) What is the PAM pulse or symbol transmission rate?
(d) If the transmission bandwidth (including filtering) equals 12kHz, determine
the bandwidth efficiency for this system.
2. You want to transmit the word ”HOW” using an 8-ary system.
(a) Encode the word ”HOW” into a sequence of bits, using 7-bit ASCII coding
followed by an eighth bit for error detection, per character. The eighth bit is
chosen so that the number of ones in the 8 bits is an even number. How many
total bits are there in the message?
(b) Partition the bit stream into k = 3 bit segments. Represent each of the 3-bit
segments as an octal number (symbol). How many octal symbols are there in
the message?
(c) If the system were designed with 16-ary modulation, how many symbols would
be used to represent the word ”HOW”?
(d) If the system were designed with 256-ary modulation, how many symbols would
be used to represent the word ”HOW”?
3. We want to transmit 800 characters/s, where each character is represented by its

7-bit ASCII codeword, followed by an eighth bit for error detection, per character,
as in problem 2. A multilevel PAM waveform with M = 16 levels is used.
(a) What is the effective transmitted bit rate?

(b) What is the symbol rate?
4. We wish to transmit a 100-character alphanumeric message in 2 s, using 7-bit ASCII

coding, followed by an eighth bit for error detection, per character, as in problem
1. A multilevel PAM waveform with M = 32 levels is used.
(a) Calculate the effective transmitted bit rate and the symbol rate.
(b) Repeat part (a) for 16-level PAM, eight-level PAM, four-level PAM, and PCM
(binary) waveforms.
5. Given an analog waveform x(t) that has been sampled at its Nyquist rate, fs , using
natural sampling, prove that a waveform (proportional to the original waveform)
can be recovered from the samples, using the recovery technique shown in Figure 1.
Figure 1: see Problem 5
The parameter mfs , is the frequency of the local oscillator, where m is an integer.
The LPF response is (
1 |f | ≤ fs /2
H(f ) =
0 otherwise .
Hint: Recall that the natural sampled waveform xs (t) is of the form
∞
cn ej2πnfs t ,
X
xs (t) = x(t)
−∞
where cn is a real number such that cn = c−n .
6. An analog signal is sampled at its Nyquist rate fs = 1/Ts , and quantised using L
quantization levels. The derived digital signal is then transmitted on some channel.
(a) Show that the time duration, T of one bit of the transmitted binary encoded
signal must satisfy T ≤ Ts / log2 L.
(b) When is the equality sign valid?
7. Determine the number of quantization levels that are implied if the number of bits
per sample in a given PCM code is
(a) 5
(b) 8
(c) x.
8. Determine the minimum sampling rate necessary to sample and perfectly reconstruct
the signal x(t) = sin(6280t)/(6280t).
9. Consider an audio signal with spectral components limited to the frequency band
300 to 3000Hz. Assume that a sampling rate of 8000 samples/s will be used to
generate a PCM signal. Assume that the ratio of peak signal power to average
quantization noise power at the output needs to be 30 dB
(a) What is the minimum number of uniform quantization levels needed, and what
is the minimum number of bits per sample needed?
(b) Calculate the system bandwidth (as specified by the main spectral lobe of the
signal) required for the detection of such a PCM signal.
10. A waveform, x(t) = 10 cos(1000t + π/3) + 20 cos(2000t + π/6) is to be uniformly

sampled for digital transmission.
(a) What is the maximum allowable time interval between sample values that will
ensure perfect signal reproduction?
(b) If we want to reproduce 1 hour of this waveform, how many sample values need
to be stored?
11. (a) A waveform that is bandlimited to 50 kHz is sampled every 10µs. Show graph-
ically that these samples uniquely characterize the waveform. (Use a sinusoidal
example for simplicity. Avoid sampling at points where the waveform is zero.)
(b) If samples are taken 30µs apart instead of 10µs, show graphically that wave-
forms other than the original can be characterized by the samples.
12. The information in an analog waveform, whose maximum frequency fm = 4000Hz,

is to be transmitted using a 16-level PAM system. The quantization distortion must
not exceed ±1% of the peak-to-peak analog signal.
(a) What is the minimum number of bits per sample or bits per PCM word that
should be used in this PAM transmission system?
(b) What is the minimum required sampling rate, and what is the resulting bit
rate?
(c) What is the 16-ary PAM symbol transmission rate?
13. AT&T first offered digital telephone transmission referred to as T1 service. With
this service, each T1 frame is partitioned into 24 channels or time slots. Each time
slot contains 8 bits (one speech sample), and there is one additional bit per frame
for alignment. The frame generation rate is 8000 frames/s, and the bandwidth used
for transmitting the composite signal is 386 kHz. Find the bandwidth efficiency
(bit/s/Hz) for this signalling scheme.
14. (a) Consider that you desire a digital transmission system, such that the quanti-
zation distortion of any audio source does not exceed ±2% of the peak-to-peak
analog signal voltage. If the audio signal bandwidth and the allowable trans-
mission bandwidth are each 4000 Hz, and sampling take place at the Nyquist
rate, what value of bandwidth efficiency (bits/s/Hz) is required?
(b) Repeat part (a) except that the audio signal bandwidth is 20 kHz (high fidelity),
yet the available transmission bandwidth is still 4000 Hz.
Tutorial 10
1. Assume that in a binary digital communication system, the signal component out
of the correlator receiver is a1 (T ) = +1V and a2 (T ) = −1V with equal probability.
If the Gaussian noise at the correlator output has unit variance, find the probability
of bit error.
2. A bipolar binary signal, si (t), is a +1 or −1V during the interval (0, T ). Additive
white Gaussian noise having two-sided power spectral density of 10−3 W/Hz is added
to the signal. If the received signal is detected with a matched filter, determine the
maximum bit rate that can be sent with a bit error probability of PB ≤ 10−3 .
3. Unipolar RZ signalling constitutes a orthogonal signalling, see Figure 3.12 (a) of the
Sklar. In particular, the unipolar RZ signalling is as follows:
s1 (t) = A 0≤t≤T for binary 1
s2 (t) = 0 0≤t≤T for binary 0
(a) Compute the bit error rate of unipolar RZ with AWGN and a correlator de-
tector assuming that a priori probabilities are equal.
(b) Assume Gaussian noise with noise-power spectral density N0 = 10−8 Watts/Hz.
Moreover, assume that the received pulses have and amplitude of 100 mV. If
the bit-error probability specification is PB = 10−3 , find the largest date rate
that can be transmitted using this system.
Tutorial 11
1. Determine whether or not s1 (t) and s2 (t) are orthogonal over the interval (−1.5T2 <
t < 1.5T2 ), where s1 (t) = cos(2πf1 t+φ1 ) and s2 (t) = cos(2πf2 t+φ2 ), and f2 = 1/T2 .
(a) f1 = f2 and φ1 = φ2
(b) f1 = f2 /3 and φ1 = φ2
(c) f1 = 2f2 and φ1 = φ2
(d) f1 = πf2 and φ1 = φ2
(e) f1 = f2 and φ1 = π/2
(f) f1 = f2 and φ1 = φ2 + π
Figure 1: see Problem 2
2. (a) Show that the three functions illustrated in Figure 3 are pairwise orthogonal
over the interval (−2, 2)
(b) Determine the value of the constant, A, that makes the set of functions in
part (a) an orthonormal set.
(c) Express the following waveform, x(t), in terms of the orthonormal set of
part (b). (
1 0≤t≤2
x(t) =
0 otherwise .
3. Express si (t), i = 1, 2, 3, in terms of φ1 (t) and φ2 (t). Determine the equivalent

signal vectors of si (t), i = 1, 2, 3 and show them in a vector diagram.
4. Determine the constant A such thatψ1(t) = exp(−|t|) andψ1(t) = 1-Aexp(−2|t|) are
orthogonal over the interval (−∞, +∞).
A discussion of the signal power
g(t) = C cos(ω0t + θ), ω0 = 0.
26
T /2 T /2
1 2 1
Pg = lim g (t)dt = lim C 2 cos2(ω0t + θ)dt
T →∞ T −T /2 T →∞ T −T /2
T /2
1 C2
= lim [1 + cos(2ω0t + 2θ)] dt
T →∞ T −T /2 2
2 T /2 2 T /2
C C
= lim 1dt + lim cos(2ω0t + 2θ)dt
T →∞ 2T −T /2 T →∞ 2T −T /2

C 2 /2 finite
→0
=C 2/2
27

ω1 = ω2, ω1 = 0, ω2 = 0
28
g(t) = C1 cos(ω1t + θ1) + C2 cos(ω2t + θ2), ω1 = ω2, ω1 = 0, ω2 = 0
T /2 T /2
1 2 1
Pg = lim g (t)dt = lim C12 cos2(ω1t + θ1)dt
T →∞ T −T /2 T →∞ T −T /2
T /2
1
+ lim C22 cos2(ω2t + θ2)dt
T →∞ T −T /2
T /2
2C1C2
+ lim cos(ω1t + θ1) cos(ω2t + θ2)dt
T →∞ T
−T /2
finite
→0
C12 C22
= +
2 2
29
30
Assume that θ1 = θ2 in general. It can be shown that
C12 + C22 + 2C1C2 cos(θ1 − θ2)

pg = .
2
31
Use the identity: cos(α + β) = cos α cos β − sin α sin β
g(t) =C1 cos(ω1 t + θ1) + C2 cos(ω1 t + θ2)
=[C1 cos(ω1t) cos θ1 − C1 sin(ω1t) sin θ1]
+ [C2 cos(ω1 t) cos θ2 − C2 sin(ω1 t) sin θ2]
= cos(ω1t) [C1 cos θ1 + C2 cos θ2]

E
− sin(ω1t) [C1 sin θ1 + C2 sin θ2]

F

= E 2 + F 2 cos(ω1t + φ)
F
where φ is the 4-quadrant inverse of tan E . The last line shows that g(t)
is a simple sinusoid with a fixed amplitude, a single frequency and a fixed
phase. Its power can be calculated as
E2 + F 2 C12 + C22 + 2C1 C2 cos(θ1 − θ2)
pg = = .
2 2
32
Question 1
A standard AM signal is given by
xAM(t) = A[1 + M cos(2πfmt)] cos(2πfc t)
where m(t) = cos(2πfmt) is the message signal, M is the modulation

index; A is the carrier amplitude, and fc fm. Show that a synchronous
or coherent detector can demodulate xAM(t) regardless of the value of M .
Solution: A coherent detector/demodulator has a local carrier vL(t) =

cos(2πfct) synchronous to that used at the transmitter. Using a multiplier
to multiply xAM(t) with vL(t), one gets the following:
xAM(t)vL(t) = A[1 + M cos(2πfmt)] cos2(2πfct)
1
Continued from previous page:
1
xAM(t)vL(t) =A[1 + M cos(ωmt)] · [1 + cos(2ωct)]
2
A A
= [1 + M cos(ωmt)] + [1 + M cos(ωmt)] cos(2ωct)
2 2
After the LPF and the DC blocker, we get
AM
vo(t) = cos(ωmt).
2
Regardless the value of M , we get the a scaled version of m(t), which

means that our message is obtained.
See the next page for an illustration.
2
1
EC3612: Communication Engineering I (2018)
Double-sideband suppressed carrier amplitude modulation (DSB-SC AM)
Lab 1 — Prelab
I. I NSTRUCTIONS
The labs will be conducted and assessed during each lab session (within 2 hours).
• Be on time for your own lab session. It will be helpful if you can arrive at least 5 minutes before each session.
If you are late, you might not be able to finish your lab tasks in time and will lose marks.
• Be prepared for the lab. Preview the lab documents. It is solely your own responsibility to go through the
materials provided.
• Finish the prelab questions before your come to the lab session. Please hand in your answers to the questions
to the instructors during the lab sessions.
• Prelab answers are required from each student.
II. L EARNING OUTCOMES

• DSB-SC signal generation and bandwidth
• Introduction to the multiplier, VCO and tuneable LPF modules
III. P RELAB QUESTIONS

1) Go through the marking scheme form and check for the assessment components on the first page. The tasks
marked as T1-T17 are the same as in DSB-SC.pdf. Read through all the attached TIMS documents. Please
indicate here whether you read it or not (Yes or No).
2) Explain the relationship between the following two units. [2 points]

• radians/second (or rad/s)
• Hz
3) In the DSB-SC.pdf, according to T1, what is the signal to trigger the oscilloscope? [2 points]
4) If a single-tone message of 2kHz is to modulate the carrier signal of 100kHz, in the MULTIPLIER, which
coupling should one use for better results, the AC coupling or the DC coupling? [3 points]
5) Read about the frequency counter from attached TIMS documents. Please indicate here whether you read it
or not (Yes or No).
2
6) If a double-sideband suppressed-carrier (DSB-SC) (amplitude-modulated) signal is given by

xDSB−SC (t) = kA cos(μt)B cos(ωt)
where ω and μ are two (angular) frequencies (in radians/second) with ω μ. We can think of ω as the
(angular) carrier frequency, while μ as the single-tone message (angular) frequency. Here k and A are two
positive constants.
a) What is the maximum value of xDSB−SC (t)? Denote this value as x max . [2 points]
b) What is the minimum value of xDSB−SC (t)? Denote this value as x min . [2 points]
c) The difference xmax − xmin is referred to as the peak-to-peak amplitude. What is the peak-to-peak
amplitude here? [2 points]
7) Read about the overload of the multiplier from the attached TIMS documents. Please indicate here whether
you read it or not (Yes or No).
8) If the modulating message signal m(t) is a single-tone sinusoid with frequency 1 kHz and the carrier is 10
kHz, what is the bandwidth of the output DSB-SC modulated signal? [3 points]
9) Read about the VCO from the TIMS documents. Please indicate here whether you read it or not (Yes or No).
10) Read about the tuneable LPF from the TIMS documents. Please indicate here whether you read it or not (Yes
or No).
11) Explain the meaning of transition bandwidth of an LPF. [4 points]

EC3612 Communication Engineering I (2017):
Lab 1 marking scheme

Students: Please read the required components for lab 1 from this form.
Instructors: Please fill the form on this page.
Prelab 20 points Collect the prelab answers in class.

Note: Below the indices T1, T2, …, T17 are the same tasks as in the TIMS document DSB‐SC.pdf.
Group marks for each Remarks from the instructor
Components Points
task (Completion status)
Lab task 1 T1 5
Lab task 2 T2 5
Lab task 3 T3, T4 5
Lab task 4 T5 5
Lab task 5 T6 5
Lab task 6 T7, T9 5
Lab task 7 T10 5
Lab task 8 T11 5
Lab task 9 T12 5
Lab task 10 T13, T14 5
Lab task 11 T15 5
Lab task 12 T16, T17 5
Lab task 13 T18 4
Lab task 14 T19 4
Lab task 15 T20 4
Lab task 16 T21 4
Lab task 17 T22 4

Information about the group members

Students: Please fill the first 3 columns. Instructors: Please fill the last column.
The marks given to each student will be based on prelab, lab group performance and individual
involvement as shown in the form on the next page.
Prelab answers
Marks (out
Student name and ID submitted? Please Signature Index
of 100)
indicated Yes or No.
A
B
C
D
E
F
G
H
I

A record of individual performance

This page is to be filled by the students of each group. For each task, please fill the
information of the students who actually perform the task. For each student who performs a
task, please write down your index in the group and your signature. Use extra paper if needed.

Leader(s) of each task in the group
Lab task 1 T1
Lab task 2 T2
Lab task 3 T3, T4
Lab task 4 T5
Lab task 5 T6
Lab task 6 T7, T9
Lab task 7 T10
Lab task 8 T11
Lab task 9 T12
Lab task 10 T13, T14
Lab task 11 T15
Lab task 12 T16, T17
Lab task 13 T18
Lab task 14 T19
Lab task 15 T20
Lab task 16 T21
Lab task 17 T22

TIMS-301 USER MANUAL
Telecommunications Instructional Modelling System

TIMS-301 USER MANUAL
Author: Alfred Breznik and Carlo Manfredini
Issue Number 1.6 October 2004

All specifications are subject to change without notice.
Published by:
EMONA INSTRUMENTS PTY LTD

a.c.n. 001 728 276
86 Parramatta Road
Camperdown NSW 2050
Sydney AUSTRALIA
web: www.tims.com.au
telephone: + 61-2-9519-3933
fax:+ 61-2-9550-1378
Copyright (C) 1988 - 2004 Emona Instruments Pty Ltd and its related entities. All rights reserved.
No part of this publication may be reproduced, distributed or translated in any form or by any
means, including any network or Web distribution or broadcast for distance learning, or stored
in any database or in any network retrieval system, without the prior written constent of Emona
Instruments Pty Ltd.
For licensing information, please contact Emona Instruments Pty Ltd.
The TIMS logo is a registered trademark of Emona TIMS Pty Ltd.
Printed in Australia
CONTENTS
Part I TIMS INTRODUCTION 1

TIMS OVERVIEW
SYSTEM CONVENTIONS 2
Front Panel Sockets
Plug-in Modules
Labelling
Basic Modules List 3
Basic Specifications
Part II BASIC MODULES USER INSTRUCTIONS

Adder 4
Audio Oscillator 5
Buffer Amplifiers 6
Dual Analog Switch 7
Frequency and Event Counter 8
Headphone Amplifier and 3kHz LPF 9
Master Signals 10
Multiplier 12
Phase Shifter 13
Quadrature Phase Splitter 15
Scope Selector 16
Sequence Generator 17
Trunks Panel 19
Tuneable LPF 20
Twin Pulse Generator 21
Utilities Module 23
Variable DC 25
Voltage Controlled Oscillator 26
60kHz Lowpass Filter 28
TIMS INTRODUCTION
TIMS OVERVIEW
TIMS is a telecommunications modelling system. It models mathematical equations representing
electrical signals, or block diagrams representing telecommunications systems.
TIMS is primarily a hands-on rather than demonstration style teaching system, which combines
both the theoretical and practical aspects of implementing systems. We are confident that TIMS
will provide the student with a clearer understanding of the concepts behind telecommunications
theory.
Physically, TIMS is a dual rack system. The top rack accepts up to 12 Eurocard sized, compat-
ible "black boxes", or modules. The lower rack houses a number of fixed modules, as well as the
system power supply.
The modules are very simple electronic circuits, which function as basic communications build-
ing blocks. Each module, fixed or plug-in, has a specific function; functions fall into four general
categories:
Signal Generation - oscillators, etc

Signal Processing - multipliers, filters, etc
Signal Measurement - frequency counter
Digital Signal Processing - TMS320C50 based
(DSP & Advanced Modules are not included in the BASIC TIMS-301 SYSTEM)
Modules are patched together via the front panel sockets using interconnecting leads, to model
the system under investigation.
TIMS-301 User Manual 1

SYSTEM CONVENTIONS
All TIMS modules conform to the following mechanical and electrical conventions.
A - FRONT PANEL SOCKETS

Signal interconnections are made via front panel, 4mm sockets
Sockets on the LEFT HAND SIDE are for signal INPUTS.

All inputs are high impedance, typically 56k ohms.
Sockets on the RIGHT HAND SIDE are for signal OUTPUTS.

All outputs are low impedance, typically 330 ohms.
YELLOW sockets are only for ANALOG signals.

ANALOG signals are held near the TIMS standard reference level of 4V pk-pk.
RED sockets are only for DIGITAL signals.

DIGITAL signals are TTL level, 0 to 5 V.
GREEN sockets are all common, or system GROUND.
Note that input and output impedances are intentionally mismatched, so that signal connections
may be made or broken without changing signal amplitudes at module outputs.
B - PLUG-IN MODULES
Any plug-in module may be placed in any of the 12 positions of the upper rack. All modules use
the back plane bus to obtain power supply : only the DSP modules (not part of the BASIC SYS-
TEM) use the bus to transfer signals. The modules are designed so that they may be plugged-in
or removed at any time, without turning off the system power. The modules are not locked
into position and may need to be held while interconnecting leads are removed.
C - LABELLING
All modules are identified as to the function they perform.
Inputs, outputs, controls and switches are labelled so that a student who has had only a brief in-
troduction to TIMS can use the modules without needlessly referring back to this USER
MANUAL.
It should be noted that no variable controls have calibration marks. This is intentional, as the phi-
losophy behind TIMS is that students setup and adjust systems by observing and measuring sig-
nals. This assists the student in gaining a much greater understanding, feel and insight into the
operation of a communications implementation.

D - BASIC MODULE LIST
Below are listed all the BASIC SYSTEM MODULES. FIXED modules are located in the lower
rack, while PLUG-IN modules can be positioned anywhere in the upper rack.
Adder - plug-in
Audio Oscillator - plug-in
Buffer Amplifiers - fixed
Dual Analog Switch - plug-in
Frequency and Event Counter - fixed
Headphone Amplifier and 3kHz LPF - fixed
Master Signals - fixed
Multiplier - plug-in
Phase Shifter - plug-in
Quadrature Phase Splitter - plug-in
Scope Display Selector - fixed
Sequence Generator - plug-in
Trunks Panel - fixed
Tuneable LPF - plug-in
Twin Pulse Generator - plug-in
Utilities Module - plug-in
Variable DC - fixed
Voltage Controlled Oscillator - plug-in
60kHz Lowpass Filter - plug-in
BASIC SPECIFICATIONS
POWER SUPPLY
Input 120, 127, 220 or 240V AC, 47Hz to 63Hz
Output + 15V, 2.2A DC
-15V, 2.2A DC
Protection short circuit, overload, thermal
Regulation 0.2%
PHYSICAL
Case Dimensions 490(W) x 330(D) x 310(H) mm
System Weight 10kg
Plug-in Card Dimensions 160 x 100 mm
Plug-in Card Bus Connectors 64 way, 2 row, Eurocard
MODULES
Specifications for each module are listed in the following pages.

ADDER
Two analog input signals A(t) and B(t) may be added together, in adjustable proportions G and
g. The resulting sum is presented at the output.
G:GAIN CONTROL
FOR INPUT A
ANALOG INPUT
g:GAIN CONTROL
FOR INPUT B
ANALOG INPUT ANALOG OUTPUT
FRONT PANEL BLOCK DIAGRAM
USE
Care must be taken when adjusting the gains to avoid overloading the following modules. Over-
loading will not cause any damage but it means non-linear operation, which is to be avoided in
analog systems. The ADDER is capable of delivering a signal well in excess of the standard refer-
ence level, 4V pk-pk, given a standard level input.
The ADDER can also be used as a normal amplifier by using only one input and turning the gain
of the other input to minimum. It is not necessary to ground the unused input.
Note that gains G and g are negative. All inputs and outputs are DC coupled.
Gain Range 0 < G < 2;
0 < g < 2;
Bandwidth approx 1MHz
Output DC Offset < 10mV, open circuit inputs
PARAMETERS TO NOTE
maximum output level; linearity; polarity inverting; phase shift

AUDIO OSCILLATOR
The AUDIO OSCILLATOR is a low distortion tuneable frequency sinewave source with a fre-
quency range from 500Hz to 10kHz. Three outputs are provided. Two outputs are sinusoidal,
with their signals in quadrature. The third output is a digital TTL level signal.
INPHASE
SYNCHRONIZE
ANALOG
INPUT
OUTPUT
FREQUENCY
ADJUST
TTL LEVEL
OUTPUT
QUADRATURE
ANALOG
OUTPUT
USE
The frequency of each of the three outputs is the same and is varied by the front panel ∆ f con-
trol. Both the in-phase and quadrature analog output signals have fixed amplitude. Their shape
is sinusoidal, having a distortion of less than 0.1%.
The AUDIO OSCILLATOR may be synchronized to an external periodic signal by connecting

such a signal to the front panel SYNC input. A signal of about 1 volt peak is adequate for this
purpose. For synchronization to be achieved, the AUDIO OSCILLATOR must be manually tuned
to within a few percent of the frequency to which synchronization is desired.
Frequency Range 300Hz to 10kHz
Analog Output Level 4V pk-pk
Distortion < 0.1% analog outputs only
Digital Output TTL level
PARAMETERS TO NOTE
frequency range; relative phase of outputs; amplitude stability with frequency range; harmonic
content; short term stability; synchronizing characteristic.

BUFFER AMPLIFIERS
Two independent variable gain amplifiers are provided.
GAIN CONTROL
ANALOG INPUT OF
ANALOG OUTPUT
FIRST AMPLIFIER
GAIN CONTROL
ANALOG INPUT OF
ANALOG OUTPUT
SECOND AMPLIFIER
USE
These buffers may be used to amplify small signals or attenuate large signals. Each amplifier
has its own gain control on the front panel.
Care should be taken to ensure that later modules are not overloaded due to excessive gain.
Overload will not cause any damage but it means non-linear operation, which is to be avoided in
analog systems. If overload occurs, turn the gain control counter clockwise.
Bandwidth DC to approx 1MHz
Gain 0 to 10

DUAL ANALOG SWITCH
Two identical analog switches are controlled by digital, TTL level signals. The outputs of the two
switches are added internally and presented at the output of the module.
ANALOG INPUT 1
TTL CONTROL
FOR INPUT 1
TTL CONTROL
FOR INPUT 2
ANALOG INPUT 2 OUTPUT
USE
Each switch may be closed independently by a TTL HIGH at the respective control input. The
switch outputs are combined internally and are presented at the common output socket. Open
circuit voltage gain between each input and the module output is unity when the switch is
closed.
Analog Input Bandwidth > 300kHz
Maximum CONTROL clock > 100kHz
CONTROL Input Levels TTL only
Maximum Analog Input Level + 8V
PARAMETERS TO NOTE
switch On/Off ratio; linearity; switching speed; analog bandwidth; channel cross talk; DC off-set

FREQUENCY COUNTER
The TIMS counter is an 8 digit, 10MHz frequency and event counter.
9
8
1 7
3 6
4 5
1 OVERflow indication LED
2 ANALOG input:
Bandwidth 40Hz to 1 MHz
Sensitivity 250mV typically, @ 100kHz
Maximum input + 12V
3 TTL Input:
Bandwidth DC to 10MHz
Input TTL level signals only
4 TTL ENABLE may be used to gate the TTL input signal.
Specifications are same as for the TTL input.
5 Mode and Range rotary switch
Frequency counter mode Gate time selection of 0.1s, 1s or 10s with reading in kHz
Event counter mode displays number of pulses counted since the last RESET
6 RESET Push Button resets the count of the Event Counter to zero
7 kHz LED is lit when counter is in FREQUENCY COUNTER mode
8 8 digit, 7 segment display of frequency or pulse counts;
maximum display 99999999
9 COUNTS LED is lit when counter is in EVENT COUNTER mode

HEADPHONE AMPLIFIER and 3kHz LPF
The HEADPHONE AMPLIFIER is a wideband, variable gain audio amplifier which will drive stand-
ard 8ohm headphones or a speaker. An independent 3kHz LOWPASS FILTER may be switched
in before the audio amplifier, if required.
FOR SWITCHING
LPF OUTPUT TO
AMPLIFIER INPUT
LOWPASS FILTER
OUTPUT
AMPLIFIER
GAIN ADJUST
AMPLIFIER AND HEADPHONE

FILTER INPUT OUTPUT
USE
This module serves as an electro-acoustic interface between the audio signals within the system
and the user. Included within the HEADPHONE AMPLIFIER module is an independent LOW-
PASS FILTER with a 5th order elliptic characteristic. The filter’s cutoff frequency is 3kHz, stop-
band attenuation is 50dB and passband ripple is 0.2dB.
AUDIO AMPLIFIER Bandwidth < 100kHz
THD 0.2% (RL= 8ohms, P= 125mW)
Maximum Gain 20
Maximum Output Power 500mW
Output Impedance 8 ohms
LOWPASS FILTER Cutoff Frequency 3kHz

Stopband Attenuation 50dB
Passband Gain approx 1
Passband Ripple 0.2dB
PARAMETERS TO NOTE
filter corner point; filter shape; passband ripple; out-of- band attenuation; amplifier distortion

MASTER SIGNALS
Five synchronized analog and digital signals are available, ranging from 2kHz to 100kHz. The
function and frequency of each signal is indicated on the front panel.
QUADRATURE ANALOG
CARRIER SIGNAL
INPHASE ANALOG
CARRIER SIGNAL
TTL LEVEL CARRIER

SIGNAL
TTL LEVEL
ANALOG SIGNAL
USE
Signals are labelled as follows:
CARRIER signals are 100kHz, which for modelling purposes is sufficiently far from the audio
channel bandwidth of 3kHz.
The SAMPLE CLOCK of 8.3kHz, which may be used to sample bandwidth-limited (3kHz) audio
message signals.
MESSAGE provides an audio frequency signal which is synchronized to a sub-multiple of the

carrier to enable ’text-book’ like displays of simple modulation schemes to be achieved.
The five signals are derived from a master crystal oscillator resulting in low frequency drift. Their
frequencies are fixed internally. The output levels are also fixed. To vary the amplitude, the sig-
nals may be applied to the neighboring buffers.
The analog signals are sinusoidal in shape, having a distortion of less than 0.1%.
Digital signals are all standard TTL level, with rise times of better than 80nsec.

Output Frequencies 100kHz, carrier
8.333kHz, sample clock
2.083kHz, audio (carrier sub-multiple)
Output Levels 4V pk-pk, analog
TTL level, digital
Distortion < 0.1%, analog outputs only
PARAMETERS TO NOTE
short term frequency stability; relative phase of quadrature outputs; harmonic content.

MULTIPLIER
Two analog input signals X(t) and Y(t) may be multiplied together. The resulting product is
scaled by a factor of approximately 1/2 so that, with standard level inputs, later stages are not
overloaded.
INPUT COUPLING
SWITCH
ANALOG INPUT
USE
The input coupling switch may be used to remove input DC components by switching to AC
coupling. It should be noted that any DC component in the output will not be removed.
The "k" factor (a scaling parameter associated with "four quadrant" multipliers) is approximately
one half. It is defined with respect to the OUTPUT from the module and may be measured ex-
perimentally.
Bandwidth approx 1MHz
Characteristic k.X(t).Y(t)
k approx 1/2
PARAMETERS TO NOTE
linearity; k factor; carrier leak; phase response; DC off-set; performance as a squarer; frequency
response; "conversion gain" as a (de)modulator.

PHASE SHIFTER
The PHASE SHIFTER introduces a phase shift between its input and output. This phase shift is
adjustable by the user. The frequency range of operation can be selected by PCB mounted
switch.
BLOCK DIAGRAM
COARSE PHASE
ADJUST
FINE PHASE
ADJUST
180O PHASE
CHANGE
FRONT PANEL
PCB VIEW
USE
This variable PHASE SHIFTER is capable of varying the magnitude of the phase shift through
360 degrees in two steps. The 180 degree switch selects the step or region of interest; the
COARSE and FINE controls are used to then obtain the required phase shift, Φ.
If the input is COS(µt), then the output is COS(µt- Φ), where Φ lies between 0 and 180 degrees.
Although the PHASE SHIFTER will operate from a few hertz up to 1MHz it has been optimized
to operate in the neighborhood of two frequencies: around 100kHz in the HI range and around
2kHz in the LO range. A PCB mounted switch is used to select the frequency range.
The open circuit gain through the PHASE SHIFTER is essentially unity for all phases, but note
that the amount of phase shift, Φ, is a function of frequency. This is NOT a wideband phase
changer: thus all the frequency components of a complex signal’s spectra are not shifted by the
same phase.

Bandwidth < 1MHz
Frequency Range HI approx 100kHz *
LO approx 2kHz *
* For 0 to 360 degree range of phase shift. The phase shift range increases (i.e. resolution
decreases) as the input frequency increases.
Coarse approx 180 degrees shift
Fine approx 20 degrees shift
PARAMETERS TO NOTE
Variation of phase change with frequency change.

QUADRATURE PHASE SPLITTER
When the same analog signal is applied to both inputs, the two output signals will differ in phase
by 90 degrees. The phase splitter networks are wideband, typically covering the range from
200Hz to 10kHz.

TO NETWORK 1 FROM NETWORK 1

TO NETWORK 2 FROM NETWORK 2
USE
The QUADRATURE PHASE SPLITTER consists of two wideband phase shifting networks. The
networks’ phase responses vary with frequency in a complimentary manner, giving a 90 degree
phase difference between the outputs, over a wide frequency range.
In communications the most important application is the generation and demodulation of Single
Sideband by the "phasing method".
Frequency Range 200Hz to 10kHz typically
Phase Response 90 degrees between outputs, given the same input signal to both networks.
PARAMETERS TO NOTE
Phase error from 90 degrees. This may be measured directly (difficult !) or calculated from side-
band suppression measurements.

SCOPE SELECTOR
(OSCILLOSCOPE DISPLAY SELECTOR)
The OSCILLOSCOPE DISPLAY SELECTOR allows 2 of 4 different signals to be viewed simultane-

ously on a 2 channel oscilloscope. A third input labeled TRIG is ideal for connecting a trigger sig-
nal to the oscilloscope’s external trigger input.
INPUT SELECTOR
CH1 INPUT "A"

CH1 OUTPUT
CH1 INPUT "B"
CONNECT TO
TRIGGER INPUT SCOPE TRIGGER
INPUT
INPUT SELECTOR
CH2 INPUT "A"

CH2 OUTPUT
CH2 INPUT "B"
FRONT PANEL DIAGRAM
USE
Connection to the oscilloscope is via BNC sockets. Inputs are standard 4mm sockets. Although
the input sockets are YELLOW (analog), either analog or digital signals may be examined.

SEQUENCE GENERATOR
(PSEUDORANDOM SEQUENCE GENERATOR)
Using a common external clock signal, the sequence generator outputs two independent
pseudorandom sequences X and Y. A SYNC output is provided which is coincident with the
start of the sequences. The sequences may be stopped and restarted at any time via front panel
controls. Sequences X and Y are available as either standard TTL or analog level output.
RESET
ANALOG OUTPUT
PUSH BUTTON
TTL LEVEL
ANALOG OUTPUT
RESET
BEGINNING OF
SEQUENCE SYNCH
ANALOG CLOCK TTL OUTPUT
TTL CLOCK TTL OUTPUT
USE
An external clock signal must be provided to operate the SEQUENCE GENERATOR. This may
be sinusoidal or TTL: separate input sockets are used.
The sequences may be stopped at any time by either depressing the RESET button or applying
a TTL HI signal to the RESET input. To restart the sequences from the beginning, release the
RESET button or apply a TTL LO to the RESET input.
The length of the sequences may be selected by a PCB mounted dip switch. Four independent
sequence pairs are available from lengths of 25 to 211.
The sequences are selected as follows:

DIP SWITCH CODE n SEQUENCE LENGTH 2n
msb 0 0 5 32
0 1 8 256
1 0 8 256
1 1 11 2048

Input Clock Range TTL 1Hz to 1MHz
Analog < 500Hz to > 10kHz
Number of Sequences 4 pairs
Sequence Lengths 25, 28, 28, 211
Sync indicates start of sequence
PARAMETERS TO NOTE
sequence distribution; noise generation using pseudorandom sequences.

TRUNKS PANEL
The TRUNKS PANEL provides inputs and outputs to signals which are transmitted along the OP-
TIONAL TIMS BUS classroom network. The three outputs SIGNAL 1, SIGNAL 2 and SIGNAL 3
present signals from the lecturer’s master system. IN and OUT allow for signals to be respec-
tively received from and transmitted to a neighboring TIMS system.
SIGNALS 1, 2 & 3
COME FROM THE
MASTER TIMS SYSTEM
IF TRUNKS IS CONNECTED
INCOMING SIGNAL FROM ADJACENT STUDENT’S

TIMS SYSTEM, IF TRUNKS IS CONNECTED
OUT SENDS A SIGNAL TO THE NEXT STUDENT’S

TIMS SYSTEM, IF TRUNKS IS CONNECTED
FRONT PANEL
USE
Note that the TRUNKS PANEL is a module that differs from the TIMS’ front panel color code and
alignment conventions.
Though the inputs and outputs are YELLOW (analog), either analog or digital signals may be
used. Also, the signal input, OUT, which accepts a signal that is to be transmitted to a neighbor-
ing TIMS system, is on the right hand side.
BASIC SPECIFICATIONS when TIMS-TRUNKS is installed

Master Channels 3 : SIGNAL 1, SIGNAL 2 and SIGNAL 3;
Master Channel Bandwidth 700kHz (typ), ac coupled.
Local Channels 2 : IN brings the incoming signal FROM an adjacent TIMS’ OUT port.
OUT carries the outgoing signal TO the other adjacent TIMS’ IN port.
Local Channel Bandwidth 350kHz (typ), ac coupled

TUNEABLE LPF
The cutoff frequency of this LOWPASS FILTER can be varied using the TUNE control. Two
frequency ranges, WIDE and NORMAL, can be selected by a front panel switch. The GAIN
control allows signal amplitudes to be varied if required.
CLK for TLPF modules V1 to V3:

CLK/880 = f-3dB; NORMAL
CUT-OFF CLK/360 = f-3dB; WIDE
FREQUENCY CLK for TLPF modules V4:
ADJUST CLK/100 = f-3dB; NORMAL & WIDE
GAIN ADJUST
FREQUENCY
RANGE SELECT
BLOCK DIAGRAM
FRONT PANEL
USE
This lowpass filter has an elliptic filter characteristic. The stopband attenuation is typically 50dB
and passband ripple is approximately 0.5dB.
The GAIN control is used to vary the amplitude of the output signal. Care should be taken to
avoid overloading/saturation. Two frequency ranges are provided. NORMAL range provides
more precise control over the lower audio band, used for telecommunications message
channels. The WIDE range expands the filter’s range to above 10kHz. The CLK output provides
an indication of the filter’s cutoff frequency.
BASIC SPECIFICATIONS for TLPF modules V1 to V3

Filter Ranges 900 Hz < NORMAL < 5 kHz and
2.0 kHz < WIDE < 12 kHz, continuously variable over each range.
Filter Order 7th order, Elliptic
Stopband Attenuation > 50dB and Passband Ripple < 0.5dB
BASIC SPECIFICATIONS for TLPF modules V4

Filter Ranges 200 Hz < NORMAL < 5 kHz and
200 Hz < WIDE < 12 kHz, continuously variable over each range.
Filter Order 5th order, Elliptic
Stopband Attenuation > 50dB and Passband Ripple < 0.5dB
Maximum Input Voltage + 5V to -5V (TTL-level input signal is acceptable)
PARAMETERS TO NOTE
corner point; phase shift; gain range; passband ripple; out of band attenuation.

TWIN PULSE GENERATOR
(TWIN PULSE GENERATOR - VERSION 2.0)
A positive going edge applied at the CLOCK input causes a positive pulse to occur at the out-
put terminals. There are two operating modes: TWIN and SINGLE. Only TWIN mode is limited to
low frequency CLOCK inputs.
In TWIN mode, Q1 outputs the leading pulse and Q2 outputs the delayed pulse. The time be-
tween pulses Q1 and Q2 can be varied, as can the pulses’ widths.
In SINGLE mode, only Q1 outputs a positive going pulse, while Q2 outputs the inverse of Q1.
The pulse width can be varied.
ERROR LED, INDICATES

IF 2tW + tD > tCLK
BLOCK DIAGRAM
PULSE WIDTH
CONTROL
DELAYED PULSE
OUTPUT:
TTL LEVEL
DELAY TIME LEADING PULSE
CONTROL OUTPUT:
AC COUPLED
DIGITAL CLOCK TTL LEVEL OUT
FRONT PANEL
TIMING DIAGRAM
USE
A digital TTL level signal is applied to the CLK input. The GENERATOR then outputs one or two
pulses, depending upon the operating mode selected. Use the PCB mounted MODE switch to
select either SINGLE or TWIN operating mode.
TWIN MODE
TWIN mode is used when two sequential pulses are needed. Two equal width positive pulses oc-
cur as a result of each CLK signal positive edge. Pulse Q1 always occurs before pulse Q2. The
width of both pulses is controlled by the front panel WIDTH control. The DELAY control varies
the spacing between the two pulses. Note that TWIN mode will only accept CLOCK input signals
of up to 50kHz, depending upon front panel settings.
If WIDTH and DELAY have been incorrectly set, causing anomalous operation, the ERROR
LED will be lit. To eliminate the error reduce DELAY and then WIDTH - by turning counter clock-
wise.

SINGLE MODE
SINGLE mode is used to obtain a train of equal width pulses from any TTL level signal.
Equal width positive pulses occur at Q1 output as a result of each CLK signal positive edge. The
width of the pulses is controlled by the front panel WIDTH control. Q2 simultaneously outputs
the compliment of Q1. The DELAY control is not used in this mode.
Note that Q1 includes both a TTL level and an AC coupled output pulse.
TWIN MODE
Clock Frequency Range < 50kHz
Pulse WIDTH 3µs < tw < 25µs
Pulse DELAY Q2-Q1 10µs < td < 120µs
Error Indication 2tw + td > tCLK
SINGLE MODE
Clock Frequency Range < 200kHz
Pulse WIDTH 3µs < tw < 25µs

UTILITIES MODULE
The Utilities Module houses 4 independent functional blocks:
(i) A signal COMPARATOR with TTL output and CLIPPER with bipolar output, for squaring analog
waveforms. The COMPARATOR’s threshold level may be set as required by applying a DC volt-
age to the REF input. The CLIPPER’s gain may be set by adjusting DIP switches SW1 and SW2.
(ii) Precision halfwave RECTIFIER.
(iii) Simple diode and single pole, audio range, RC Lowpass Filter.
(iv) Single pole, audio range, RC Lowpass Filter.
ANALOG REFERENCE CLIPPER

INPUT BIPOLAR OUTPUT
ANALOG SIGNAL COMPARATOR

INPUT TTL OUTPUT
USE
COMPARATOR
The COMPARATOR will square any analog signal and provide a standard TTL level output. The
switching threshold level is determined by the voltage level applied to the REF input.
NOTE: For correct COMPARATOR operation, the REF input must never be left unconnected.
The REF input may be connected to GROUND, VARIABLE DC or any other signal source.
CLIPPER
The CLIPPER will amplify any analog TIMS level signal and then clip the amplitude of the ampli-
fied signal, to a fixed level of approximately + 1.8V. The clipping action is performed by stand-
ard small signal diodes.
NOTE: The REF input is NOT used by the CLIPPER.

Adjusting the gain of the CLIPPER, determines whether the clipping action is "SOFT", "MEDIUM"
or "HARD". The following table, UTIL-1, relates DIP switch settings to CLIPPER gains and clip-
ping action,
CLIPPING GAIN DIP SWITCH SETTINGS

ACTION (approx) SW1 (both a & b) SW2 (both a & b)
SOFT x0.8 ON - ON OFF - OFF
MEDIUM x8 OFF - OFF OFF - OFF
HARD x40 OFF - OFF ON - ON
- not applicable ON - ON ON - ON
Table UTIL-1: CLIPPER gain settings
DIP switches SW1 and SW2 will be found in the middle of the UTILITIES module’s circuit board.
NOTE: Both halves (bits) of each switch must be in the SAME position at all times.
COMPARATOR
Operating Range > 500kHz
TTL Output Risetime 100nsec (typ)
CLIPPER
Operating Range > 500kHz
Output Level 1.8Vpk (typ)
Adjustable Gains 3 steps; x0.8, x8 and x40 (approx)
RECTIFIER
Bandwidth DC to 500kHz (approx)
DIODE & LPF

LPF -3dB 2.8kHz (approx)
RC LPF
LPF -3dB 2.8kHz (approx)

VARIABLE DC
The VARIABLE DC module is a stable, bipolar DC source.
+ 5V
DC VOLTAGE
CONTROL
DC OUTPUT
GROUND
REFERENCE
USE
The DC voltage output varies from about -2.5V when the control is fully counter clockwise
through zero to + 2.5V when control is turned fully clockwise. If greater resolution or wider
range is required, then one of the BUFFER AMPLIFIERS can be used in conjunction with the
VARIABLE DC module.
Voltage Range + 2.5V DC
Short-term Stability < 2mV/hr
Resolution approx 20mV
Output Current < 5mA

VCO
The Voltage Controlled Oscillator module functions in two modes: either as a VOLTAGE CON-
TROLLED OSCILLATOR with analog input voltage or as an FSK GENERATOR with digital input.
Both modes have two frequency ranges of operation which are selected by a range switch. The
VCO frequency and input sensitivity can be controlled from the front panel.
FSK DATA INPUT

TTL LEVEL
- TTL LEVEL
OUTPUT
SET SENSITIVITY
OF INPUT CONTROL
VOLTAGE Vin OVERLOAD
INDICATION LED
SET CENTER BLOCK DIAGRAM
FREQUENCY
FREQUENCY
RANGE SELECTION
CONTROL VOLTAGE ANALOG OUTPUT

INPUT
FRONT PANEL
PCB VIEW
VCO USE
STANDARD VCO OPERATION
The VCO output frequency is controlled by an analog input voltage. The input voltage, Vin, is
scaled - amplified - by the front panel GAIN control. A DC voltage can be added to Vin inter-
nally, thus setting the start or CENTER FREQUENCY, fo . The CENTER FREQUENCY is defined
as the VCO output frequency, when no voltage is applied to the Vin connector. The Vin input is
internally tied to ground if no signal is applied.
The Vin OVERLOAD LED is lit when the sum of these voltages - scaled Vin plus CENTER FRE-
QUENCY DC offset - exceed the oscillator’s internal operating limits. Decrease the GAIN - turn
counter clockwise - and/or shift the CENTER FREQUENCY, fo , to extinguish the LED.
The frequency range switch selects between the HI or carrier band and the LO or audio band.
Both sinewave and digital outputs are available.

SPECIAL VCO OPERATION - FINE FREQUENCY CONTROL
In some applications, fine control may be required over the output frequency of the VCO. This
may be easily achieved by following these four steps:
MODULES REQUIRED: VCO and VARIABLE DC.
STEP 1 - Set the VARIABLE DC module’s output close to zero (marker knob at 12 o’clock
position).
STEP 2.1 - Turn the GAIN control of the VCO to zero, fully counter-clockwise.
STEP 2.2 - Now, turn the GAIN control up, clockwise, just a little (only a few degrees).
STEP 3 - Set the VCO module’s output frequency as close as possible to the frequency of
interest. Use the frequency adjust knob, fo. Use the FREQUENCY COUNTER to
measure the VCO’s output frequency.
STEP 4 - Finally, patch the VARIABLE DC module’s output to the VCO module’s frequency
control input, Vin, with a standard patching lead.
FINE FREQUENCY CONTROL of the VCO module is now achieved by turning the VARIABLE DC
module’s voltage control knob.
FSK USE
A PCB mounted slide switch selects between FSK and VCO modes of operation. The two out-
put frequencies, FSK1 and FSK2 , (MARK and SPACE), are set by varying the PCB mounted, fin-
ger adjustable trimmers. As in VCO mode, the frequency range switch selects between the HI or
carrier band and the LO or audio band. The digital data input accepts only TTL level signals.
Both sinewave and digital outputs are available.
GAIN and CENTER FREQ, fo , controls and the Vin connector are not used in the FSK mode.
VCO MODE
Frequency Ranges 1.5kHz < LO < 17kHz; sinewave and TTL
( < 300Hz with external input voltage, Vin )
70kHz < HI < 130kHz; sinewave and TTL
Input Voltage -3V < Vin < 3V
Overload limit indication LED Vvco > + 3V;
Vvco is the internal voltage finally applied to the VCO circuitry.
GAIN G.Vin : 1 < G < 2
Center Frequency Voltage Range - 3V < Vfc < 3V;
Vfc is a DC voltage added INTERNALLY to G.Vin
FSK MODE
Frequency Ranges 1.5kHz < FSK1, LO < 9kHz
500Hz < FSK2, LO < 4kHz
80kHz < FSK1, HI < 200kHz

20kHz < FSK2, HI < 120kHz
Data Input TTL level message

60kHz LOWPASS FILTER
An elliptic lowpass filter is provided with a cutoff point of approximately 60kHz. The input signal
amplitude can be adjusted with the gain control.
USE
The 60kHz LPF allows carrier signals to be removed from a given signal spectrum.
For example, as the lowpass filter for envelope detector applications.
The GAIN control allows input signals to be attenuated, to avoid overloading the filter.
Cutoff Frequency approx 60kHz
Passband Gain variable, 0 to 5 (approx)
Stopband Attenuation 50dB (typ)
Passband Ripple 0.1dB (typ)
PARAMETERS TO NOTE
corner point; response shape; passband ripple; phase shift; out of band attenuation.

TIMS-301/C Safety Information
The Emona TIMS-301/C User Manual contains important information and warnings which have to be
followed by the user to ensure safe operation.
The TIMS-301/C System is Safety Class I laboratory equipment and is designed to meet the requirements
of EN61010-1. The Installation Category is Category II, intended for operation from a normal single phase
supply.
The TIMS-301/C System has been designed for indoor use in a Pollution Degree I environment (no
pollution, or only dry non-conductive pollution) in the temperature range of 5 degrees C to 40 degrees C,
20% to 80% RH (non-condensing). It may occasionally be subjected to temperature between +5 degrees
and -10 degrees without degradation of its safety.
WARNING! TIMS EQUIPMENT MUST BE EARTHED
Any interruption of the mains earth conductor inside or outside the equipment will make the equipment
dangerous. Intentional interruption is prohibited.
Make sure that only fuses with the required rated current and of the specified type are used for
replacement.
Use of this equipment in a manner not specified by these instructions may impair the safety protection
provided. Do not operate the equipment outside its rated supply voltages or environmental range. In
particular, excessive moisture may impair safety.
Front Panel Connections

- INPUT SOCKETS, RED
WARNING!
DO NOT APPLY EXTERNAL VOLTAGES EXCEEDING +5.5V or -0.5V
and FREQUENCIES EXCEEDING 1MHz.
- INPUT SOCKETS, YELLOW

WARNING!
DO NOT APPLY EXTERNAL VOLTAGES EXCEEDING +/-15V
and FREQUENCIES EXCEEDING 200kHz.
- OUTPUT SOCKETS, RED and YELLOW

WARNING!
DO NOT APPLY EXTERNAL VOLATGES TO ANY OUTPUT SOCKET
Safety Terms and Symbols
WARNING. Warning of potential hazard
O mains supply OFF
I mains supply ON
Installation
Check the operating voltage marked on the rear panel is suitable for the local supply. Should it be
necessary to change the operating voltage, please contact your supplier or Emona Instruments Pty Ltd.
FUSE
Ensure that the correct mains fuse is fitted for the set operating voltage, as follows:
for 220 or 240V operation: 630mA (T) 250 V

for 110V or 127V operation 1.2A (T) 250V
MAINS LEAD
Ensure an approved type IEC mains lead with earth connection is used.
WARNING! TIMS EQUIPMENT MUST BE EARTHED
Any interruption of the mains earth conductor inside or outside the equipment will make the equipment
dangerous. Intentional interruption is prohibited.
MOUNTING
The equipment is designed for bench use.
Sicherheitsanweisung für das System TIMS-301/C
Die Gebrauchsanweisung des Systems TIMS-301/C beinhaltet wichtige Informationen und

Sicherheitswarnungen, die vom Benutzer beachtet werden müssen.
Dieses System ist ein Laborgerät der Sicherheitsklasse 1, das der Norm EN 61010-1 und der
Kategorie 2 entspricht und für einphasigen Strom vorgesehen ist.
Es ist nur für den Betrieb in Innenräumen zulässig, in einer Umgebung des Verschmutzungsgrades 1
( keine Verschmutzung oder trockene, nichtleitende Verschmutzung ), bei Temperaturen
zwischen 5 und 40°Celsius und 20-80% relativer Luft feuchtigkeit (ohne Kondensation).
Achtung! Das TIMS System muss geerdet werden
Jedes Abtrennen der Masseleitung im Inneren oder Äußeren des Systems ist lebensgefährlich. Der
Schutzleiter darf auf keinen Fall unterbrochen werden.
Benutzen Sie ausschließlich Sicherungen, deren Nennstrom und Typ den spezifizierten Werten
entsprechen.
Jeder Gebrauch dieses Apparats unter Bedingungen, die in dieser Gebrauchsanweisung nicht
festgesetzt sind, kann die Sicherheit beeinträchtigen. Benutzen Sie den Apparat keinesfalls außerhalb
der spezifizierten Versorgungsspannung oder der beschriebenen atmosphärischen Bedingungen. Ein
Übermaß an Feuchtigkeit vermindert die Sicherheit.
Frontplattenanschlüsse
ROTE EINGÄNGE
Achtung! Keine externe Spannung > +5,5 V / < -0,5 V oder Signale > 1 MHz
anschließen.
GELBE EINGÄNGE
Achtung! Keine externe Spannung die +/-15 V übersteigt oder Signale mit einer
Frequenz > 200 MHz anschließen
ROTE UND GELBE AUSGÄNGE

Achtung! Keine Signale an die Ausgänge anlegen
Sicherheitssymbole und Sicherheitsbegriffe

ACHTUNG. Warnung bezüglich eines Gefahrenpotentials
O Stromversorgung ausgeschaltet
I Stromversorgung eingeschaltet
Installierung
Stellen Sie sicher, dass die Betriebsspannung, die auf der Rückwand angegeben ist, Ihrer
Stromversorgung entspricht. Um die Betriebsspannung zu verändern, kontaktieren Sie Ihren
Lieferanten oder Emona Instruments Pty Ltd.
SICHERUNG
Stellen Sie sicher, dass Sie eine Sicherung benutzen, die Ihrer Betriebsspannung entspricht:
Für einen Gebrauch von 220 oder 240 V: 630 mA (T) 250 V
Für einen Gebrauch von 110 oder 127 V: 1,2 A (T) 250 V
VERSORGUNGSKABEL
Benutzen nur Netzkabel mit einem Schutzleiter.
Consignes de sécurité concernant le système TIMS-301/C
Le manuel utilisateur du système TIMS-301/C comprend d’importantes informations et mises en garde de
sécurité devant être prises en compte par l’utilisateur.
Ce système est un appareil de laboratoire de classe de sécurité 1 conforme à la norme EN61010-1 et de

catégorie d’installation 2, prévu pour fonctionner sous courant monophasé.
Il a été conçu pour être utilisé en intérieur, dans un environnement de degré de pollution 1 (pas de pollution
ou pollution sèche non conductrice) sous des températures variant entre 5 et 40 ºC, avec 20 à 80 %
d’humidité relative (sans condensation). Il peut occasionnellement être utilisé à des températures entre 5
et -10 ºC.
ATTENTION ! LE SYSTÈME TIMS DOIT ÊTRE MIS À LA TERRE
Toute déconnexion du fil de masse à l'intérieur ou à l'extérieur du système rendra son utilisation
dangereuse. Ce fil ne doit en aucun cas être déconnecté.
N’utilisez que des fusibles dont le courant nominal et le type correspondent aux valeurs spécifiées.
Toute utilisation de cet appareil dans des conditions non stipulées dans le manuel est susceptible
d’endommager les dispositifs de sécurité. N’utilisez pas l’appareil hors des tensions d’alimentation ou des
conditions atmosphériques prescrites. Un excès d’humidité peut notamment nuire à la sécurité du système.
Connexions du panneau avant

- ENTRÉES, ROUGES
ATTENTION !
NE PAS APPLIQUER DE TENSIONS EXTÉRIEURES SUPÉRIEURES À +5,5 V OU
INFÉRIEURES À -0,5 V
ni de FRÉQUENCES SUPÉRIEURES À 1 MHz.
- ENTRÉES, JAUNES
ATTENTION !
NE PAS APPLIQUER DE TENSIONS EXTÉRIEURES DÉPASSANT +/-15 V
ni de FRÉQUENCES SUPÉRIEURES À200 MHz
- SORTIES, ROUGES et JAUNES

ATTENTION !
NE PAS APPLIQUER DE TENSIONS EXTÉRIEURES AUX SORTIES
Termes et symboles de sécurité
ATTENTION. Avertissement concernant un danger potentiel
O alimentation secteur désactivée
I alimentation secteur activée

Installation
Vérifier que la tension de fonctionnement stipulée sur le panneau arrière correspond à votre tension
d’alimentation. Pour changer la tension de fonctionnement, veuillez contacter votre fournisseur ou Emona
Instruments Pty Ltd.
FUSIBLE
Vérifiez que vous utilisez le fusible d’alimentation correspondant à votre tension de fonctionnement :
pour une utilisation en 220 ou 240 V : 630 mA (T) 250 V

pour une utilisation en 110 ou 127 V : 1,2 A (T) 250 V
CÂBLE D’ALIMENTATION
Utilisez un câble d’alimentation avec prise de terre approuvé par la CEI.
ATTENTION ! LE SYSTÈME TIMS DOIT ÊTRE MIS À LA TERRE
Toute déconnexion du fil de masse à l'intérieur ou à l'extérieur du système rendra son utilisation
dangereuse. Ce fil ne doit en aucun cas être déconnecté.
MONTAGE
Cet appareil est conçu pour une utilisation en laboratoire.
EC Declaration of Conformity
for the
Emona TIMS series of Telecommunications Teaching Equipment –
Modular electronic building-blocks of open architecture intended to be used

exclusively for the purposes of experimentation, learning and practical training
to investigate the phenomena of modulation, coding and signal transmission
in telecommunications.
Manufactured by:
Emona Instruments Pty Ltd
86 Parramatta Road, Camperdown NSW AUSTRALIA
Statement of Conformity:
Based on test results using appropriate standards, the product is in conformity
with Low Voltage Directive 73/23/EEC.
“The use of the apparatus outside the classroom, laboratory, study area or
similar such place invalidates conformity with the protection requirements of
the Electromagnetic Compatibility Directive (89/336/EEC) and could lead to
prosecution.
The equipment when operated does not cause electromagnetic disturbance to

apparatus situated outside its immediate electromagnetic environment.”
Standard used:
EN 61010-1 (1993) Safety Requirements for Electrical Equipment for
Measurement, Control, and Laboratory Use.
The tests have been performed in a typical configuration.
This Conformity is indicated by the symbol.
Alfred Breznik
Technical Director
INTRODUCTION TO
MODELLING WITH TIMS
model building.............................................................................2
why have patching diagrams ?....................................................................2
organization of experiments ........................................................3
who is running this experiment ?.................................................3
early experiments.........................................................................4
modulation..................................................................................................4
messages ......................................................................................4
analog messages .........................................................................................4
digital messages..........................................................................................5
bandwidths and spectra................................................................5
measurement...............................................................................................6
graphical conventions ..................................................................6
representation of spectra.............................................................................6
filters ..........................................................................................................8
other functions............................................................................................9
measuring instruments .................................................................9
the oscilloscope - time domain ...................................................................9
the rms voltmeter......................................................................................10
the spectrum analyser - frequency domain ...............................................10
oscilloscope - triggering ............................................................10
what you see, and what you don`t..............................................11
overload. ....................................................................................11
overload of a narrowband system.............................................................12
the two-tone test signal.............................................................................12
Fourier series and bandwidth estimation ...................................13
multipliers and modulators ........................................................13
multipliers ................................................................................................13
modulators................................................................................................14
envelopes ...................................................................................15
extremes.....................................................................................15
analog or digital ? ......................................................................15
SIN or COS ? .............................................................................16
the ADDER - G and g..............................................................16
abbreviations..............................................................................17
list of symbols............................................................................18
Introduction to modelling with TIMS Vol A1, ch 1, rev 1.0 - 1

INTRODUCTION TO
MODELLING WITH TIMS
model building
With TIMS you will be building models. These models will most often be
hardware realizations of the block diagrams you see in a text book, or have
designed yourself. They will also be representations of equations, which
themselves can be depicted in block diagram form.
What ever the origin of the model, it can be patched up in a very short time. The
next step is to adjust the model to perform as expected. It is perfectly true that you
might, on occasions, be experimenting, or just ‘doodling’, not knowing what to
expect. But in most cases your goal will be quite clear, and this is where a
systematic approach is recommended.
If you follow the steps detailed in the first few experiments you will find that the
models are adjusted in a systematic manner, so that each desired result is obtained
via a complete understanding of the purpose and aim of the intermediate steps
leading up to it.
why have patching diagrams ?

Many of the analog experiments, and all of the digital experiments, display
patching diagrams. These give all details of the interconnections between
modules, to implement a model of the system under investigation.
It is not expected that a glance at the patching diagram

will reveal the nature of the system being modelled.
The patching diagram is presented as firm evidence that a model of the system can
be created with TIMS.
The functional purpose of the system is revealed through the

block diagram which precedes the patching diagram.
2 - A1 Introduction to modelling with TIMS

It is the block diagram which you should study to gain insight into the workings of
the system.
If you fully understand the block diagram you should not need the patching
diagram, except perhaps to confirm which modules are required for particular
operations, and particular details of functionality. These is available in the TIMS
User Manual.
You may need an occasional glance at the patching diagram for confirmation of a
particular point.
Try to avoid patching up ‘mechanically’,

according to the patching diagram, without
thought to what you are trying to achieve.
organization of experiments
Each of the experiments in this Text is divided into three parts.
1. The first part is generally titled PREPARATION. This part should be studied
before the accompanying laboratory session.
2. The second part describes the experiment proper. Its title will vary. You will
find the experiment a much more satisfying experience if you arrive at the
laboratory well prepared, rather than having to waste time finding out what has
to be done at the last moment. Thus read this part before the laboratory
session.
3. The third part consists of TUTORIAL QUESTIONS. Generally these
questions will be answered after the experimental work is completed, but it is a
good idea to read them before the laboratory session, in case there are special
measurements to be made.
While performing an experiment you should always have access to the TIMS user
manuals - namely the TIMS User Manual (fawn cover) which contains
information about the modules in the TIMS Basic Set of modules, and the TIMS
Advanced Modules and TIMS Special Applications Modules User Manual (red
cover).
who is running this experiment ?

These experiments and their Tasks are merely suggestions as to how you might go
about carrying out certain investigations. In the final assessment it is you who are
running the experiment, and you must make up your mind as to how you are going
to do it. You can do this best if you read about it beforehand.
If you do not understand a particular instruction, consider what it is you have been
trying to achieve up to that point, and then do it your way.
Introduction to modelling with TIMS A1 - 3

early experiments
The first experiment assumes no prior knowledge of telecommunications - it is
designed to introduce you to TIMS, and to illustrate the previous remarks about
being systematic. The techniques learned will be applied over and over again in
later work.
The next few experiments are concerned with analog modulation and
demodulation.
modulation
One of the many purposes of modulation is to convert a message into a form more
suitable for transmission over a particular medium.
The analog modulation methods to be studied will generally transform the analog
message signal in the audio spectrum to a higher location in the frequency
spectrum.
The digital modulation methods to be studied will generally transform a binary
data stream (the message), at baseband 1 frequencies, to a different format, and
then may or may not translate the new form to a higher location in the frequency
spectrum.
It is much easier to radiate a high frequency (HF) signal than it is a relatively low
frequency (LF) audio signal. In the TIMS environment the particular part of the
spectrum chosen for HF signals is centred at 100 kHz.
It is necessary, of course, that the reverse process, demodulation, can be carried
out - namely, that the message may be recovered from the modulated signal upon
receipt following transmission.
messages
Many models will be concerned with the transmission or reception of a message,
or a signal carrying a message. So TIMS needs suitable messages. These will
vary, depending on the system.
analog messages
The transmission of speech is often the objective in an analog system.
High-fidelity speech covers a wide frequency range, say 50 Hz to 15 kHz, but for
communications purposes it is sufficient to use only those components which lie in
the audio frequency range 300 to 3000 Hz - this is called ‘band limited speech’.
Note that frequency components have been removed from both the low and the
high frequency end of the message spectrum. This is bandpass filtering.
Intelligibility suffers if only the high frequencies are removed.
Speech is not a convenient message signal with which to make simple and precise
measurements. So, initially, a single tone (sine wave) is used. This signal is more
easily accommodated by both the analytical tools and the instrumentation and
measuring facilities.
1 defined later

The frequency of this tone can be chosen to lie within the range expected in the
speech, and its peak amplitude to match that of the speech. The simple tone can
then be replaced by a two-tone test signal, in which case intermodulation tests can
be carried out 2.
When each modulation or demodulation system has been set up quantitatively
using a single tone as a message (or, preferably with a two-tone test signal), a final
qualitative check can be made by replacing the tone with a speech signal. The
peak amplitude of the speech should be adjusted to match that of the tone. Both
listening tests (in the case of demodulation) and visual examination of the
waveforms can be very informative.
digital messages
The transmission of binary sequences is often the objective of a digital
communication system. Of considerable interest is the degree of success with
which this transmission is achieved. An almost universal method of describing the
quality of transmission is by quoting an error rate 3.
If the sequence is one which can take one of two levels, say 0 and 1, then an error
is recorded if a 0 is received when a 1 was sent, or a 1 received when a 0 was sent.
The bit error rate is measured as the number of errors as a proportion of total bits
sent.
To be able to make such a measurement it is necessary to know the exact nature of
the original message. For this purpose a known sequence needs to be transmitted,
a copy of which can be made available at the receiver for comparison purposes.
The known sequence needs to have known, and useful, statistical properties - for
example, a ‘random’ sequence. Rather simple generators can be implemented
using shift registers, and these provide sequences of adjustable lengths. They are
known as pseudo-random binary sequence (PRBS) generators. TIMS provides
you with just such a SEQUENCE GENERATOR module. You should refer to a
suitable text book for more information on these.
bandwidths and spectra

Most of the signals you will be examining in the experiments to follow have well
defined bandwidths. That is, in most cases it is possible to state quite clearly that
all of the energy of a signal lies between frequencies f1 and f2 Hz, where f1 < f2.
• the absolute bandwidth of such a signal is defined as (f2 - f1) Hz.
It is useful to define the number of octaves a signal occupies. The octave measure
for the above signal is defined as
octaves = log2(f2 / f1)
Note that the octave measure is a function of the ratio of two frequencies; it says
nothing about their absolute values.
• a wideband signal is generally considered to be one which occupies one or
more octaves.
2 the two-tone test signal is introduced in the experiment entitled ‘Amplifier overload’.
3 the corresponding measurement in an analog system would be the signal-to-noise ratio (relatively
easy to measure with instruments), or, if speech is the message, the ‘intelligibility’; not so easy to
define, let alone to measure.

• a narrowband signal is one which occupies a small fraction of an octave.
Another name, used interchangeably, is a bandpass signal.
An important observation can be made about a narrowband signal; that is, it can
contain no harmonics.
• a baseband signal is one which extends from DC (so f1 = 0) to a finite
frequency f2. It is thus a wideband signal.
Speech, for communications, is generally bandlimited to the range 300 to
3000 Hz. It thus has a bandwidth in excess of 3 octaves. This is considered to be
a wideband signal. After modulation, to a higher part of the spectrum, it becomes
a narrowband signal, but note that its absolute bandwidth remains unchanged.
This reduction from a wideband to a narrowband signal is a linear process; it can
be reversed. In the context of communications engineering it involves
modulation, or frequency translation.
You will meet all of these signals and phenomena when working with TIMS.
measurement
The bandwidth of a signal can be measured with a SPECTRUM ANALYSER.
Commercially available instruments typically cover a wide frequency range, are
very accurate, and can perform a large number of complex measurements. They
are correspondingly expensive.
TIMS has no spectrum analyser as such, but can model one (with the TIMS320
DSP module), or in the form of a simple WAVE ANALYSER with TIMS analog
modules. See the experiment entitled Spectrum analysis - the WAVE ANALYSER
(within Volume A2 - Further & Advanced Analog Experiments).
Without a spectrum analyser it is still possible to draw conclusions about the
location of a spectrum, by noticing the results when attempting to pass it through
filters of different bandwidths. There are several filters in the TIMS range of
modules. See Appendix A, and also the TIMS User Manual.
graphical conventions
representation of spectra
It is convenient to have a graphical method of depicting spectra. In this work we
do not get involved with the Fourier transform, with its positive and negative
frequencies and double sided spectra. Elementary trigonometrical methods are
used for analysis. Such methods are more than adequate for our purposes.
When dealing with speech the mathematical analysis is dropped, and descriptive
methods used. These are supported by graphical representations of the signals and
their spectra.
In the context of modulation we are constantly dealing with sidebands, generally
derived from a baseband message of finite bandwidth. Such finite bandwidth
signals will be represented by triangles on the spectral diagrams.
The steepness of the slope of the triangle has no special significance, although
when two or more sidebands, from different messages, need to be distinguished,
each can be given a different slope.

frequency
a baseband signal (eg., a message)
Although speech does not have a DC component, the triangle generally extends
down to zero (the origin) of the frequency scale (rather than being truncated just
before it). For the special case in which a baseband signal does have a DC
component the triangle convention is sometimes modified slightly by adding a
vertical line at the zero-frequency end of the triangle.
a DSBSC
The direction of the slope is important. Its significance becomes obvious when
we wish to draw a modulated signal. The figure above shows a double sideband
suppressed carrier (DSBSC) signal.
Note that there are TWO triangles, representing the individual lower and upper
sidebands. They slope towards the same point; this point indicates the location of
the (suppressed) carrier frequency.
an inverted baseband signal

The orientation is important. If the same message was so modulated that it could
be represented in the frequency spectrum as in the figure above, then this means:
• the signal is located in the baseband part of the spectrum
• spectral components have been transposed, or inverted; frequency
components which were originally above others are now below them.
• since the signal is at baseband it would be audible (if converted with an
electric to acoustic transducer - a pair of headphones, for example), but
would be unintelligible. You will be able to listen to this and other such
signals in TIMS experiments to come.
It is common practice to use the terms erect and inverted to describe these bands.

In the Figure above, a message (a) is frequency translated to become an upper
single sideband (b), and a lower single sideband (c). A three-channel frequency
division multiplexed (FDM) signal is also illustrated (d).
Note that these spectral diagrams do not show any phase information.
Despite all the above, be prepared to accept that these diagrams are used for
purposes of illustration, and different authors use their own variations. For
example, some slope their triangles in the opposite sense to that suggested here.
filters
In a block diagram, there is a simple technique for representing filters. The
frequency spectrum is divided into three bands - low, middle, and high - each
represented by part of a sinewave. If a particular band is blocked, then this is
indicated by an oblique stroke through it. The standard responses are represented
as in the Figure below.
block-diagrammatic representations of filter responses
The filters are, respectively, lowpass,

bandpass, highpass, bandstop, and
allpass.
In the case of lowpass and highpass
responses the diagrams are often further
simplified by the removal of one of the
cancelled sinewaves, the result being as
in the figure opposite.

other functions
amplify add multiply amplitude integrate

limit
some analog functions
measuring instruments
the oscilloscope - time domain

The most frequently used measuring facility with TIMS is the oscilloscope. In
fact the vast majority of experiments can be satisfactorily completed with no other
instrument.
Any general purpose oscilloscope is ideal for all TIMS experiments. It is intended
for the display of signals in the time domain 4. It shows their waveforms - their
shapes, and amplitudes
From the display can be obtained information regarding:
• waveform shape
• waveform frequency - by calculation, using time base information
• waveform amplitude - directly from the display
• system linearity - by observing waveform distortion
• an estimate of the bandwidth of a complex signal; eg, from the sharpness of
the corners of a square wave
When concerned with amplitude information it is customary to record either:
• the peak-to-peak amplitude

• the peak amplitude
of the waveform visible on the screen.

Unless the waveform is a simple sinewave it is always important to record the
shape of the waveform also; this can be:
1. as a sketch (with time scale), and annotation to show clearly what amplitude
has been measured.
2. as an analytic expression, in which case the parameter recorded must be
clearly specified.
4 but with adaptive circuitry it can be modified to display frequency-domain information

the rms voltmeter
The TIMS WIDEBAND TRUE RMS METER module is essential for
measurements concerning power, except perhaps for the simple case when the
signal is one or two sinewaves. It is particularly important when the measurement
involves noise.
Its bandwidth is adequate for all of the signals you will meet in the TIMS
environment.
An experiment which introduces the WIDEBAND TRUE RMS METER, is
entitled Power measurements. Although it appears at the end of this Volume, it
could well be attempted at almost any time.
the spectrum analyser - frequency domain

The identification of the spectral composition of a signal - its components in the
frequency domain - plays an important part when learning about communications.
Unfortunately, instruments for displaying spectra tend to be far more expensive
than the general purpose oscilloscope.
It is possible to identify and measure the individual spectral components of a
signal using TIMS modules.
Instruments which identify the spectral components on a component-by-
component basis are generally called wave analysers. A model of such an
instrument is examined in the experiment entitled Spectrum analysis - the WAVE
ANALYSER in Volume A2 - Further & Advanced Analog Experiments.
Instruments which identify the spectral components of a signal and display the
spectrum are generally called spectrum analysers. These instruments tend to be
more expensive than wave analysers. Something more sophisticated is required
for their modelling, but this is still possible with TIMS, using the digital signals
processing (DSP) facilities - the TIMS320 module can be programmed to provide
spectrum analysis facilities.
Alternatively the distributors of TIMS can recommend other affordable methods,
compatible with the TIMS environment.
oscilloscope - triggering
synchronization
As is usually the case, to achieve ‘text book like’ displays, it is important to
choose an appropriate signal for oscilloscope triggering. This trigger signal is
almost never the signal being observed ! The recognition of this point is an
important step in achieving stable displays.
This chosen triggering signal should be connected directly to the oscilloscope
sweep synchronizing circuitry. Access to this circuitry of the oscilloscope is
available via an input socket other than the vertical deflection amplifier input(s).
It is typically labelled ‘ext. trig’ (external trigger), ‘ext. synch’ (external
synchronization), or similar.
sub-multiple frequencies
If two or more periodic waveforms are involved, they will only remain stationary
with respect to each other if the frequency of one is a sub-multiple of the other.

This is seldom the case in practice, but can be made so in the laboratory. Thus
TIMS provides, at the MASTER SIGNALS module, a signal of 2.083 kHz (which
is 1/48 of the 100 kHz system clock), and another at 8.333 kHz (1/12 of the
system clock).
which channel ?
Much time can be saved if a consistent use of the SCOPE SELECTOR is made.
This enables quick changes from one display to another with the flip of a switch.
In addition, channel identification is simplified if the habit is adopted of
consistently locating the trace for CH1 above the trace for CH2.
Colour coded patching leads can also speed trace identification.
what you see, and what you don`t

Instructions such as ‘adjust the phase until there is no output’, or ‘remove the
unwanted signal with a suitable filter’ will be met from time to time.
These instructions seldom result in the amplitude of the signal in question being
reduced to zero. Instead, what is generally meant is ‘reduce the amplitude of the
signal until it is no longer of any significance’.
Significance here is a relative term, made with respect to the system signal-to-
noise ratio (SNR). All systems have a background noise level (noise threshold,
noise floor), and signals (wanted) within these systems must over-ride this noise
(unwanted).
TIMS is designed to have a ‘working level’, the TIMS ANALOG REFERENCE LEVEL,
of about 4 volts peak-to-peak. The system noise level is claimed to be at least
100 times below this 5.
When using an oscilloscope as a measuring instrument with TIMS, the vertical
sensitivity is typically set to about 1 volt/cm. Signals at the reference level fit
nicely on the screen. If they are too small it is wise to increase them if possible
(and appropriate), to over-ride the system noise; or if larger to reduce them, to
avoid system overload.
When they are attenuated by a factor of 100, and if the oscilloscope sensitivity is
not changed, they appear to be ‘reduced to zero’; and in relative terms this is so.
If the sensitivity of the oscilloscope is increased by 100, however, the screen will
no longer be empty. There will be the system noise, and perhaps the signal of
interest is still visible. Engineering judgement must then be exercised to evaluate
the significance of the signals remaining.
overload
If wanted signal levels within a system fall ‘too low’ in amplitude, then the signal-
to-noise ratio (SNR) will suffer, since internal circuit noise is independent of
signal level.
If signal levels within a system rise ‘too high’, then the SNR will suffer, since the
circuitry will overload, and generate extra, unwanted, distortion components;
these distortion components are signal level dependent. In this case the noise is
5 TIMS claims a system signal-to-noise ratio of better than 40 dB

derived from distortion of the signal, and the degree of distortion is usually quoted
as signal-to-distortion ratio (SDR).
Thus analog circuit design includes the need to maintain signal levels at a pre-
defined working level, being ‘not to high’ and ‘not too low’, to avoid these two
extremes.
These factors are examined in the experiment entitled Amplifier overload within
Volume A2 - Further & Advanced Analog Experiments.
The TIMS working signal level, or TIMS ANALOG REFERENCE LEVEL, has been set
at 4 volts peak-to-peak. Modules will generally run into non-linear operation
when this level is exceeded by say a factor of two. The background noise of the
TIMS system is held below about 10 mV - this is a fairly loose statement, since
this level is dependent upon the bandwidth over which the noise is measured, and
the model being examined at the time. A general statement would be to say that
TIMS endeavours to maintain a SNR of better than 40 dB for all models.
overload of a narrowband system

Suppose a channel is narrowband. This means it is deliberately bandlimited so
that it passes signals in a narrow (typically much less than an octave 6) frequency
range only. There are many such circuits in a communications system.
If this system overloads on a single tone input, there will be unwanted harmonics
generated. But these will not pass to the output, and so the overload may go
unnoticed. With a more complex input - say two or more tones, or a speech-
related signal - there will be, in addition, unwanted intermodulation components
generated. Many of these will pass via the system, thus revealing the existence of
overload. In fact, the two-tone test signal should always be used in a narrowband
system to investigate overload.
the two-tone test signal

A two-tone test signal consists of two sine waves added together ! As discussed in
the previous section, it is a very useful signal for testing systems, especially those
which are of narrow-bandwidth. The properties of the signal depend upon:
• the frequency ratio of the two tones.
• the amplitude ratio of the two tones.
For testing narrowband communication systems the two tones are typically of
near-equal frequency, and of identical amplitude. A special property of this form
of the signal is that its shape, as seen in the time domain, is very well defined and
easily recognisable 7.
After having completed the early experiments you will recognise this shape as that
of the double sideband suppressed carrier (DSBSC) signal.
If the system through which this signal is transmitted has a non-linear transmission
characteristic, then this will generate extra components. The presence of even
small amounts of these components is revealed by a change of shape of the test
signal.
6 defined above
7 the assumption being that the oscilloscope is set to sweep across the screen over a few periods of
the difference frequency.

Fourier series and bandwidth
estimation
Fourier series analysis of periodic signals reveals that:
• it is possible, by studying the symmetry of a signal, to predict the presence or
absence of a DC component.
• if a signal is other than sinusoidal, it will contain more than one harmonic
component of significance.
• if a signal has sharp discontinuities, it is likely to contain many harmonic
components of significance
• some special symmetries result in all (or nearly all) of the ODD (or EVEN)
harmonics being absent.
With these observations, and more, it is generally easy to make an engineering
estimate of the bandwidth of a periodic signal.
multipliers and modulators

The modulation process requires multiplication. But a pure MULTIPLIER is
seldom found in communications equipment. Instead, a device called a
MODULATOR is used.
In the TIMS system we generally use a MULTIPLIER, rather than a
MODULATOR, when multiplication is called for, so as not to become diverted by
the side effects and restrictions imposed by the latter.
In commercial practice, however, the purpose-designed MODULATOR is
generally far superior to the unnecessarily versatile MULTIPLIER.
multipliers
An ideal multiplier performs as a multiplier should ! That is, if the two time-
domain functions x(t) and y(t) are multiplied together, then we expect the result to
be x(t).y(t), no more and no less, and no matter what the nature of these two
functions. These devices are called four quadrant multipliers.
There are practical multipliers which approach this ideal, with one or two
engineering qualifications. Firstly, there is always a restriction on the bandwidth
of the signals x(t) and y(t).
There will inevitably be extra (unwanted) terms in the output (noise, and
particularly distortion products) due to practical imperfections.
Provided these unwanted terms can be considered ‘insignificant’, with respect to
the magnitude of the wanted terms, then the multiplier is said to be ‘ideal’. In the
TIMS environment this means they are at least 40 dB below the TIMS ANALOG
REFERENCE LEVEL 8.
Such a multiplier is even said to be linear. That is, from an engineering point of
view, it is performing as expected.
8 defined under ‘what you see and what you don`t’

In the mathematical sense it is not linear, since the mathematical definition of a
linear circuit includes the requirement that no new frequency components are
generated when it performs its normal function. But, as will be seen,
multiplication always generates new frequency components.
DC off-sets
One of the problems associated with analog circuit design is minimization of
unwanted DC off-sets. If the signals to be processed have no DC component
(such as in an audio system) then stages can be AC coupled, and the problem is
overcome. In the TIMS environment module bandwidths must extend to DC, to
cope with all possible conditions; although more often than not signals have no
intentional DC component.
In a complex model DC offsets can accumulate - but in most cases they can be
recognised as such, and accounted for appropriately. There is one situation,
however, where they can cause much more serious problems by generating new
components - and that is when multiplication is involved.
With a MULTIPLIER the presence of an unintentional DC component at one
input will produce new components at the output. Specifically, each component at
the other input will be multiplied by this DC component - a constant - and so a
scaled version will appear at the output 9.
To overcome this problem there is an option for AC coupling in the
MULTIPLIER module. It is suggested that the DC mode be chosen only when the
signals to be processed actually have DC components; otherwise use AC
coupling.
modulators
In communications practice the circuitry used for the purpose of performing the
multiplying function is not always ideal in the four quadrant multiplier sense;
such circuits are generally called modulators.
Modulators generate the wanted sum or difference products but in many cases the
input signals will also be found in the output, along with other unwanted
components at significant levels. Filters are used to remove these unwanted
components from the output (alternatively there are ‘balanced’ modulators. These
have managed to eliminate either one or both of the original signals from the
output).
These modulators are restricted in other senses as well. It is allowed that one of
the inputs can be complex (ie., two or more components) but the other can only be
a single frequency component (or appear so to be - as in the switching modulator).
This restriction is of no disadvantage, since the vast majority of modulators are
required to multiply a complex signal by a single-component carrier.
Accepting restrictions in some areas generally results in superior performance in
others, so that in practice it is the switching modulator, rather than the idealized
four quadrant multiplier, which finds universal use in communications electronics.
Despite the above, TIMS uses the four quadrant multiplier in most applications
where a modulator might be used in practice. This is made possible by the
relatively low frequency of operation, and modest linearity requirements
9 this is the basis of a voltage controlled amplifier - VCA

envelopes
Every narrowband signal has an envelope, and you probably have an idea of what
this means.
Envelopes will be examined first in the experiment entitled DSB generation in
this Volume.
They will be defined and further investigated in the experiments entitled
Envelopes within this Volume, and Envelope recovery within Volume A2 -
Further & Advanced Analog Experiments.
extremes
Except for a possible frequency scaling effect, most experiments with TIMS will
involve realistic models of the systems they are emulating. Thus message
frequencies will be ‘low’, and carrier frequencies ‘high’. But these conditions
need not be maintained. TIMS is a very flexible environment.
It is always a rewarding intellectual exercise to

imagine what would happen if one or more of
the ‘normal’ conditions was changed severely 10.
It is then even more rewarding to confirm our imaginings by actually modelling

these unusual conditions. TIMS is sufficiently flexible to enable this to be done in
most cases.
For example: it is frequently stated, for such-and-such a requirement to be
satisfied, that it is necessary that ‘x1 >> x2’. Quite often x1 and x2 are frequencies
- say a carrier and a message frequency; or they could be amplitudes.
You are strongly encouraged to expand your horizons by questioning the reasons
for specifying the conditions, or restrictions, within a model, and to consider, and
then examine, the possibilities when they are ignored.
analog or digital ?
What is the difference between a digital signal and an analog signal ? Sometimes
this is not clear or obvious.
In TIMS digital signals are generally thought of as those being compatible with
the TTL standards. Thus their amplitudes lie in the range 0 to +5 volts. They
come from, and are processed by, modules having RED output and input sockets.
It is sometimes necessary, however, to use an analog filter to bandlimit these
signals. But their large DC offsets would overload most analog modules, . Some
digital modules (eg, the SEQUENCE GENERATOR) have anticipated this, and
provide an analog as well as a digital (TTL) output. This analog output comes
10 for an entertaining and enlightening look at the effects of major changes to one or more of the
physical constants, see G. Gamow; Mr Tompkins in Wonderland published in 1940, or easier Mr.
Tompkins in Paperback, Cambridge University Press, 1965.

from a YELLOW socket, and is a TTL signal with the DC component removed
(ie, DC shifted).
SIN or COS ?
Single frequency signals are generally referred to as sinusoids, yet when
manipulating them trigonometrically are often written as cosines. How do we
obtain cosωt from a sinusoidal oscillator !
There is no difference in the shape of a sinusoid and a cosinusoid, as observed
with an oscilloscope. A sinusoidal oscillator can just as easily be used to provide
a cosinusoid. What we call the signal (sin or cos) will depend upon the time
reference chosen.
Remember that cosωt = sin(ωt + π/2)
Often the time reference is of little significance, and so we choose sin or cos, in
any analysis, as is convenient.
the ADDER - G and g

Refer to the TIMS User Manual for a description of the ADDER module. Notice
it has two input sockets, labelled ‘A’ and ‘B’.
In many experiments an ADDER is used to make a linear sum of two signals a(t)
and b(t), of amplitudes A and B respectively, connected to the inputs A and B
respectively. The proportions of these signals which appear at the ADDER output
are controlled by the front panel gain controls G and g.
The amplitudes A and B of the two input signals are seldom measured, nor the
magnitudes G and g of the adjustable gains.
Instead it is the magnitudes GA and gB which are of more interest, and these are
measured directly at the ADDER output. The measurement of GA is made when
the patch lead for input B is removed; and that of gB is measured when the patch
lead for input A is removed.
When referring to the two inputs in this text it would be formally correct to name
them as ‘the input A’ and ‘the input B’. This is seldom done. Instead, they are
generally referred to as ‘the input G’ and ‘the input g’ respectively (or sometimes
just G and g). This should never cause any misunderstanding. If it does, then it is
up to you, as the experimenter, to make an intelligent interpretation.

abbreviations
This list is not exhaustive. It includes only those abbreviations used in this Text.
abbreviation meaning
AM amplitude modulation
ASK amplitude shift keying (also called OOK)
BPSK binary phase shift keying
CDMA code division multiple access
CRO cathode ray oscilloscope
dB decibel
DPCM differential pulse code modulation
DPSK differential phase shift keying
DSB double sideband (in this text synonymous with DSBSC)
DSBSC double sideband suppressed carrier
DSSS direct sequence spread spectrum
DUT device under test
ext. synch. external synchronization (of oscilloscope). ‘ext. trig.’ preferred
ext. trig. external trigger (of an oscilloscope)
FM frequency modulation
FSK frequency shift keying
FSD full scale deflection (of a meter, for example)
IP intermodulation product
ISB independent sideband
ISI intersymbol interference
LSB analog: lower sideband digital: least significant bit
MSB most significant bit
NBFM narrow band frequency modulation
OOK on-off keying (also called ASK)
PAM pulse amplitude modulation
PCM pulse code modulation
PDM pulse duration modulation (see PWM)
PM phase modulation
PPM pulse position modulation
PRK phase reversal keying (also called PSK)
PSK phase shift keying (also called PRK - see BPSK)
PWM pulse width modulation (see PDM)
SDR signal-to-distortion ratio
SNR signal-to-noise ratio
SSB single sideband (in this text is synonymous with SSBSC)
SSBSC single sideband suppressed carrier
SSR sideband suppression ratio
TDM time division multiplex
THD total harmonic distortion
VCA voltage controlled amplifier
WBFM wide band frequency modulation

list of symbols
The following symbols are used throughout the text, and have the following
meanings
a(t) a time varying amplitude

α, φ, ϕ, phase angles
β deviation, in context of PM and FM
δf a small frequency increment
∆φ peak phase deviation
δt a small time interval
φ(t) a time varying phase
m in the context of envelope modulation, the depth of modulation
µ a low frequency (rad/s); typically that of a message (µ << ω).
ω a high frequency (rad/s); typically that of a carrier (ω >> µ)
y(t) a time varying function

DSBSC GENERATION
PREPARATION................................................................................. 34
definition of a DSBSC .............................................................. 34
block diagram...........................................................................................36
viewing envelopes ..................................................................... 36
multi-tone message.................................................................... 37
linear modulation .....................................................................................38
spectrum analysis ...................................................................... 38
EXPERIMENT ................................................................................... 38
the MULTIPLIER ..................................................................... 38
preparing the model................................................................... 38
signal amplitude. ....................................................................... 39
fine detail in the time domain.................................................... 40
overload ...................................................................................................40
bandwidth.................................................................................. 41
alternative spectrum check ........................................................ 44
speech as the message ............................................................... 44
TUTORIAL QUESTIONS ................................................................. 45
TRUNKS................................................................................... 46
APPENDIX......................................................................................... 46
TUNEABLE LPF tuning information ....................................... 46
DSBSC generation Vol A1, ch 3, rev 1.1 - 33

DSBSC GENERATION
ACHIEVEMENTS: definition and modelling of a double sideband suppressed

carrier (DSBSC) signal; introduction to the MULTIPLIER, VCO,
60 kHz LPF, and TUNEABLE LPF modules; spectrum estimation;
multipliers and modulators.
PREREQUISITES: completion of the experiment entitled ‘Modelling an equation’

in this Volume.
PREPARATION
This experiment will be your introduction to the MULTIPLIER and the double
sideband suppressed carrier signal, or DSBSC. This modulated signal was probably
not the first to appear in an historical context, but it is the easiest to generate.
You will learn that all of these modulated signals are derived from low frequency
signals, or ‘messages’. They reside in the frequency spectrum at some higher
frequency, being placed there by being multiplied with a higher frequency signal,
usually called ‘the carrier’ 1.
definition of a DSBSC
Consider two sinusoids, or cosinusoids, cosµt and cosωt. A double sideband
suppressed carrier signal, or DSBSC, is defined as their product, namely:
DSBSC = E.cosµt . cosωt ........ 1
Generally, and in the context of this experiment, it is understood that::

ω >> µ ........ 2
Equation (3) can be expanded to give:

cosµt . cosωt = (E/2) cos(ω - µ)t + (E/2) cos(ω + µ)t ...... 3
Equation 3 shows that the product is represented by two new signals, one on the sum
frequency (ω + µ), and one on the difference frequency (ω - µ) - see Figure 1.
1 but remember whilst these low and high qualifiers reflect common practice, they are not mandatory.
34 - A1 DSBSC generation
Remembering the inequality of eqn. (2) the two new components are located close to
the frequency ω rad/s, one just below, and the other just above it. These are referred
to as the lower and upper sidebands 2 respectively.
These two components were

E
2 derived from a ‘carrier’ term on
ω rad/s, and a message on
µ rad/s. Because there is no term
at carrier frequency in the
product signal it is described as a
ω µ ω +µ frequency double sideband suppressed
carrier (DSBSC) signal.
Figure 1: spectral components
The term ‘carrier’ comes from the context of ‘double sideband amplitude
modulation' (commonly abbreviated to just AM).
AM is introduced in a later experiment (although, historically, AM preceded
DSBSC).
The time domain appearance of a DSBSC (eqn. 1) in a text book is generally as
shown in Figure 2.
+1
message
0
-1
DSBSC
E
time
-E
Figure 2: eqn.(1) - a DSBSC - seen in the time domain
Notice the waveform of the DSBSC in Figure 2, especially near the times when the
message amplitude is zero. The fine detail differs from period to period of the
message. This is because the ratio of the two frequencies µ and ω has been made
non-integral.
Although the message and the carrier are periodic waveforms (sinusoids), the
DSBSC itself need not necessarily be periodic.
2 when, as here, there is only one component either side of the carrier, they are better described as side
frequencies. With a more complex message there are many components either side of the carrier, from
whence comes the term sidebands.
DSBSC generation A1 - 35
block diagram
A block diagram, showing how eqn. (1) could be modelled with hardware, is shown
in Figure 3 below.
AUDIO A.cosµ t DSBSC

OSC.
µ
E . cos µ t . cos ω t
CARRIER B.cos ω t
ω
Figure 3: block diagram to generate eqn. (1) with hardware.
viewing envelopes
This is the first experiment dealing with a narrow band signal. Nearly all modulated
signals in communications are narrow band. The definition of 'narrow band' has
already been discussed in the chapter Introduction to Modelling with TIMS.
You will have seen pictures of DSB or DSBSC signals (and amplitude modulation -
AM) in your text book, and probably have a good idea of what is meant by their
envelopes 3. You will only be able to reproduce the text book figures if the
oscilloscope is set appropriately - especially with regard to the method of its
synchronization. Any other methods of setting up will still be displaying the same
signal, but not in the familiar form shown in text books. How is the 'correct method'
of synchronization defined ?
With narrow-band signals, and particularly of the type to be examined in this and the
modulation experiments to follow, the following steps are recommended:
1) use a single tone for the message, say 1 kHz.

2) synchronize the oscilloscope to the message generator, which is of fixed
amplitude, using the 'ext trig.' facility.
3) set the sweep speed so as to display one or two periods of this message on
one channel of the oscilloscope.
4) display the modulated signal on another channel of the oscilloscope.
With the recommended scheme the envelope will be stationary on the screen. In all
but the most special cases the actual modulated waveform itself will not be stationary
- since successive sweeps will show it in slightly different positions. So the display
within the envelope - the modulated signal - will be 'filled in', as in Figure 4, rather
than showing the detail of Figure 2.
3 there are later experiments addressed specifically to envelopes, namely those entitled Envelopes, and
Envelope Recovery.
Figure 4: typical display of a DSBSC, with the message from
which it was derived, as seen on an oscilloscope. Compare with
Figure 2.
multi-tone message
The DSBSC has been defined in eqn. (1), with the message identified as the low
frequency term. Thus:
message = cosµt ........ 4
For the case of a multi-tone message, m(t), where:

n
m(t ) = ∑ a cos µ t
i =1
i i
........ 5
then the corresponding DSBSC signal consists of a band of frequencies below ω, and
a band of frequencies above ω. Each of these bands is of width equal to the
bandwidth of m(t).
The individual spectral components in these sidebands are often called
sidefrequencies.
If the frequency of each term in the expansion is expressed in terms of its difference
from ω, and the terms are grouped in pairs of sum and difference frequencies, then
there will be ‘n’ terms of the form of the right hand side of eqn. (3).
Note it is assumed here that there is no DC term in m(t). The presence of a DC term
in m(t) will result in a term at ω in the DSB signal; that is, a term at ‘carrier’
frequency. It will no longer be a double sideband suppressed carrier signal. A
special case of a DSB with a significant term at carrier frequency is an amplitude
modulated signal, which will be examined in an experiment to follow.
A more general definition still, of a DSBSC, would be:
DSBSC = E.m(t).cosωt ........ 6
where m(t) is any (low frequency) message. By convention m(t) is generally

understood to have a peak amplitude of unity (and typically no DC component).
linear modulation
The DSBSC is a member of a class known as linear modulated signals. Here the
spectrum of the modulated signal, when the message has two or more components, is
the sum of the spectral components which each message component would have
produced if present alone.
For the case of non-linear modulated signals, on the other hand, this linear addition
does not take place. In these cases the whole is more than the sum of the parts. A
frequency modulated (FM) signal is an example. These signals are first examined in
the chapter entitled Analysis of the FM spectrum, within Volume A2 - Further &
Advanced Analog Experiments, and subsequent experiments of that Volume.
spectrum analysis
In the experiment entitled Spectrum analysis - the WAVE ANALYSER, within Volume
A2 - Further & Advanced Analog Experiments, you will model a WAVE ANALYSER.
As part of that experiment you will re-examine the DSBSC spectrum, paying
particular attention to its spectrum.
EXPERIMENT
the MULTIPLIER
This is your introduction to the MULTIPLIER module.
Please read the section in the chapter of this Volume entitled Introduction to
modelling with TIMS headed multipliers and modulators. Particularly note the
comments on DC off-sets.
preparing the model

Figure 3 shows a block diagram of a system suitable for generating DSBSC derived
from a single tone message.
Figure 5 shows how to model this block diagram with TIMS.
SCOPE
ext. trig.
Figure 5: pictorial of block diagram of Figure 3
The signal A.cosµt, of fixed amplitude A, from the AUDIO OSCILLATOR,

represents the single tone message. A signal of fixed amplitude from this oscillator
is used to synchronize the oscilloscope.
The signal B.cosωt, of fixed amplitude B and frequency exactly 100 kHz, comes
from the MASTER SIGNALS panel. This is the TIMS high frequency, or radio,
signal. Text books will refer to it as the 'carrier signal'.
The amplitudes A and B are nominally equal, being from TIMS signal sources.
They are suitable as inputs to the MULTIPLIER, being at the TIMS ANALOG
REFERENCE LEVEL. The output from the MULTIPLIER will also be, by design
of the internal circuitry, at this nominal level. There is no need for any amplitude
adjustment. It is a very simple model.
T1 patch up the arrangement of Figure 5. Notice that the oscilloscope is

triggered by the message, not the DSBSC itself (nor, for that matter,
by the carrier).
T2 use the FREQUENCY COUNTER to set the AUDIO OSCILLATOR to about

1 kHz
Figure 2 shows the way most text books would illustrate a DSBSC signal of this
type. But the display you have in front of you is more likely to be similar to that of
Figure 4.
signal amplitude.
T3 measure and record the amplitudes A and B of the message and carrier
signals at the inputs to the MULTIPLIER.
The output of this arrangement is a DSBSC signal, and is given by:

DSBSC = k A.cosµt B.cosωt ...... 7
The peak-to-peak amplitude of the display is:
peak-to-peak = 2 k A B volts ...... 8
Here 'k' is a scaling factor, a property of the MULTIPLIER. One of the purposes of
this experiment is to determine the magnitude of this parameter.
Now:
T4 measure the peak-to-peak amplitude of the DSBSC
Since you have measured both A and B already, you have now obtained the
magnitude of the MULTIPLIER scale factor 'k'; thus:
k = (dsbsc peak-to-peak) / (2 A B) ...... 9
Note that 'k' is not a dimensionless quantity.
fine detail in the time domain

The oscilloscope display will not in general show the fine detail inside the DSBSC,
yet many textbooks will do so, as in Figure 2. Figure 2 would be displayed by a
single sweep across the screen. The normal laboratory oscilloscope cannot retain
and display the picture from a single sweep 4. Subsequent sweeps will all be slightly
different, and will not coincide when superimposed.
To make consecutive sweeps identical, and thus to display the DSBSC as depicted in
Figure 2, it is necessary that ‘µ’ be a sub-multiple of ‘ω’. This special condition can
be arranged with TIMS by choosing the '2 kHz MESSAGE' sinusoid from the fixed
MASTER SIGNALS module. The frequency of this signal is actually 100/48 kHz
(approximately 2.08 kHz), an exact sub-multiple of the carrier frequency. Under
these special conditions the fine detail of the DSBSC can be observed.
T5 obtain a display of the DSBSC similar to that of Figure 2. A sweep speed of,
say, 50µs/cm is a good starting point.
overload
When designing an analog system signal overload must be avoided at all times.
Analog circuits are expected to operate in a linear manner, in order to reduce the
chance of the generation of new frequencies. This would signify non-linear
operation.
A multiplier is intended to generate new frequencies. In this sense it is a non-linear
device. Yet it should only produce those new frequencies which are wanted - any
other frequencies are deemed unwanted.
4 but note that, since the oscilloscope is synchronized to the message, the envelope of the DSBSC
remains in a fixed relative position over consecutive sweeps. It is the infill - the actual DSBSC itself -
which is slightly different each sweep.
A quick test for unintended (non-linear) operation is to use it to generate a signal
with a known shape -a DSBSC signal is just such a signal. Presumably so far your
MULTIPLIER module has been behaving ‘linearly’.
T6 insert a BUFFER AMPLIFIER in one or other of the paths to the

MULTIPLIER, and increase the input amplitude of this signal until
overload occurs. Sketch and describe what you see.
bandwidth
Equation (3) shows that the DSBSC signal consists of two components in the
frequency domain, spaced above and below ω by µ rad/s.
With the TIMS BASIC SET of modules, and a DSBSC based on a 100 kHz carrier,
you can make an indirect check on the truth of this statement. Attempting to pass the
DSBSC through a 60 kHz LOWPASS FILTER will result in no output, evidence that
the statement has some truth in it - all components must be above 60 kHz.
A convincing proof can be made with the 100 kHz CHANNEL FILTERS module 5.
Passage through any of these filters will result in no change to the display (see
alternative spectrum check later in this experiment).
Using only the resources of the TIMS BASIC SET of modules a convincing proof is
available if the carrier frequency is changed to, say, 10 kHz. This signal is available
from the analog output of the VCO, and the test setup is illustrated in Figure 6
below. Lowering the carrier frequency puts the DSBSC in the range of the
TUNEABLE LPF.
oscilloscope
trigger
AUDIO
OSC. A.cosµ t DSBSC
µ =1kHz
TUNEABLE
LPF
vco B.cos ω t
ω =10kHz
Figure 6: checking the spectrum of a DSBSC signal
T7 read about the VCO module in the TIMS User Manual. Before plugging the
VCO in to the TIMS SYSTEM UNIT set the on-board switch to VCO.
Set the front panel frequency range selection switch to ‘LO’.
T8 read about the TUNEABLE LPF in the TIMS User Manual and the
Appendix A to this text.
5 this is a TIMS ADVANCED MODULE.
T9 set up an arrangement to check out the TUNEABLE LPF module. Use the
VCO as a source of sinewave input signal. Synchronize the
oscilloscope to this signal. Observe input to, and output from, the
TUNEABLE LPF.
T10 set the front panel GAIN control of the TUNEABLE LPF so that the gain
through the filter is unity.
T11 confirm the relationship between VCO frequency and filter cutoff frequency
(refer to the TIMS User Manual for full details, or the Appendix to
this Experiment for abridged details).
T12 set up the arrangement of Figure 6. Your model should look something like
that of Figure 7, where the arrangement is shown modelled by TIMS.
ext. trig
Figure 7: TIMS model of Figure 6
T13 adjust the VCO frequency to about 10 kHz
T14 set the AUDIO OSCILLATOR to about 1 kHz.
T15 confirm that the output from the MULTIPLIER looks like Figures 2 and/or 4.
Analysis predicts that the DSBSC is centred on 10 kHz, with lower and upper
sidefrequencies at 9.0 kHz and 11.0 kHz respectively. Both sidefrequencies should
fit well within the passband of the TUNEABLE LPF, when it is tuned to its widest
passband, and so the shape of the DSBSC should not be altered.
T16 set the front panel toggle switch on the TUNEABLE LPF to WIDE, and the
front panel TUNE knob fully clockwise. This should put the passband
edge above 10 kHz. The passband edge (sometimes called the ‘corner
frequency’) of the filter can be determined by connecting the output
from the TTL CLK socket to the FREQUENCY COUNTER. It is given
by dividing the counter readout by 360 (in the ‘NORMAL’ mode the
dividing factor is 880).
T17 note that the passband GAIN of the TUNEABLE LPF is adjustable from the
front panel. Adjust it until the output has a similar amplitude to the
DSBSC from the MULTIPLIER (it will have the same shape). Record
the width of the passband of the TUNEABLE LPF under these
conditions.
Assuming the last Task was performed successfully this confirms that the DSBSC
lies below the passband edge of the TUNEABLE LPF at its widest. You will now
use the TUNEABLE LPF to determine the sideband locations. That this should be
possible is confirmed by Figure 8 below.
dB
50
Figure 8: the amplitude response of the TUNEABLE LPF

superimposed on the DSBSC spectrum.
Figure 8 shows the amplitude response of the TUNEABLE LPF superimposed on the
DSBSC, when based on a 1 kHz message. The drawing is approximately to scale. It
is clear that, with the filter tuned as shown (passband edge just above the lower
sidefrequency), it is possible to attenuate the upper sideband by 50 dB and retain the
lower sideband effectively unchanged.
T18 make a sketch to explain the meaning of the transition bandwidth of a

lowpass filter. You should measure the transition bandwidth of your
TUNEABLE LPF, or instead accept the value given in Appendix A to
this text.
T19 lower the filter passband edge until there is a just-noticeable change to the
DSBSC output. Record the filter passband edge as fA. You have
located the upper edge of the DSBSC at (ω + µ) rad/s.
T20 lower the filter passband edge further until there is only a sinewave output.
You have isolated the component on (ω - µ) rad/s. Lower the filter
passband edge still further until the amplitude of this sinewave just
starts to reduce. Record the filter passband edge as fB.
T21 again lower the filter passband edge, just enough so that there is no
significant output. Record the filter passband edge as fC
T22 from a knowledge of the filter transition band ratio, and the measurements fA
and fB , estimate the location of the two sidebands and compare with
expectations. You could use fC as a cross-check.
alternative spectrum check

If you have a 100kHz CHANNEL FILTERS module, or from a SPEECH module,
then, knowing the filter bandwidth, it can be used to verify the theoretical estimate of
the DSBSC bandwidth.
speech as the message

If you have speech available at TRUNKS you might like to observe the appearance
of the DSBSC signal in the time domain.
Figure 9 is a snap-shot of what you might see.
Figure 9: speech derived DSBSC
TUTORIAL QUESTIONS
Q1 in TIMS the parameter ‘k’ has been set so that the product of two sinewaves,
each at the TIMS ANALOG REFERENCE LEVEL, will give a
MULTIPLIER peak-to-peak output amplitude also at the TIMS
ANALOG REFERENCE LEVEL. Knowing this, predict the expected
magnitude of 'k'
Q2 how would you answer the question ‘what is the frequency of the signal
y(t) = E.cosµt.cosωt’ ?
Q3 what would the FREQUENCY COUNTER read if connected to the signal

y(t) = E.cosµt.cosωt ?
Q4 is a DSBSC signal periodic ?
Q5 carry out the trigonometry to obtain the spectrum of a DSBSC signal when
the message consists of three tones, namely:
message = A1.cosµ1t + A2.cosµ2t + A3 cosµ3t
Show that it is the linear sum of three DSBSC, one for each of the
individual message components.
Q6 the DSBSC definition of eqn. (1) carried the understanding that the message
frequency µ should be very much less than the carrier frequency ω.
Why was this ? Was it strictly necessary ? You will have an
opportunity to consider this in more detail in the experiment entitled
Envelopes (within Volume A2 - Further & Advanced Analog
Experiments).
TRUNKS
If you do not have a TRUNKS system you could obtain a speech signal from a
SPEECH module.
APPENDIX
TUNEABLE LPF tuning information

Filter cutoff frequency is given by:
NORM range: clk / 880
WIDE range: clk / 360
See the TIMS User Manual for full details.
EC3611: Communication Engineering I (2017)
Lab 02 – PRELAB
I. Introduction
The labs will be conducted and assessed during each lab session (within 2 hours)
Be prepare for the lab. Preview the lab documents. It is the responsibility of the students to go through the
materials provided.
Finish the prelab questions before you come to the lab session. Please hand in your answers to the
instructors assigned during the lab sessions
II. Learning Outcomes

 Introduction to PCM Encoding and Decoding.
III. Prelab
1. What is PCM ?
2. What is Nyquist Theorem?
3. Draw the block diagram of the PCM Encoding and Decoding?
4. What is the Minimum number of Bits required for Encoding (State the equation using the
error Percentage –p). If the error percentage p = 5%, Calculate the minimum number of
bits (Round to the nearest value)
5. Below you have given a Matlab Code for the PCM Encoding. Answer the questions by
referring to the code.
function [y Bitrate MSE Stepsize QNoise]=pcm(A,fm,fs,n)
%A=amplitute of cosine signal
%fm=frequency of cosine signal
%fs=sampling frequency
%n= number of bits per sample
t=0:1/(100*fm):1;
x=A*cos(2*pi*fm*t);
%---Sampling-----
ts=0:1/fs:1;
xs=A*cos(2*pi*fm*ts);
%xs Sampled signal
%--Quantization---
x1=xs+A;
x1=x1/(2*A);
L=(-1+2^n); % Levels
x1=L*x1;
xq=round(x1);
r=xq/L;
r=2*A*r;
r=r-A;
%r quantized signal
%----Encoding---
y=[];
y_row=[];
q_recovered=[];
recovered_signal=[];
for i=1:length(xq)
d=dec2bin(xq(i),n);
y=[y double(d)-48];
end
%-------Decoding----------
y_row = vec2mat(y,n);
q_recovered = bi2de(y_row,'left-msb');
q_recovered = transpose(q_recovered);
q_recovered = (q_recovered/L).*2*A - A;
butter_cut_off = (fm + 100);
[n,Wn] = buttord(butter_cut_off/(fs/2),(butter_cut_off+100)/(fs/2),3,60);
[B,A] = butter(n,Wn,'low');
recovered_signal = filter(B,A,q_recovered);
figure(1)
plot(t,x,'linewidth',2)
title('Sampling')
ylabel('Amplitute')
xlabel('Time t(in sec)')
hold on
stem(ts,xs,'r','linewidth',2)
hold off
legend('Original Signal','Sampled Signal');
figure(2)
stem(ts,x1,'linewidth',2)
title('Quantization')
ylabel('Levels L')
hold on
stem(ts,xq,'r','linewidth',2)
plot(ts,xq,'--r')
%plot(t,(x + A)*L/(2*A),'--b')
grid
hold off
legend('Sampled Signal','Quantized Signal');
figure(3)
stairs([y y(length(y))],'linewidth',2)
title('Encoding')
ylabel('Binary Signal')
xlabel('bits')
axis([0 length(y) -1 2])
grid
figure(4)
subplot(2,1,1);
plot(t,x);
title('Original message')
subplot(2,1,2);
plot(ts,recovered_signal);
title('Recovered message')
1. Study the given Matlab code.

2. What is the Matlab syntax buttord and butter mean? (Lengthy answers are not required)
Reference
This code has been modified from the code available in the following reference.
 "Pulse Code Modulation - File Exchange - MATLAB Central". In.mathworks.com. N.p.,

2017. Web. 11 Apr. 2017.
PCM ENCODING
ACHIEVEMENTS: Introduction to pulse code modulation (PCM) and the PCM

ENCODING procedure using MATLAB.
PREREQUISITE : An understanding of theory of sampling, quantizing and encoding

procedure.
PCM Encoding
The input to the PCM ENCODER module is an analog signal. In this lab we restrict the
signal to be a low pass one. The maximum allowable signal bandwidth may depend upon the
sampling rate allowed by the encoder.
Following procedure may be followed to develop MATLAB program for PCM encoding
/decoding.
 The input analog signal is sampled periodically. Minimum sampling rate can be
determined by using the Nyquist criteria
 Implement the sample and hold operation
 Each sample amplitude is compared with a finite set of amplitude levels. That is an
uniform quantizing is considered
 Each quantizing level is assigned a number, starting from zero for the lowest (most
negative) level, with the highest number being (L-1), where L is the available number
of levels.
 Each sample is assigned a digital code word (PCM sequence) representing the number
associated with the quantizing level which is closest to the sample amplitude. The
number of bits ‘n’ in the digital code word will depend upon the number of quantizing
levels, i.e. , n = log2(L).
Experiment
T1 : Use the MATLAB program given and observe the sampled, quantized and encoded
PCM sequences for the given sampling frequency, quantization levels and signal frequency.
T2: A sinusoidal signal which has a frequency of 200 Hz
 What is the minimum sampling rate ?
The quantization distortion is specified not to exceed 3% of the peak-to-peak analog signal.
 What is the number of bits per sample?
 How many quantizing levels are there ?
Suppose sampling frequency of the PCM encoder is 2500 Hz.
 What is the data rate?
T3 : What are the bit patterns of selected parameters (Draw maximum 5 bit patterns)?
T4: Include the results in your log book.
Discussion
Q1:
Discuss the aliasing phenomena.
Q2:
Discuss the necessity of non linear quantization.
1
EC3611 Communication Engineering I

Assignment 1
Due date: Thursday, March 15, 2018 (by 4:00 pm)
Q UESTION 1: S WITCHING DSB-SC SIGNALS (7.5 × 4 POINTS , 4 SUB - QUESTIONS ,

MODULATION FOR
7.5 POINTS EACH )
Consider the square pulse train w(t) centered at t = 0 with fundamental frequency fc shown in Fig. 1.
Let the baseband signal be denoted as m(t), which is bandlimited to B Hz and f c B. By multiplying
Fig. 1. The square pulse train signal.
the slow-varying message signal m(t) with the above pulse train, one may also increase the changing
rate of the resulting signal. This suggests that w(t) can be used for modulation. Suppose we want to
perform the DSB-SC modulation using m(t), so that the spectrum of m(t) is moved to the frequency
band centered at fc .
1) Let one period of w(t) be specified as (T = 1/fc )

1, |t| < T4
w(t) = (in one period)
0, T4 < |t| < T2
and consider w(t) as the periodic expansion of the above. Find out the Fourier series of w(t).
2) Write down the produce signal m(t) · w(t) and identify the desired item for the DSB-SC signal.
3) Explain how to get rid of the un-wanted items in the above product signal. Draw a system diagram
for the whole process to produce the desired DSB-SC signal.
4) Write down the Fourier transform of the DSB-SC signal.
2
Q UESTION 2 (7.5 × 4 POINTS , 4 SUB - QUESTIONS , 7.5 POINTS EACH )

In the modulation scheme shown in Fig. 2, let m 1 (t) and m2 (t) be two band-limited base-band signals
with bandwidth B (Hz), where B fc .
1) Write an expression for the multiplexed signal s(t) (see part (a) of Fig. 2).
2) For a coherent receiver (shown in (b) of Fig. 2), find an expression for the signals after the product
modulator, both in the upper (in-phase) branch and in the lower (quadrature) branch.
3) Show that the final outputs are the same as given in the figure (part (b) of Fig. 2).
4) What are the requirements on the low-pass filter in part (b) of Fig. 2?
3
Fig. 2.
4
Q UESTION 3 (10 × 4 POINTS , 4 SUB - QUESTIONS , 10 POINTS EACH )

Given the following angle-modulated voltage signal:
s(t) = cos(1000πt) cos[5 sin(100πt)] − sin(1000πt) sin[5 sin(100πt)],
Determine the following:
1) the average power developed across a 50-ohm load resistor;
2) the carrier frequency;
3) the instantaneous frequency;
4) the maximum and minimum instantaneous frequencies.
Hint: Use the formula

cos α cos β − sin α sin β = cos(α + β).
Assignment 2
1. A signal in the frequency range 300 to 3300 Hz is limitted to peak-to-peak swing of
10 V. It is sampled at 8000 samples/s and the samples are quantized to 64 evenly
spaced levels.Calculate and compare the bandwidth and ratio of peak signal power
to rms quantization noise if the quantized samples are transmitted either as binary
pulses or as four-level pulses. Assume that the system bandwidth is defined by the
main spectral lobe of the signal.
2. Let x denote the maximum possible voltage magnitude of a signal. Then the peak
signal power is designated by x2 . Now suppose you are given a system where
the peak-to-peak value of the signal is Vpp and the samples are quantized to L
evenly spaced levels, which are symmetrical about zero. Show that the ratio of the
peak-signal power to the peak-quantization noise power (S/N )peak ≈ L2 , whenever
L 1.
3. In the compact disc (CD) digital audio system, an analog signal is digitized so that
the ratio of the peak-signal power to the peak-quantization noise power is at least
96 dB. The sampling rate is 44.1 kilosamples/s.
(a) How many quantization levels of the analog signal are needed for (S/N )peak =
96 dB. (Hint: see Question 1 above).
(b) How many bits/sample are needed for the number of levels found in part (a)
above.
(c) What is the data rate in bits/s?
4. Bipolar pulse signal, si (t) (i = 1, 2), of amplitude 1V, −1V are received in the
presence of AWGN that has a variance of 0.1 V2 . The minimum probability of error
decision making criterion is given by
p(z|s1 ) H1 P (s2 )
R .
p(z|s2 ) H2 P (s1 )
(a) Show that " #

p(z|s1 ) z(a1 − a2 ) a21 − a22
= exp − ,
p(z|s2 ) σ02 2σ02
where a1 denote the signal component of the correlator detector with reference
signal +1 V, a2 denote the signal component of the correlator detector with
reference signal −1 V, and σ02 is the noise variance after the receiver.
(b) Determine the optimum detection threshold, γ0 , for matched filter detection if
the a priori probabilities are: P (s1 ) = 0.5; P (s1 ) = 0.7; P (s1 ) = 0.2.
5. Unipolar RZ signalling constitutes a orthogonal signalling, see Figure 3.12 (a) of the
Sklar. In particular, the unipolar RZ signalling is as follows:
s1 (t) = A 0≤t≤T for binary 1
s2 (t) = 0 0≤t≤T for binary 0

(a) Compute the bit error rate of unipolar RZ with AWGN and a correlator de-
tector assuming that a priori probabilities are equal.
(b) Assume Gaussian noise with noise-power spectral density N0 = 10−8 Watts/Hz.
Moreover, assume that the received pulses have and amplitude of 100 mV. If
the bit-error probability specification is PB = 10−3 , find the largest date rate
that can be transmitted using this system.

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DEPARTMENT OF ELECTRICAL & COMPUTER

At the end of the module student will be able to:

 Midterm Examination 20% LO1

Learning Recommended Texts

MODULE ADMINISTRATION PROCEDURE

4. Introduction to Basic Digital Communication Systems

Any type of plagiarism is not allowed.

End of Module Outline

Year 3, Semester 1: 06thFeb. – 14th May, 2018

Feb. 6 The detailed

(M. Ding) Linear modulation

8 Soft week Mid-term

April 30- Rescheduling

Chathuranga Weeraddana, Minhua Ding

• Lab, assignment and exam policies:

• Introduction to communication systems

• Today communication systems are ubiquitous.

Early days: the development of communication technology followed that

• Discovery of electromagnetism by Oersted and Ampère in the 1820s

• Hertz’s veriﬁcation of Maxwell’s postulation (1873) in 1880s, predicting

• Invention of diode by Fleming in 1904 and of triode ampliﬁer by de

Subsequently: the development of communication technology drove that

• The success of telephone, patented by Alexander Graham Bell in 1876

• Telephone (late 19th century)

• Wireless telegraphy (late 19th and early 20th century)

• Radio (early 20th century)

• Public Switched Telephone Network (PSTN)

• the aggregate of the world’s circuit-switched telephone networks

• LAN: computers/devices in a limited geographical area

• a communication network where the last link is wireless

• satellites cover very large areas

• A satellite navigation system

• Channel: the physical medium for message delivery

• Destination: user of the information

• Analog communication systems deal with analog signals in the baseband

• Mainly modulation theory (starting from next week)

• A signal is a real (or complex) valued function of one or more real

• continuous-time and discrete-time

(a) analog, continuous-time

• A signal g(t) is said to be periodic if for some positive constant T0 > 0,

g(t) = g(t + T0), for all t

• A signal is said to be aperiodic if there is no such value of T0 that

• the instantaneous power of a signal g(t) is p(t) = |g(t)|2

Practically meaningful only when it is ﬁnite

• Signals such as periodic signals have inﬁnite energy. A suitable measure

This limit can be 0, e.g., for some signals of ﬁnite energy.

• Calculating the power of g(t) = C cos(ω0t + θ)

g(t) = C1 cos(ω1t + θ1) + C2 cos(ω2t + θ2), ω1 = ω2

Answer: 12 (C12 + C22)

• A signal with ﬁnite energy is an energy signal.

• A signal with ﬁnite power is a power signal.

• g(t)δ(t) = g(0)δ(t) (assuming g(t) continuous at t = 0)

• Convolution (for regular continuous function g(t))

• Periodic signals can be expressed1 as a sum of sinusoids whose frequencies

where the coeﬃcients can be calculated as

• Fourier series using complex exponentials

A periodic square wave of period 2π is deﬁned as follows. Over the

We want to express it as:

Periodic square wave, when T0 → ∞

Let g(t) denote a signal (real or complex).

• The inverse Fourier transform is given by

Ͳ2/ʏ Ͳ1/ʏ 0 1/ʏ 2/ʏ

• Signal Bandwidth: the diﬀerence between the highest (signiﬁcant)

‫ݔ‬ሺ‫ݐ‬ሻ ൌ ݉ሺ‫ݐ‬ሻ ሺʹߨ݂௖ ‫ݐ‬ሻ ݁ሺ‫ݐ‬ሻ ‫ݒ‬௢ ሺ‫ݐ‬ሻ