Slide 1 - EEE 4597

Communication
Engineering I
EEE 4403
Dr. Khondokar Habibul Kabir

Room 202
Contact: 01768-451-451
habib@iut-dhaka.edu
Osaka
University
Dr. K. Habibul Kabir
Communication Engineering I
Recommended Text Books

“Modern Digital and Analog Communication Systems”,
Mainly
By - B. P. Lathi, Zhi Ding,
Oxford Series in Electrical and Computer Engineering,
(ISBN-10: 0195331451, ISBN-13: 978-0195331455)
Analog and Digital

Communications (Schaum's
Outlines)
by Hwei P. Hsu
2nd Edition
3rd Edition 4th Edition
“Digital & Analog Communication Systems”

If Required
By Leon W. Couch
Prentice Hall; 7 edition (July 28, 2006)
(ISBN-10: 0131424920, ISBN-13: 978-0131424920)
Communication Systems; an Introduction to Signals and Noise in Electrical
Communication
By A. Bruce Carlson
McGraw-Hill Education; International 2nd edition (1986)
(ISBN-10: 0071005609, ISBN-13: 978-0071005609)
2
Before modern Times, messages were carried by runners, carrier pigeons, flags, lights, and
fires.
Modern Communication Technique is served by Electrical Communication Systems which

can transmit signals over much longer distances (even to distant planets and galaxies) and
at the light speed.
Analog Communication Technique

Digital Communication Technique
Cellular Phone, GPRS, 3G, 4G
The Internet
Wireless Communication
Communication Technologies improve productivity and energy conversion.
Teleconference, Email, E-commerce
Television, radio, newspaper improve a lot to spread up to date information

as quick as possible.
3
Communication System
Source: The source originates a message, such as a human voice, a television picture, a
teletype message, or data.
Input Transducer: If the data is nonelectrical (human voice, teletype message, television
picture), it must be converted by an input transducer into an electrical waveform of the
original signal referred to as the message signal or baseband signal.
Transmitter: The transmitter modifies the baseband signal for efficient transmission.
Channel: The channel is a medium--such as wire, coaxial cable, a waveguide, an optical

fiber, or a radio link through which the transmitter output is sent.
4
Communication System
Receiver: The receiver reprocesses the signal received from the channel by undoing the
signal modifications made at the transmitter and the channel.
Output Transducer: The receiver output is fed to the output transducer, which converts
the electrical signal to its original form, i.e., the message.
Destination: The destination is the unit to which the message is communicated.
5
Input Input Transmitted Signal Received Output signal Output

Message Signal Wired or Signal Message
Wireless
Input Output
Transmitter Channel Receiver
Transducer Transducer
Source Destination
Distortion
and noise
Distortion and noise

Signals passing through communication channels not only experience channel distortions but
also are corrupted along the path by undesirable noises.
Signal waveforms are distorted because of physical phenomenon such as

• frequency dependent gains,
• multipath effects, and
• Doppler shift effect.
The signal attenuation increases with the length of the channel.
6
The waveform is distorted because of different amounts of attenuation and phase shift suffered
by different frequency components of the signal. This type of distortion, called linear distortion.
For example, a square pulse is rounded or “spread out” during transmission over a low-pass
channel.
Linear distortion can be partly corrected at the receiver by an equalizer with gain and phase
characteristics complementary to those of the channel.
The channel may also cause nonlinear distortion through attenuation that varies with the signal
amplitude.
Such nonlinear distortion can also be partly corrected by a complementary equalizer at the
receiver.
Channel distortion, if known (or predicted), can be also pre-compensated by transmitters by

applying channel-dependent pre-distortions.
7
These interfering signals are random and are unpredictable from both external and internal
sources.
External noise includes

• interference signals transmitted on nearby channels,
• human made noise generated by faulty contact switches,
• automobile ignition radiation,
• fluorescent lights or natural noise from lightning,
• microwave ovens, and cellphone emission, etc.
With proper care in system design, external noise can be minimized or even eliminated in
some cases.
Internal noise results from

• thermal motion of charged particles in conductors,
• random emission, and
• diffusion or recombination of charged carriers in electrical devices.
Proper care can reduce the effect of internal noise but can never eliminate it. Noise is one of
the basic factors that set limits on the rate of communication.
8
Signal-to-noise ratio (SNR): The signal-to-noise ratio (SNR) is defined as the ratio of
signal power to noise power.
In analog and digital communications, signal-to-noise ratio, (often written SNR or S/N) is a
measure used in science and engineering that compares the level of a desired signal to the
level of background noise.
The ratio is usually measured in decibels (dB). A ratio higher than 1:1 (greater than 0 dB)
indicates more signal than noise.
Signal-to-noise ratio is defined as the power ratio between a signal (meaningful information)
and the background noise (unwanted signal):
Psignal
SNR 
Pnoise
Where, P is average power. Both signal and noise power must be measured at the same or
equivalent points in a system, and within the same system bandwidth.
9

If the signal and the noise are measured across the same impedance, then the SNR can be
obtained by calculating the square of the amplitude ratio:
2
Psignal  Asignal 
SNR   
Pnoise A
 noise 
Where, A is root mean square (RMS) amplitude (e.g, RMS voltage or current). Because many
signals have a very wide dynamic range, SNRs are often expressed using the logarithmic
decibel scale. In decibels, the SNR is defined as
 Psignal 
SNRdB  10 log10    Psignal ,dB  Pnoise,dB
 Pnoise 
which may equivalently be written using amplitude ratios as
2
 Asignal   Asignal 
SNRdB  10 log10    20 log10  
A
 noise  A
 noise 
Measuring signal-to-noise ratios requires the selection of a representative or reference
signal. In audio engineering, the reference signal is usually a sine wave at a standardized
nominal level, such as 1 kHz at +4 dBu (1.228 VRMS).
10
If the incoming signal strength is Vs in microvolts, and the noise level is Vn in microvolts,
then the signal-to-noise ratio, SNR or S/N, in decibels is given by the formula
V
SNR  20 log10 Vns
If Vs = Vn, then SNR = 0. In this situation, the signal is unreadable, because the noise
level severely competes with it. In digital communications, this will probably cause a
Vs = Vn
reduction in data speed because of frequent errors that require the source
(transmitting) terminal to resend some packets of data.
If Vn = 1.0 microvolt and the signal is much weaker but still above the noise, say 1.30
Vs > Vn microvolts, then SNR = 20 log10(1.30) = 2.28 dB
which is a marginal situation. There might be some reduction in data speed under these
conditions.
Ideally, Vs is greater than Vn, so SNR is positive.
As an example, suppose that Vs = 10.0 microvolts and Vn = 1.0 microvolt. Then SNR = 20
Vs >> Vn
log10(10.0) = 20.0 dB
which results in the signal being clearly readable.
If Vs is less than Vn, then SNR is negative.
In this type of situation, reliable communication is generally not possible unless steps are
Vs < Vn
taken to increase the signal level and/or decrease the noise level at the destination
(receiving) terminal.
11

The concepts of signal-to-noise ratio and dynamic range are closely related.
Dynamic range: Dynamic range measures the ratio between the strongest un-distorted
signal on a channel and the minimum recognized signal, which for most purposes is the noise
level.
ct
On the other hand, SNR measures the ratio by taking the average of arbitrary signal level
al (not necessarily the most powerful signal possible) and noise.
l,
of The channel distorts the signal, and noise accumulates along the path.
Worse yet, the signal strength decreases while the noise level increases with distance from
the transmitter.
l Thus, the SNR is continuously decreasing along the length of the channel.
Amplification of the received signal to eliminate the attenuation is failed

because the noise associated with the signal is being amplified in the same proportion, and
the SNR remains, at best, unchanged.
12
Baseband Signals may be directly transmitted over a suitable channel. Baseband

signals produced by various information sources are not always suitable for direct
transmission over a given channel.
When signal and channel frequency bands do not match exactly, channel cannot be
moved. Hence, messages must be moved to the right channel frequency bandwidth.
These signals are usually further modified to facilitate transmission. This conversion
process is known as modulation.
In this process, the base band signal is used to modify (i.e., modulate), some parameter
of a high-frequency carrier signal.
A carrier is a sinusoid of high frequency, and one of its parameters--such as amplitude,

frequency, or phase -is varied in proportion to the baseband signal m(t).
13
Accordingly, we have
• Amplitude modulation (AM),
• Angle Modulation
• Frequency modulation (FM), or
• Phase modulation (PM).
In AM, the carrier amplitude In FM, the carrier frequency

varies in proportion to m(t), varies in proportion to m(t).
At the receiver, the modulated signal must pass through a reverse process called demodulation in
order to reconstruct the baseband signal.
14
For efficient radiation of electromagnetic energy, the radiating antenna should be on

the order of one-tenth or more of the wavelength of the signal radiated.
For many baseband signals, the wavelengths are too large for reasonable antenna
dimensions.
For example, the power in a speech signal is concentrated at frequencies in the range of
100 to 3000 Hz. The corresponding wavelength is 100 to 3000 km. This long
wavelength would necessitate an impracticably large antenna.
Instead, we modulate a high-frequency carrier, thus translating the signal spectrum to

the region of carrier frequencies that corresponds to a much smaller wavelength.
For example, a 1-MHz carrier has a wavelength of only 300m and requires an antenna
whose size is on the order of 30 m.
15
Simultaneous Transmission of Several Signals
Consider the case of several radio stations broadcasting audio baseband signals directly,
without any modification.
They would interfere with each other because the spectra of all the signals occupy more
or less the same bandwidth.
Thus, it would be possible to broadcast from only one radio or television station at a time.
This is wasteful because the channel bandwidth may be much larger than that of the
signal. One way to solve this problem is to use modulation.
16
Modulation often refers to a process that modifies the message signal into a specific
frequency band that can be readable by the physical channel (e.g., voice band telephone
modems).
Baseband communication: The communication that does not use Modulation
Carrier Communication: The communication that uses modulation.
17
Frequency-Shifting Property
FT :
If 
G( )   g (t )e  jt dt
g (t )  G ( ) 
Inverse FT
then
1 
g (t )e jct  G (  c )
g(t)  
2 
G( )e jt d 
By definition, 
F  g (t )e jct    g (t )e jct e  jt dt


  g (t )e  j ( c )t dt

 G (  c )
This property states that multiplication of a signal by a factor e jspectrum

shifts the
ct
of
that signal by ω = ωc.
We also have
g (t )e  jct  G (  c )
18
The Fourier transform relates the function's

time domain, shown in red, to the function's
frequency domain, shown in blue. The
component frequencies, spread across the
frequency spectrum, are represented as peaks
in the frequency domain
19
Because factor eisjcnot

t
a real function that can be generated.
Hence, frequency shifting in practice is achieved by multiplying g(t) by a sinusoid.
This can be seen from the fact that
1
g (t ) cos ct  [ g (t )e jct  g (t )e  jct ]
2
It follows that
g (t )e jct  G (  c )
1
g (t ) cos ct  [G (  c )  G (  c )] g (t )e  jct  G (  c )
2
This shows that the multiplication of a signal g(t) by a sinusoid of frequency ωc, i.e., cos
ωct shifts the spectrum G(ω) by ± ωc.
This also implies that multiplication of a sinusoid cos ωct by g(t) simply modulate the
sinusoid amplitude.
This type of modulation is known as amplitude modulation.

• The sinusoid cosωct is called the carrier,
• the signal g(t) is the modulating signal, and
• the signal g (t) cosωct is the modulated signal.
20
Amplitude modulation: Double Side Band (DSB)
Amplitude modulation is characterized by the fact that the amplitude A of the carrier A
cos(ωct+θc) is varied in proportion to the baseband signal m(t), the modulating signal.
The frequency ωc and the phase θc are constant.

We can assume θc = 0 without a loss of generality.
If the carrier amplitude A is made directly proportional to the modulating signal m(t),
the modulated signal is m(t)cosωct.
As was seen earlier this type of modulation simply shifts the spectrum of m(t) to the
carrier frequency. Thus, if then
m(t )  M ( )
1
m(t ) cos ct  [ M (  c )  M (  c )]
2
21
Recall that M(ω-ωc) is M(ω) shifted to the right by ωc and M(ω+ωc) is M(ω) shifted to
the left by ωc .
• Thus, the process of modulation shifts the spectrum of the modulating signal to
the left and the right by ωc.
Note also that if the bandwidth of m(t) is B Hz, then the bandwidth of the modulated
signal is 2B Hz.
22
We also observe that the modulated signal spectrum centered at ωc is composed of two
parts:
a portion that lies above ωc, known as the upper sideband (USB), and
a portion that lies below ωc, known as the lower sideband (LSB).
Similarly, the spectrum centered at - ωc has upper and lower sidebands.
Hence, this is a modulation scheme with double sidebands. This scheme does not
contain a discrete component of the carrier frequency ωc. For this reason it is called
double-sideband suppressed carrier (DSB-SC) modulation.
23
The relationship of B to ωc shows that ωc ≥ 2πB in order to avoid the overlap of the spectra centered
at ωc and - ωc .
• If ωc ≤ 2πB, these spectra overlap and the information of m(t) is lost in the process of
modulation, which makes it impossible to get back m(t) from the modulated signal m(t)cosωct.
The bandwidth of a channel is the range of frequencies that it can transmit with reasonable fidelity.
For example, if a channel can transmit a signal whose frequency components occupy a range from 0
up to a maximum of 5000 Hz (5kHz), the channel bandwidth B is 5 kHz.
24
Demodulation/ Detection
The DSB-SC modulation translates or shifts the frequency spectrum to the left and the right by ωc
(that is, at +ωc and -ωc).
To recover the original signal m(t) from the modulated signal, it is necessary to retranslate the
spectrum to its original position.
The process of recovering the signal from the modulated signal (retranslating the spectrum to its
original position) is referred to as demodulation, or detection.
25
Observe that if the modulated signal spectrum is shifted to the left and to the right by ωc (and
multiplied by one-half), we obtain the spectrum as shown,
This contains the desired baseband spectrum plus an unwanted spectrum at ±2ωc. The latter can be
suppressed by a low pass filter.
A low-pass filter is a filter that passes signals with a frequency lower than a certain cutoff
frequency and attenuates signals with frequencies higher than the cutoff frequency.
Attenuation: In electrical engineering and communications, attenuation affects the propagation of

waves and signals in electrical circuits, optical fibers, and in air (radio waves). Attenuation is gradual loss
in intensity (reducing force) of flux through a medium.
26
Thus, demodulation, which is almost identical to modulation, consists of multiplication

of the incoming modulated signal m(t)cosωct by a carrier cosωct followed by a low pass
filter.
Modulation
Demodulation
27
We can verify this conclusion directly in the time domain by observing that the signal e(t) is
Demodulation
e(t )  m(t ) cos 2 ct
1 2 cos 2 x  (1  cos 2 x)
 [m(t )  m(t ) cos 2ct ]
2
Therefore, the Fourier transform of the signal e(t) is
1 1
E ( )  M ( )  [ M (  2c )  M (  2c )]
2 4
28
This shows that the signal e(t) consists of two components (1/2)m(t) and
(1/2)m(t)cos2ωct , with their spectra as shown
The spectrum of the second component, being a modulated signal with carrier
frequency 2ωc, is centered at ±2ωc. Hence, this component is suppressed by the low pass
filter and we can achieve the desired original signal.
The desired component (1/2)M(ω), being a low pass spectrum (centered at ω = 0),
passes through the filter unharmed, resulting in the output (1/2)m(t).
29
We can get rid of the inconvenient fraction 1/2 in the output by using a carrier 2cosωct
instead of cosωct. This strategy, which does not affect general conclusions.
This method of recovering the baseband signal is called synchronous detection, or

coherent detection,
This method of detection uses a carrier of exactly the same frequency (and phase) as the
carrier used for modulation.
Thus, for demodulation, it needs to generate a local carrier at the receiver in frequency
and phase coherence (synchronism) with the carrier used at the modulator.
30
Modulators
Modulation can be achieved in several ways.
Multiplier Modulators:
Here modulation is achieved directly by multiplying m(t) by cosωct using an analog multiplier
whose output is proportional to the product of two input signals.
Such a multiplier may be obtained from a variable-gain amplifier in which the gain parameter
(such as the ß of a transistor) is controlled by one of the signals, say, m(t). When the signal cosωct
is applied at the input of this amplifier, the output is proportional to m(t)cosωct .
A low-pass filter is a filter that passes signals with a frequency lower than
a certain cutoff frequency and attenuates signals with frequencies higher
than the cutoff frequency.
A high-pass filter is an electronic filter that passes signals with a

frequency higher than a certain cutoff frequency and attenuates
signals with frequencies lower than the cutoff frequency.
A band-pass filter is a device that passes frequencies within a certain range

and rejects (attenuates) frequencies outside that range. These filters can be
created by combining a low-pass filter with a high-pass filter.
31
Nonlinear Modulators:
Modulation can also be achieved by using nonlinear devices, such as a semiconductor diode
or a transistor.
Let the input-output characteristics of either of the nonlinear elements be approximated by

a power series:
x(t) y(t)
y (t )  ax(t )  bx 2 (t ) Nonlinear
where x(t) and y(t) are the input and the output, respectively, of the nonlinear elements.
y1 (t )  ax1 (t )  bx12 (t )
cosωct + m(t)
m(t) x1(t) y1(t)
 Nonlinear
+ z (t )  y1 (t )  y2 (t )
+ z(t)
 Band pass
filter (±ωc)
4bm(t)cosωct
-
-
cosωct
 x2(t)
Nonlinear
y2(t)
cosωct - m(t) y2 (t )  ax2 (t )  bx2 2 (t )

32
Nonlinear Modulators:
The summer output z(t) is given by
z (t )  y1 (t )  y2 (t )
 [ ax1 (t )  bx12 (t )]  [ ax2 (t )  bx2 2 (t )]
Substituting the two inputs x1(t) = cosωct + m(t) and x2(t) = cosωct - m(t) in this equation
yields,
z (t )  2am(t )  4bm(t ) cos ct (Home work)
y1 (t )  ax1 (t )  bx12 (t )
cosωct + m(t)
m(t) x1(t) y1(t)
 Nonlinear
+ z (t )  y1 (t )  y2 (t )
+ z(t)
 Band pass
filter (±ωc)
4bm(t)cosωct
-
-
cosωct
 x2(t)
Nonlinear
y2(t)
cosωct - m(t) y2 (t )  ax2 (t )  bx2 2 (t )

33
The spectrum of m(t) is centered at the origin, whereas the spectrum of m(t)cosωct is
centered at ±ωc. Consequently, when z(t) is passed through a bandpass filter tuned to ωc,
the signal am(t) is suppressed and the desired modulated signal 4bm(t)cosωct passes
through unharmed.
In this circuit there are two inputs: m(t) and cosωct . The summer output z(t) does not
contain one of the inputs, the carrier signal cosωct.
Consequently, the carrier signal does not appear at the input of the final bandpass filter.
The circuit acts as a balanced bridge for one of the inputs (the carrier). Circuits which have
this characteristic are called balanced circuits.
The nonlinear modulator is an example of a class of modulators known as balanced

modulators.
This circuit is balanced with respect to only one input (the carrier); the other input m(t)
still appears at the final bandpass filter, which must reject it. For this reason, it is called a
single balanced modulator.
34
Demodulator/Detector of DSB-SC Signals
As discussed earlier, demodulation of a DSB-SC signal is identical to modulation.
For multiplier demodulator, at the receiver, we multiply the incoming signal by a local
carrier of frequency and phase in synchronism with the carrier used at the modulator.
The product is then passed through a lowpass filter. The only difference between the
modulator and the demodulator is the output filter.
In the nonlinear modulator, the multiplier output is passed through a banpass filter
tuned to ωc, whereas in the demodulator, the multiplier output is passed through a
low-pass filter. Therefore, all the modulators discussed earlier can also be used as
demodulators, provided the bandpass filters at the output are replaced by low-pass
filters of bandwidth B.
For demodulation, the receiver must generate a carrier in phase and frequency
synchronism with the incoming carrier. These demodulators are called synchronous or
coherent (also homodyne) demodulators.
35
Frequency Mixer or Converter

It is often desirable to translate or shift the modulated signal to a new frequency band. For
example, in most commercial AM radio receivers, the received radio-frequency (RF)signal [540
to 1600 kHz] is shifted to the intermediate frequency (IF) (455kHz) band for processing. The
received signal, now translated to a fixed IF, can easily be amplified, filtered, and demodulated.
A frequency mixer, or frequency converter, used to change the carrier frequency of a modulated
signal m(t)cosωct from ωc to some other frequency ωl.
This can be done by multiplying m(t)cosωct by 2cosωmixt ,

where ωmix = ωc + ωl or ωc - ωl, and then bandpass-filtering the product.
The product x(t)

x(t )  2m(t ) cos ct cos mixt
=m(t )[cos(c  mix )t  cos(c  mix )t ]
If we select ωmix = ωc – ωl,
x(t )  m(t )[cos l t  cos(2c  l )t ]
If we select ωmix = ωc + ωl,
x(t )  m(t )[cos l t  cos(2c  l )t ]
36
Frequency Mixer or Converter

In either case, a bandpass filter at the output, tuned to ωl, will pass the m(t)cosωlt and suppress
the other term, yielding the output m(t)cosωlt
(Assuming that ωc - ωl ≥ 2πB and ωl ≥ 2πB so that various spectra do not overlap).
Thus, the carrier frequency has been translated to ωl from ωc .
The operation of frequency mixing, or frequency conversion (also known as heterodyning), is

basically a shifting of spectra by an additional ωmix. This is equivalent to the operation of
modulation with a modulating carrier frequency (the mixer oscillator frequency ωmix) that
differs from the incoming carrier frequency by ωl. Any one of the modulators discussed earlier
can be used for frequency mixing.
When we select the local carrier frequency ωmix = ωc + ωl, the operation is called up-
conversion,
when we select the local carrier frequency ωmix = ωc - ωl, the operation is called down-
conversion.
37
Home work Problem:
x(t) y(t)
x(t )  [ A  m(t )]cos 2 c t

1 1
 [ A  m(t )]  [ A  m(t )]cos 2c t
2 2
The first term is a lowpass signal because-its spectrum is centered at to ω = 0. The lowpass
filter allows this term to pass, but suppresses the second term, whose spectrum is centered
at +-2ωc. Hence the output of the lowpass filter is
y(t) = A + m(t)
When this signal is passed through a dc block, the dc term A is suppressed yielding the
output m(t). This shows that the system can demodulate AM signal regardless of the value of
A. This is a synchronous or coherent demodulation. '
38
Solved Problem
39
Solved Problem
40
Book Name: Schaum Analog and Digital Communcations

(PDF version has been given)
Assignment and Home work Problem:
Chapter 2: Problem 2.1 to 2.33 for self study and understanding filter
characteristics. (Page 36 to 46)
Chapter 3: Problem 3.1to 3.21for understanding Amplitude Modulation

characteristics. (Page 60 to 74)
3.5, 3.7, 3.9, 3.10, 3.13, 3.14, 3.15
41
Please read each question carefully before answering.
Answer each question or each sub-question according to the given mark.
Answer each question to the point by arranging/managing Total Exam time.
Wish you all the best
42
Bandwidth Efficient Amplitude Modulation
The DSB spectrum has two sidebands:

the upper sideband (USB) and the lower sideband (LSB), both containing the complete
information of the baseband signal m(t).
As a result, for a baseband signal m(t) with bandwidth B Hz,

DSB modulations require twice the radio frequency bandwidth
to transmit.
To improve the spectral efficiency of amplitude modulation,

there exist two basic schemes to either utilize or remove the
100% spectral redundancy.
• Single sideband (SSB) modulation, which removes either the LSB or the USB that uses
only bandwidth of B Hz for one message signal m(t).
• Quadrature amplitude modulation (QAM), which utilizes the spectral redundancy by

sending two messages over the same bandwidth of 2B Hz.
43
Amplitude modulation: Single Sideband (SSB)
A scheme in which only one sideband is transmitted is known as single-sideband

(SSB) transmission, which requires only one-half the bandwidth of the DSB signal.
An SSB signal can be coherently (synchronously) demodulated just like DSB-SC

signals.
For example, multiplication of a USB signal

by cosωct shifts its spectrum to the left and
right by ωc, yielding the spectrum.
Low-pass filtering of this signal yields the

desired baseband signal.
The case is similar with LSB signals.
44
Since the demodulation of SSB identical to that of DSB-SC signals, the transmitters can
now utilize only half the DSB-SC signal bandwidth without any additional cost to the
receivers.
Since no additional carrier accompanies the modulated SSB signal, the resulting
modulator outputs are known as suppressed carrier signals (SSB-SC).
45
Generation of SSB Signals:
Frequency Discrimination Method:
The straightforward way to generate an SSB signal is to generate a DSB signal first and then
suppress one of the sidebands by filtering. This is known as the frequency discrimination
method. In practice, this method is not easy because the filter must have sharp cutoff
characteristics.
46

Phase-Shift Method:
Another method for generating an SSB signal, known as the phase shift method.
Hilbert-Transformer flips the signal phase by -90° for positive signal and +90° for negative
signal.
Thus, if we delay the phase of every component of m(t) by π/2 (without changing its
amplitude), the resulting signal is mh(t), the Hilbert transform of m(t). Therefore, a Hilbert
transformer is an ideal phase shifter that shifts the phase of every spectral component by
-π/2.
47

Phase-Shift Method:
Another method for generating an SSB signal, known as the phase shift method.
The box marked - π/2 is a π/2 phase shifter that delays the phase of every frequency
component by π/2 . Hence, it is a Hilbert transformer.
48
Home work Problem: Using the single-tone modulating signal cosωm(t), verify that the output
of the SSB generator by phase shifter is indeed an SSB signal, and show that an upper-
sideband (USB) or a lower-sideband (LSB) signal results from subtraction or addition at the
summation junction. Also demonstrate the coherent demodulation of this SSB signal.
The redrawing of SSB generator by phase shifter with a single-tone modulating signal is
Recall that the Hilbert transform

delays the phase of each spectral
component by π/2.
Here,
m(t )  cos mt

cos(c t  )  sin ct
2

mh (t )  cos(mt  )  sin mt
2
Hence,
y (t )  xSSB (t )  cos mt cos ct sin mt sin ct
 cos(c  m )t Hence, SSB generator by phase shifter is indeed an SSB signal
49
Thus, with subtraction we have

y (t )  xUSB (t )  cos(c  m )
And with addition we have
y (t )  xLSB (t )  cos(c  m )
Finally, the coherent demodulation of the

SSB tone modulation is can be achieved by
xSSB (t ).2 cos c t  2 cos(c  m )t cos ct

 cos mt  cos(c  m )t
Which can be sent to a low pass filter to

retrieve the message tone cosωm(t).
50
Demodulation of SSB Signals:

Demodulation of SSB signals can he achieved easily by using the coherent detector as used
in the DSB demodulation, that is, by multiplying X SSB(t) by a local carrier and passing the
resulting signal through a low-pass filter.
51
QUADRATURE AMPLITUDE MODULATION (QAM)
The DSB signals occupy twice the bandwidth required for the baseband.
This disadvantage can be overcome by transmitting two DSB signals using carriers of the
same frequency but in phase quadrature.
In the figure, the boxes labeled -π/2 are phase shifters, which delay the phase of an input
sinusoid by -π/2 rad.
If the two baseband signals to be transmitted are m1(t) and m2(t), the corresponding QAM signal
φQAM(t), the sum of the two DSB-modulated signals is
QAM  m1 (t ) cos c t  m2 (t ) sin c t
52
Both modulated signals occupy the same band.
Yet two baseband signals can be separated at the

receiver by synchronous detection using two local
carriers in phase quadrature.
This can be shown by considering the multiplier output

x1(t) of the upper arm of the receiver:
x1 (t )  2QAM (t ) cos ct

 2[ m1 (t ) cos ct  m2 (t ) sin ct ]cos ct
 m1 (t )  m1 (t ) cos 2ct  m2 (t ) sin 2ct
The last two terms are suppressed by the low-pass filter, yielding the desired output m1(t).
Similarly, the output of the lower receiver branch can be shown to be m2(t).
53
This scheme is known as quadrature amplitude modulation (QAM) or quadrature

multiplexing.
Thus, two baseband signals, each of bandwidth B Hz, can be transmitted simultaneously
over a bandwidth 2B by using DSB transmission and quadrature multiplexing.
The upper channel is also known as the in-phase (I) channel and the lower channel is the
quadrature (Q) channel.
Demonstration
54
VESTIGIAL SIDEBAND (VSB) AMPLITUDE

MODULATION
As seen earlier, the generation of SSB signals is rather difficult.
The selective-filtering method demands dc null in the

modulating signal spectrum.
A phase shifter required in the phase-shift method is

sometime unrealizable, or realizable only
approximately.
The generation of DSB signals is much simpler, but

requires twice the signal bandwidth.
A vestigial-sideband (VSB), also called asymmetric

sideband system is a compromise between DSB and
SSB.
In this modulation scheme, one sideband is passed almost completely, whereas just a trace, or
vestige, of the other sideband is retained.
55

MODULATION
VSB inherits the advantages of DSB and SSB but avoids their disadvantages at a small cost.
VSB signals are relatively easy to generate, and, at the same

time, their bandwidth is only (typically 25%, i.e., about
1.25 times) greater than that of SSB signals.
In VSB, instead of rejecting one sideband completely (as

in SSB), a gradual cutoff of one sideband is accepted.
The baseband signal can be recovered exactly by a synchronous detector in conjunction

with an appropriate equalizer filter Ho(ω) at the receiver output.
If a large carrier is transmitted along with the VSB signal (VSB+C), the baseband signal can
be recovered by an envelope (or a rectifier) detector.
56
Use of VSB in Broadcast Television
VSB is a clever compromise between SSB and DSB, which makes it very attractive for
television broadcast systems. The reasons are:
The baseband video signal of television occupies an enormous bandwidth of 4.5 MHz, and
hence, a DSB signal needs a bandwidth of 9 MHz. VSB requires 5.6 MHz.
It would seem desirable to use SSB in order to conserve the bandwidth.
Unfortunately, this creates some problems.
First, the baseband video signal has sizable power in the low-frequency region, and
consequently it is difficult to suppress one sideband completely.
Second, for a broadcast receiver, an envelope detector is preferred over a synchronous one
in order to reduce the receiver cost. However, in SSB+C (SSB with large carrier) case, the
required carrier amplitude is much larger than that of AM, and, consequently, SSB+C has a
very low power efficiency.
Moreover, use of SSB will increase the receiver cost.
57

MODULATION
If the transfer function of vestigial shaping filter that produces VSB from DSB is Hi(ω), then the
resulting VSB signal spectrum is
VSB ( )  [ M (  c )  M (  c )]H i ( ) Eq.(A)
This VSB shaping filter Hi(ω) allows the transmission of one sideband, but suppresses the
other sideband, not completely, but gradually.
This makes it easy to realize such a filter, but the transmission bandwidth is now somewhat
higher than that of the SSB (where the other sideband is suppressed completely).
The bandwidth of the VSB signal is typically 25 to 33% higher than that of the SSB signals.
58

MODULATION
We require that m(t) be recoverable from φVSB(t) using synchronous demodulation at the
receiver. This is done by multiplying the incoming VSB signal φVSB(t) by 2cosωct. The
product e(t) is given by
e(t )  2VSB (t ) cos ct  [VSB (  c )  VSB (  c )]
The signal e(t) is further passed through the low-pass equalizer filter of transfer function
Ho(ω). The output of the equalizer filter is required to be m(t). Hence, the output signal
spectrum is given by
M ( )  [VSB (  c )  VSB (  c )]H o ( )
59

MODULATION
Substituting Eq.(A) into this equation and eliminating the spectra at -+ 2ωc [suppressed by a
low-pass filter Ho(ω)l, we obtain
M ( )  M ( )[ H i (  c )  H i (  c )]H o ( )
1
H o ( )  ||  2 B
H i (  c )  H i (  c )
Note that because Hi(ω) is a bandpass filter, the terms Hi(ω+-ωc) contain low-pass
components.
If we choose Hi(ω) such that
H i (  c )  H i (  c )  1 ||  2 B
The output filter is just a simple low-pass filter with transfer function
Ho(ω) = 1
over the baseband |ω| = 2πB
60
ORDINARY AMPLITUDE MODULATION (AM) (DSB-

LC)
For the suppressed carrier scheme (DSB-SC), a receiver must generate a carrier in
frequency and phase synchronise with the carrier at the transmitter that may be located
hundreds or thousands of miles away. This calls for a sophisticated receiver and could be
quite costly.
The other alternative is for the transmitter to transmit a carrier Acosωc(t) along with the
modulated signal m(t)cosωc(t) so that there is no need to generate a carrier at the receiver. In
this case the transmitter needs to transmit much larger power, which makes it rather
expensive.
In point-to-point communications, where there is one transmitter for each receiver,
substantial complexity in the receiver system can be justified, provided it results in a large
enough saving in expensive high-power transmitting equipment.
On the other hand, for a broadcast system with a multitude of receivers for each transmitter, it
is more economical to have one expensive high-power transmitter and simpler, less
expensive receivers.
The second option (transmitting a carrier along with the modulated signal) is the obvious
choice for this case. This is the so-called AM (amplitude modulation).
61
ORDINARY AMPLITUDE MODULATION (AM) (DSB-

LC)
An ordinary amplitude-modulated signal is generated by adding a large carrier
signal to the DSB signal. It is called (DSB-LC). The ordinary AM signal (or simply
AM signal) has the form
x AM (t )  m(t ) cos ct  A cos ct  [ A  m(t )]cos c t
The spectrum of XAM(t) is given by
1
X AM ( )  [ M (  c )  M (  c )]   A[ (  c )   (  c )]
2
An example of an AM signal, in
both time domain and frequency
domain 
62
Demodulation of AM Signals:
The advantage of AM over DSB modulation is that a very simple scheme, known as envelope
detection, can be used for demodulation if sufficient carrier power is transmitted.
If A is large enough, the envelope (amplitude) of the modulated waveform given by A +

m(t) will be proportional to m(t).
Demodulation in this case simply reduces to the detection of
the envelope of a modulated carrier with no dependence on
the exact phase or frequency of the carrier. If A is not large
enough, then the envelope of XAM(t) is not always proportional
to m(t).
min{m(t)}
Thus, the condition for demodulation of AM by an

envelope detector is
A  m(t )  0 for all t

or , A  min{m(t )}
where min{m(t)} is the minimum value of m(t).
63
Envelope Detector:
A simplest form of an envelope detector consisting of a diode and a resistor-capacitor
combination is shown bellow:
The operation of the envelope detector is as follows: During the positive half-cycle of the input
signal, the diode is forward-biased and the capacitor C charges up rapidly to the peak value of
the input signal. As the input signal falls below its maximum, the diode turns off. This is
followed by a slow discharge of the capacitor through resistor R until the next positive half-
cycle, when the input signal becomes greater than the capacitor voltage and the diode turns on
again. The capacitor charges to the new peak value, and the process is repeated.
64
Envelope Detector:
A simplest form of an envelope detector consisting of a diode and a resistor-capacitor
combination is shown bellow:
The operation of the envelope detector is as follows: During the positive half-cycle of the input
signal, the diode is forward-biased and the capacitor C charges up rapidly to the peak value of
the input signal. As the input signal falls below its maximum, the diode turns off. This is
followed by a slow discharge of the capacitor through resistor R until the next positive half-
cycle, when the input signal becomes greater than the capacitor voltage and the diode turns on
again. The capacitor charges to the new peak value, and the process is repeated.
65
Envelope Detector:
For proper operation of the envelope detector, the discharge time constant RC must be
chosen properly. In practice, satisfactory operation requires two conditions:
fc >> bandwidth of m(t), that is ωc>> ω and
A  m(t )  0 for all t.
Capacitor discharge between positive peaks causes a ripple signal of frequency ωc in the output.
This ripple can be reduced by choosing a larger time constant RC so that the capacitor
discharges very little between the positive peaks (RC>>1/ωc). Picking RC too large, however,
would make it impossible for the capacitor voltage to follow a fast declining envelop. Because
the maximum rate of AM envelop decline is dominated by the bandwidth B of the message
signal m(t).
The design criterion of RC should be
1 1 1
RC  or 2 B  c
c 2 B RC
The envelop detector output is vc(t)= A + m(t) with a ripple of frequency ωc. The dc value A can
be blocked out by a capacitor or a simple RC high pass filter. The ripple may be reduced
further by another (low pass) RC filter.
66
Amplitude Modulation Index:
The modulation index μ for AM is defined as

min{m(t )}

A
As A  min{m(t )} ,
the condition for demodulation of AM by an envelope detector can be
 1
When μ > 1, the carrier is μ<1

said to be over-
modulated, resulting in
envelope distortion.
μ=1
μ>1
67
Angle modulation: Phase modulation (PM) and

Frequency modulation (FM)
Angle modulation encompasses phase modulation (PM) and frequency modulation

(FM) and refers to the process by which the phase angle of a sinusoidal carrier wave is
varied according to the message signal m(t).
In amplitude modulation the spectrum of the modulated signal is essentially the

translated message spectrum, and the transmission never exceeds twice the message
bandwidth.
In angle modulation, the spectral components of the modulated wave form are not
related in any simple fashion to the message spectrum.
Furthermore, superposition does not apply, and the bandwidth of the angle-modulated
usually much greater than twice the message bandwidth.
The increase in bandwidth and complexity is compensated by the improved performance

in the face of noise and interference.
68
For angle modulation, the modulated carrier is represented by
xc (t )  A cos[c t   (t )]
where A and ω are constants and the phase angle (ϕ) is a function of the message signal m(t).
If we rewrite the equation,
xc (t )  A cos  (t )
where,  (t )  c t   (t )
Then we can define the instantaneous radian frequency of xC(t), denoted by ωi, as
d (t ) d (t )
i   c 
dt dt
Note that when Φ(t) = constant, then ωi = ωc.
The function ϕ(t) is known as the instantaneous phase deviation and dϕ(t)/dt is known as
instantaneous frequency deviation of xc(t).
The quantity ∆ω defined by
 | i  c |max
is called the maximum (Or peak) radian frequency deviation of the angle-modulated
signal.
69
PHASE MODULATION (PM) AND FREQUENCY MODULATION (FM)
In PM, the instantaneous phase deviation of the carrier is proportional to the message signal;
that is,
 (t )  m(t )  (t )  k p m(t )
where kp is the phase deviation constant, expressed in radians per unit of m(t).
In FM, the instantaneous frequency deviation of the carrier is proportional to the message
signal; that is
d (t ) d (t )
 m(t )  k f m(t )
dt dt
t
or ,  (t )  k f  m( )d    (to )
to
where kf is the frequency deviation constant, expressed in radians per second per unit of m(t),
and ϕ(t0) is the initial phase angle at t=t0.
Thus, we can express the angle-modulated signal as
xPM (t )  A cos[ct  k p m(t )]
t
xFM (t )  A cos[c t  k f  m( ) d  ]

70
By definition d (t ) d (t )
i   c 
we have dt dt
t
for FM [ (t )  k f  m( )d    (to )]
to
i  c  k f m(t )
for PM [ (t )  k p m(t )]
dm(t )
i  c  k p
dt
In FM, ωi varies linearly with the modulating

signal.
Thus, in PM, the instantaneous frequency ωi

varies linearly with the derivative of the
modulation signal m(t).
71
FOURIER SPECTRA OF ANGLE-MODULATED SIGNALS

An angle-modulated carrier can be represented in exponential form by writing Eq.
xc (t )  A cos[c t   (t )]
xc (t )  Re( Ae j (ct  (t )) )  Re( Ae jct e j ( t ) )

where Re means the “real part of.” Because of this representation, the angle modulation is
also referred to as the exponential modulation.
Expanding the equation in a power series yields

 j t   2 (t ) n  (t )
n
 
xc (t )  Re  Ae 1  j (t ) 
c
 j   
  2! n!  
  2 (t )  3 (t ) 
=A cos c t   (t )sin c t  cos c t  sin ct 
 2! 2! 
Thus, the angle-modulated signal consists of an unmodulated carrier cosωct plus various
amplitude modulated terms, such as ϕ(t)sinωct, ϕ2(t)cosωct, ϕ3(t)sinωct, ..., and so on.
Hence, its Fourier spectrum consists of an unmodulated carrier plus spectra of ϕ(t), ϕ2(t),
ϕ3(t) ..., and so on, centered at ωct.
It is clear that the Fourier spectrum of an angle-modulated signal is not related to the
message signal spectrum in any simple way, as was the case in AM.
72
NARROWBAND ANGLE MODULATION
If
then the angle modulated signal can be approximated by [neglecting all higher-power term
of ϕ(t)]
This signal is called the narrowband (NB) angle-modulated signal. Thus.
This equation of xc(t) indicate that a narrowband angle-modulated signal contains an

unmodulated carrier plus a term in which ϕ(t) [a function of m(t)] multiplies a π/2 (rad)
phase-shifted carrier.
This multiplication generates a pair of sidebands, and if ϕ(t) has a bandwidth WB, the
bandwidth of an NB angle-modulated signal is 2WB.
73
Modulation Index.
If the message signal m(t) is a pure sinusoid, that is.
Then Eqs.of ϕ(t) for both PM and FM give
 (t )  k p m(t )
d (t )
 k f m(t )
dt
t
or ,  (t )  k f  m( )d    (to )
to
The parameter ß is known as the modulation index for angle modulation and is the
maximum value of phase deviation for both PM and FM.
Note that ß is defined only for sinusoidal modulation and it can be expressed as
where ∆ω, is the maximum frequency deviation.
74
GENERATION OF ANGLE-MODULATED
SIGNALS
Narrowband Angle-Modulated Signals:
The generation of narrowband angle-modulated signals is easily accomplished in view of the
following Equations.
Neglecting all higher-power term of ϕ(t) the angle modulated signal becomes
The narrowband (NB) angle-modulated signal becomes
75
Wideband Angle-Modulated Signal
There are two methods of generating wideband (WB) angle-modulated signals; the indirect
method and the direct method.
Indirect Method
In this method, an NB angle-modulated signal is produced by using first narrow band angle
modulation technique and then converted to a WB angle-modulated signal by using
frequency multipliers.
The frequency multiplier multiplies the argument of the input sinusoid by n.
Thus, if the input of a frequency multiplier is
Then the output of the frequency multiplier is
76
Use of frequency multiplication normally increases the carrier frequency to an impractically

high value.
To avoid this, a frequency conversion (using a mixer or DSB modulator) is necessary to shift
the spectrum.
77
Direct Method
In the direct method of generating an FM signal, the modulating signal directly controls the
carrier frequency.
A common method used for generating FM directly is to vary the inductance or capacitance
of a tuned electric oscillator.
Any oscillator whose frequency is controlled by the modulating signal voltage is called a
voltage controlled oscillator (VCO).
The main advantage of direct FM is that large frequency deviations are possible, and thus less
frequency multiplication is required.
The major disadvantage is that the carrier frequency tends to drift, and so additional
circuitry is required for frequency stabilization.
78
DEMODULATION 0F ANGLE-MODULATED
SIGNALS
Demodulation of an FM signal requires a system that produces an output proportional
to the instantaneous frequency deviation of the input signal, Such a system is called a
frequency discriminator.
If the input to an ideal discriminator is an angle-modulated signal
then the output of the discriminator is
yd (t )  i  d (t ) d (t )
i   c 
dt dt
where kd is the discriminator sensitivity.
For FM, ϕ(t) is given by
so that equation of yd(t) becomes
79
The frequency discriminator also can be used to demodulate PM signals. For PM, ϕ(t)
is given by
Then yd(t) becomes
Integration of the discriminator output yields a signal that is proportional to m(t).
A demodulator for PM can therefore be implemented as an FM demodulator followed

by an Integrator.
80
A simple approximation to the ideal discriminator is an ideal differentiator followed by an

envelope detector. If the input to the differentiator is
then the output of the differentiator is
The signal x’c(t) both amplitude and angle-modulated. The envelope of x’c(t) is
The output of the envelope detector is

d (t ) d (t )
i   c 
dt dt
which is the instantaneous frequency of the xc.
81
Digital Communication
The trend in the design of new communication systems has been toward increasing the
use of digital techniques.
Digital communications offer several important advantages compared to analog

communications, for example. higher performance, greater versatility, and higher
security,
To transmit analog message signals. such as voice and video signals, by digital means,
the signal has to be converted to a digital signal.
This process is known as the analog-to-digital conversion (AD), or sometimes referred

as digital pulse code modulation (PCM).
82
Message are digital or analog. Digital messages are ordered combination of finite symbols or
codewords. For example,
• Printed English consists of 26 letters, 10 numbers, a space, and several punctuation marks.
Thus, a text document written in English is a digital message, because it is from the ASCII
keyboard of 128 symbols.
• Human speech is also a digital message, because it is made up from a finite vocabulary in
a language (without considering the pronunciation and variation of pitch).
• Music notes are also digital, even though the music sound itself is analog. ***
• A Morse-coded telegraph message is a digital message constructed from a set of only two
symbols- dash and dot. Its a binary messages,
• A digital message constructed with M symbols is called an M-ary message.
Analog messages, on the other hand, are characterized by data whose values vary over a
continuous range and are defined for a continuous range of time.
• For example, the temperature, atmospheric pressure.
• A piece of music recorded by a pianist is also an analog signal. ***
• Similarly, any particular speech waveform has amplitudes that vary over a continuous
range is analog.
The human speech signal from a microphone contains all the details of pronunciation and variation of
pitch and is therefore an analog signal.
83
Message extraction at the receiver is often easier from digital signals than from analog
signals.
If the distortion is within limits, we can recover the digital signal data without error because
we need only to make a simple binary decision as to whether the received pulse is positive
or negative.
A/2
t Transmitted signal (Source) (010110001--)
-A/2
Received distorted signal (without noise).
Received distorted signal (with noise).
Regenerated signal by recognizing zero

crossing value, i.e., - or + value (delayed).
at Receiving end
A digital communication system is more robust than an analog communication system in

the sense that it can better withstand noise and distortion (as long as they are within a limit).
84
Please read these sections

from the book
85
The frequency spectrum of a signal indicates relative magnitudes of various frequency

components.
Sampling Theorem: The sampling theorem states that if the highest frequency in the signal
spectrum is B (in hertz), the signal can be reconstructed from its samples, taken at a rate not
less than 2B samples per second.
This means that in order to transmit the information in a continuous-time signal, we need
only transmit its samples.
Unfortunately, the sample values are still not digital because they lie in a continuous range
and can take on any one of the infinite values in the range.
86
This difficulty is neatly resolved by quantization technology; where each sample is

approximated, or "rounded off," to the nearest quantized level
Amplitudes of the signal m(t) lie in the range (-mp, mp), which is partitioned into L intervals,
each of magnitude ∆v = 2 mp /L.
Each sample is now approximated to one of the L numbers. The information is thus digitized.
mp m(t)
Quantized samples of m(t)
t
2 mp /L
-mp
87
The quantized signal is an approximation of the original one. We can improve the
accuracy of the quantized signal to any desired degree by increasing the number of
levels L.
• For voice signals, for example, L = 8 or 16 is sufficient.

• For commercial use, L = 32 is a minimum, and for telephone communication, L =
128 or 256 is commonly used.
88
During each sampling interval, we transmit one

quantized sample, which takes on one of the L
values. This requires L distinct waveforms.
This may be constructed, for example, by using …

a basic rectangular pulse of amplitude A/2 and
its multiples (for instance, ±A/2, ± 3A/2, ±
5A/2 ..... ±[(L - 1)A/2], to form L distinct …
waveforms to be assigned to the L values to be
transmitted.
Amplitudes of any two of these waveforms are

separated by at least A to guard against noise
interference and channel distortion.
89
Digit Binary equivalent Pulse code waveform
The second alternative is to use combinations of only

two basic pulses, A/2 and -A/2.
A sequence of four such pulses gives 2 x 2 x 2 x 2 = 16

distinct patterns.
We can assign one pattern to each of the 16 quantized

values to be transmitted. Each quantized sample is now
coded into a sequence of four binary pulses.
This is the so-called binary case, where signaling is

carried out by means of only two basic pulses (or
symbols).
The binary case is of great practical importance because of its simplicity and ease of detection.
Virtually all digital communication today is binary.
This scheme of transmitting data by digitizing and then using pulse codes to transmit the digitized
data is known as pulse-code modulation (PCM).
90
PULSE CODE MODULATION (PCM)
The essential processes of PCM are sampling, quantizing, and encoding.
Sampling is the process in which a continuous-time signal is sampled by measuring its

amplitude at discrete instants.
Representing the sampled values of the amplitude by a finite set of levels is called quantizing.
Designating each quantized level by a code is called encoding.
While sampling converts a continuous-time signal to a discrete-time signal, quantizing

converts a continuous amplitude sample to a discrete-amplitude sample. Thus, sampling and
quantizing operation transform an analog signal to a digital signal.
The quantizing and encoding operations are usually performed in the same circuit, which is
called an analog-to-digital (AD) converter.
91
We can use various audio signals to modulate different carrier frequencies, thus
translating each signal to a different frequency range.
If the various carriers are chosen sufficiently far apart in frequency, the spectra of the
modulated signals will not overlap and thus will not interfere with each other.
At the receiver, one can use a tunable band-pass filter to select the desired station or
signal.
This method of transmitting several signals simultaneously is known as frequency-

division multiplexing (FDM).
Here the bandwidth of the channel is shared by various signals without any overlapping.
92
Example:
FDM 4 users
frequency
Time Channel
93
Another method of multiplexing several signals is known as time-division

multiplexing (TDM).
This method is suitable when a signal is in the form of a pulse train (as in PCM).
The pulses are made narrower, and the spaces that are left between pulses are used
for pulses from other signals.
Thus, in effect, the transmission time is shared by a number of signals by

interleaving the pulse trains of various signals in a specified order.
At the receiver, the pulse trains corresponding to various signals are separated.
94
FDM versus TDM

Example:
FDM 4 users
frequency
Time Channel
TDM
frequency
Time Channel
Combination of FDM and TDM can accommodate more users.
95
T1 Time Division Multiplexing

An attempt was made to increase the capacity by frequency-division-multiplexing several
voice channels through amplitude modulation.
Unfortunately, the cables were primarily designed for the audio voice range (0 to 4 kHz) and
suffered severely from noise.
Furthermore, cross talk at high frequencies between pairs of channels on the same cable was
unacceptable.
Ironically, PCM-requiring a bandwidth several times larger than that required for FDM
signals- offered the solution.
This is because digital systems with closely spaced regenerative repeaters can work
satisfactorily on noisy, poor-high-frequency-performance lines.
The repeaters, spaced approximately 6000 feet apart, clean up the signal and regenerate new
pulses before the pulses get too distorted and noisy.
A pair of wires that used to transmit one audio signal of bandwidth 4kHz is now used to
transmit 24 time division-multiplexed PCM telephone signals with a total bandwidth of 1.544
MHz.
96
T1 Time Division Multiplexing Transmitter
Receiver
T1 Time Division Multiplexing has 24 channels. All 24 channels are sampled in a sequence.
The sampler output represents a time-division-multiplexed PAM (pulse amplitude modulated)

signal.
The multiplexed PAM signal is now applied to the input of an encoder that quantizes each sample
and encodes it into eight binary pulses- a binary code word.
97
T1 Time Division Multiplexing Transmitter
Receiver
The signal, now converted to digital form, is sent over the transmission medium.
Regenerative repeaters spaced approximately 6000 feet apart detect the pulses and transmit new
pulses.
At the receiver, the decoder converts the binary pulses into samples (decoding).
The samples are then demultiplexed (i.e., distributed to each of the 24 channels).
The desired audio signal is reconstructed by passing the samples through a low-pass filter in each
channel.
98
Synchronizing and Signaling

Binary code words corresponding to samples of each of the 24 channels are multiplexed in a
sequence.
A segment containing one code word (corresponding to one sample) from each of the 24
channels is called a frame.
Each frame has 24 × 8 = 192 information bits.
Because the sampling rate is 8000 samples per second, each frame takes 125 μs.
99

At the receiver, it is necessary to be sure where each frame begins in order to separate
information bits correctly.
For this purpose, a framing bit is added at the beginning of each frame. This makes a total of 193
bits per frame.
Framing bits are chosen so that a sequence of framing bits, one at the beginning of each frame,
forms a special pattern that is unlikely to be formed in a speech signal.
100

The sequence formed by the first bit from each frame is examined by the logic of
the receiving terminal.
If this sequence does not follow the given coded pattern (framing bit pattern), then a
synchronization loss is detected.
It takes about 0.4 to 6ms to detect and about 50ms (worst case) to reframe.
In addition to information and framing bits, we need to transmit signaling bits corresponding
to dialing pulses, as well as telephone on-hook/off-hook signals.
101
DIGITAL CARRIER SYSTEMS
The spectrum of a signal can be shifted to a higher frequency by modulating a high-

frequency sinusoid (carrier) by the baseband signal.
Two basic forms of modulation exist:

Amplitude modulation (AM) and
Angle modulation (FM or PM).
In amplitude modulation, the carrier amplitude is varied in proportion to the modulating

signal (i.e., the baseband signal).
102
DIGITAL CARRIER SYSTEMS
An unmodulated cartier cosωct
The on-off baseband signal m(t) (the

modulating signal)
When the carrier amplitude is varied in

proportion to m(t), we have the modulated
carrier m(t)cosωct
Note that the modulated signal is still an on-

off signal.
This modulation scheme of transmitting binary data is known as on-off keying (OOK) or
amplitude-shift keying (ASK).
103
If the baseband signal m(t) were polar, the

corresponding modulated signal m(t) cosωct
would appear as.
In this case, if p(t) is the basic pulse, we are

transmitting 1 by a pulse p(t)cosωct and 0 by
-p(t)cosωct = p(t)cos(ωct+π).
Hence, the two pulses are π radians apart in

phase.
The information resides in the phase of the

pulse.
For this reason this scheme is known as phase-shift keying (PSK). Note that the transmission is
still polar.
104
When the data are transmitted by varying the

frequency, we have the case of frequency shift
keying (FSK).
A 0 is transmitted by a pulse of frequency ωc0

and 1 is transmitted by a pulse of frequency
ωc1..
The information about the transmitted data

resides in the carrier frequency.
105
106
Book Name: SchaumAnalog and Digital Communcations

(PDF version has been given)
Assignment and Home work Problem:
Chapter 3: 3.7, 3.9, 3.10, 3.13, 3.14, 3.15
Chapter 4: 4.11,4.12, 4.16, 4.18, 4.19
Chapter 5: 5.12, 5.13, 5.14, 5.30, 5.31, 5.32
Problem in Slides: 50, 51, 52, 108, 109, 110, 111, 112
107
Solved Problem
108
Solved Problem
109
Solved Problem
110
Solved Problem
111
Solved Problem
112
Follow all previous years’ questions (MID + Final)
Please read each question carefully before answering.
Answer each question or each sub-question according to the given mark.
Answer each question to the point by arranging/managing Total Exam time.
Wish you all the best
113

Slide 1 - EEE 4597

Uploaded by

Document Information

Original Description:

Original Title

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Slide 1 - EEE 4597

Uploaded by

Copyright:

Available Formats

Communication

Dr. Khondokar Habibul Kabir

Recommended Text Books

Analog and Digital

3rd Edition 4th Edition

“Digital & Analog Communication Systems”

Modern Communication Technique is served by Electrical Communication Systems which

Analog Communication Technique

Television, radio, newspaper improve a lot to spread up to date information

Channel: The channel is a medium--such as wire, coaxial cable, a waveguide, an optical

Destination: The destination is the unit to which the message is communicated.

Input Input Transmitted Signal Received Output signal Output

Distortion and noise

Signal waveforms are distorted because of physical phenomenon such as

The signal attenuation increases with the length of the channel.

Distortion and noise

Channel distortion, if known (or predicted), can be also pre-compensated by transmitters by

Distortion and noise

External noise includes

Internal noise results from

Distortion and noise

Distortion and noise

Distortion and noise

Amplification of the received signal to eliminate the attenuation is failed

Baseband Signals may be directly transmitted over a suitable channel. Baseband

A carrier is a sinusoid of high frequency, and one of its parameters--such as amplitude,

In AM, the carrier amplitude In FM, the carrier frequency

For efficient radiation of electromagnetic energy, the radiating antenna should be on

Instead, we modulate a high-frequency carrier, thus translating the signal spectrum to

Simultaneous Transmission of Several Signals

Baseband communication: The communication that does not use Modulation

Carrier Communication: The communication that uses modulation.

This property states that multiplication of a signal by a factor e jspectrum

The Fourier transform relates the function's

Because factor eisjcnot

This type of modulation is known as amplitude modulation.

Amplitude modulation: Double Side Band (DSB)

The frequency ωc and the phase θc are constant.

Attenuation: In electrical engineering and communications, attenuation affects the propagation of

Thus, demodulation, which is almost identical to modulation, consists of multiplication

This method of recovering the baseband signal is called synchronous detection, or

A high-pass filter is an electronic filter that passes signals with a

A band-pass filter is a device that passes frequencies within a certain range

Let the input-output characteristics of either of the nonlinear elements be approximated by

cosωct - m(t) y2 (t )  ax2 (t )  bx2 2 (t )

The summer output z(t) is given by

cosωct - m(t) y2 (t )  ax2 (t )  bx2 2 (t )

The nonlinear modulator is an example of a class of modulators known as balanced

Demodulator/Detector of DSB-SC Signals

As discussed earlier, demodulation of a DSB-SC signal is identical to modulation.

Frequency Mixer or Converter

This can be done by multiplying m(t)cosωct by 2cosωmixt ,

The product x(t)

Frequency Mixer or Converter

Thus, the carrier frequency has been translated to ωl from ωc .

The operation of frequency mixing, or frequency conversion (also known as heterodyning), is

Home work Problem:

x(t )  [ A  m(t )]cos 2 c t

Book Name: Schaum Analog and Digital Communcations

Assignment and Home work Problem:

Chapter 3: Problem 3.1to 3.21for understanding Amplitude Modulation

3.5, 3.7, 3.9, 3.10, 3.13, 3.14, 3.15