Chapter 3

Overview of mobile and cellular radio

communication modulation and

Block diagram of general

communication system

Information
source

Converter
To
electricity

Transmiter

channel

Noise
sources

Receiver

Destination

(please refer ECS by KennedyDavis, simultaneously

Modulation

What is Modulation????

AM,FM,PM
Need of modulation
For easy propagation as electromagnetic waves with low
loss and low dispersion
Simultaneous transmission without interference from
other signals
Enables the construction of small antennas (a fraction,
usually a quarter of the wavelength)
Enables the multiplexing (combining) multiple signals
for transmission at the same time over the same carrier

Amplitude modulation
In AM, amplitude of carrier wave is varied

in proportion to instantaneous amplitude

of the modulating signal
let Vc(t)=Vc sin(ct+c) carrier signal
Vm(t)=Vm sin(mt+m) modulating
signal
Amplitude modulated wave is
V(t)=A sin

Representation of AM
A=Vc+ Vm(t)
Modulating index is given

as
m=Vm/Vc
V(t)=Vc sinct+mVc sinmt.
Sinct
Using trigonometric relation
sinA.sinB=1/2[cos(A-B)cos(A+B)
V(t)=Vcsinct+mVc/2[cos
(c- m)t-cos(c+m)t]

m=(Vmax -Vmin)/ (Vmax+Vmin)

Bandwith required=2fm

Power relation on AM
AM wave contains 3 components: carrier and 2

side bands
Amplitude of sideband depends on m,
Hence total power also depend on m
Pt=Pcar+ PLSB+PUSB
Pt=Pc(1+m /2)
Effective voltage and current
E=Ec(1+m2 /2)
I=Ic(1+m 2/2)

Modulation by several sine

waves

To calculate total power we need to first find

the modulation index

2 methods of finding modulation index
1)if v1,v2,.. Are modulatimg voltages
toltal modulating voltage is
vt=v12 +v22 +v32 +
Divide by vc
mt=m1 +m2 +m3
2)Pt=Pc(1+m2 /2)
PSBT=PSB1+PSB2+PSB3+..

Generation of AM

Fig1:AM wave

In order to generate AM waves , it is required that series of current pulses be

Applied to a tuned(resonant) circuit, then each pulse would induce a damped
Oscillation in the tuned circuit. The oscillation would have initial amplitude
Prop. To size of current pulse and a decay rate dependant on time constant of
tuned circuit.Since continous pulses are applied, each will form a unique sine
Prop. To amplitude of each of those pulses.

Transistor modulator
Base modulated class c amplifier
Collector modulated class c amplifier

Base modulated class c amplifier

VCC

carrier

R1

L1

L2

Q1

C1

Af signal

C4

R2

C2

R3

C3

 The BASE-INJECTION MODULATOR is similar to the control-

grid modulator in electron-tube circuits. It is used to

produce low-level modulation in equipment operating at
very low power levels.
In figure 1-49, the bias on Q1 is established by the voltage

divider R1 and R2. With the rf carrier input at T1, and no

modulating signal, the circuit acts as a standard rf amplifier.
When a modulating signal is injected through C1, it
develops a voltage across R1 that adds to or subtracts from
the bias on Q1. This change in bias changes the gain of Q1,
causing more or less energy to be supplied to the collector
tank circuit. The tank circuit develops the modulation
envelope as the rf frequency and af modulating frequency
are mixed in the collector circuit. Again, this action is
identical to that in the plate modulator.
Figure 1-49. - Base-injection modulator.
Because of the extremely low-level signals required to

produce modulation, the base-injection modulator is well

suited for use in small, portable equipment, such as

Collector modulation -Adv. Over base

modulation
Better linearity
Higher collector efficiency
Collector saturation prevents

100%Modulation from being achieved, with

only Collector being modulated
High o/p power.
But requires more modulating (input)
power.

In figure 1-47, the rf carrier is applied to the base of modulator Q1. The modulating signal is
applied
to the collector in series with the collector supply voltage through T3. The output is then taken
from the
secondary of T2. With no modulating signal, Q1 acts as an rf amplifier for the carrier
frequency. When
the modulation signal is applied, it adds to or subtracts from the collector supply voltage. This
causes the
rf current pulses of the collector to vary in amplitude with the collector supply voltage. These
collector
current pulses cause oscillations in the tank circuit (C4 and the primary of T2). The tank circuit
is tuned to

Categories of AM
demodulation
Non coherent
Coherent

Non coherent detection

Diode detector or

envelop detector

Antenna

Coherent detection
Coherent demodulation

requires the knowledge of

transmitted carrier freq &
phase at the receiver
If input to product detector
is AM signal of form
R(t) cos(2fct+r)

Then o/p of multiplier is

V1(t)=R(t) cos(2fct+r)A0cos(2fct+
r received signal phase
0 oscillating phase

V1(t)=1/2[A0R(t)cos(r-0)]+1/2[A0R(t)cos[4f ct+ r+ 0]]

LPF following the product detector removes the double
carrier freq term then the o/p is
Vout t)=1/2[A0R(t) Cos[ r
Kgain constant

0]= K R(t)

Side Band Technique

AM wave contains 3 components
DSBFC [A3E]
PT=Pc[1+m2/2]
If carrier is suppressed 2/3rd of power saved
If one side band is suppressed 50% power is

saved over suppressed carrier

Advantage of SSB
SSB is used to save power in mobiles
Low bandwidth is required to transmit SSB

Disadvantages of SSB over A3E

Difficulty in modulation and detection
Expensive

Problem
1.

Find the output power saving when

carrier and one side band is suppressed
in AM wave to depth of a)100% b)50%

Methods of obtaining SSB

Filter method
Phase shift method(or phase cancellation)

Some Pre-requisites for

Balanced Modulator Proof

The relation ship between voltage and current in a

linear resistance is given by i=bv, where b is some

constant of proportionality(transconductance, if this is a
resistor), i can be the collector current if the above
equation applies to collector current and base voltage of
a transistor. v will be the base voltage. For class A, there
will be a dc component of collector current (a), does not
depend on the base voltage.
i=a+bv
For non-linear resistance, if curve of current vs voltage
is plotted it is seen that the device reaches saturation
or some current multiplication takes place, current now
becomes proportional to the square, cube and higher
powers of voltage: i=a+bv+cv2+dv3+..

Balanced modulator
(V1+V2)
Carrier
V1

T1

id1

C2

Af in
V2

C3

TX3

ip

TX2

C4

T2

V1-V2

id2

C1
1n

V0

Principle
The modulating voltage v2 is fed to push pull and carrier voltage

v1 to a pair of FETs which are in parallel.

The carrier voltage is applied to the gates in phase; the

modulating voltage appears 1800 out of phase at the gates as

they are at opposite ends of the centre-tapped transformer.
The modulated output currents of the FETs are combined in the
centre-tapped primary of the push-pull output transformer, they
subtract as indicated.
If the system is symmetrical, carrier will be cancelled out,
however this is not the case, its heavily suppressed by 45dB or
so.
The output of the balanced modulator contains the 2 sidebands
and some extra components which are eradicated by
transformers secondary winding. The final o/p is only SBs.

Proof
The i/p voltage will be v +v

at gate of T1 and v1 v2 at gate of

T2.
If perfect symmetry is assumed(it should be understood that
the 2 devices used in balanced modulator must be matched,
whether transistors or diodes); the prop. Constants will
therefore be the same for both FETs and may be called a,b,c as
prev. mentioned.
The 2 drain currents calculated will be,
id1=a + b(v1+v2) + c(v1+v2)2 =a + bv1 +bv2 + cv12 +cv22 +2cv1v2
id2=a + b(v1-v2)+c(v1-v2)2 =a+bv1-bv2 +cv12 + cv22 -2cv1v2
as indicated the primary current is given by difference

between indiv. Drain currents thus

ip =id1 id2 =2bv2 + 4cv1v2
we may now represent the carrier voltage v 1 by Vcsinct and

modulating voltage v2 by Vmsinm t.

Substiting in ip we get,

ip=2bVm sin c t+4cVc Vm sinct . sinmt

=2bVmsinmt +4cVmVc 1/2[cos(c - m)t]-cos[(c + m)t]

 The output voltage Vo is proportional to the primary current.

Let the constant of proportionality be then,

Vo = ip =2 bVmsinmt +2cVm Vc [cos(c -m )t-cos(c +m )t]
simplifying, let P= 2bVm and Q=2cVm Vc then,
Vo =Psinmt +Qcos(c - m)t-Qcos(c + m)t

i.e. modulation frequency+lower SB+UpperSB

Suppression of unwanted
sideband

Filter method

Crystal
oscillator

Buffer

Balanced
modulator

Audio processing
& amplifier
Audio i/p

Sideband
Suppression
filter

Filter for
Other
sidebands

Balance
mixer

synthesizer

Linear
Amplifier
(class B or A)

 Buffer(data buffer-telecomm.,courtesy wikipedia):interconnecting

two digital circuits operating at different rates. compensates for a

difference in rate of flow of data, or time of occurrence of events,
when transferring data from one device to another.
A linear amplifier is an electronic circuit whose output is proportional
to its input, but capable of delivering more power into a load.(class
A,50 %efficiency,class B65%efficiency).
A frequency synthesizer is an electronic system for generating any
of a range of frequencies from a single fixed timebase or oscillator.
a mixer or frequency mixer is a nonlinear electrical circuit that
creates new frequencies from two signals applied to it. In its most
common application, two signals at frequencies f1 and f2 are applied
to a mixer, and it produces new signals at the sum f1 + f2 and
difference f1 - f2 of the original frequencies. Other frequency
components may also be produced in a practical frequency
mixer.Mixers are widely used to shift signals from one frequency
range to another, a process known as heterodyning, for convenience
in transmission or further signal processing.(e.g is a multiplier also
called product detector)

Phase shift method

Balanced
Modulator
M1
Af inAudio
amplifier

Carrier90
phase
shifter
Carrier
source
Af 90 phase
shifter

Balanced
Modulator
M2

Adder

SSB

Advantage of phase shift

method

In case of filter method 2 filters are used

which are very expensive

In filter method we can send only USB

M1 will receive sin(ct+90) and sinmt

M2 will receive sinct and sin(mt+90)
Output of M1 contains sum and

difference frequencies

V1=cos[(ct+90)- mt]-cos [(ct+90)+mt]

V2=cos[ct (mt+90)]-cos [ct+ (mt+90)]
The o/p of adder is
Vo=V1+V2
=2cos [(ct+ mt)+90]

Disadvantages of phase shift

2 balanced modulator
Inability to generate SSB at any freq

Frequency modulation(refer
Kennedy-Davis)
What is FM???
Resting freq
Frequency deviation
Carrier swing
Guard band

Mathematical representation of
FM wave(see Davis)
Instantaneous modulated freq fi(t) is given as

fi(t)=fc+kVm(t)
k freq deviation Hz/Volt
Vm(t)=Vmcosmt
fi(t)=fc+kVmcosmt
fi(t)=fc+fcosmt
Sinusoidally modulated carrier becomes
e(t)=Ec sin(t)
i(t)=2 fi(t)
The modulation can be graphically represented
by means of rotating phasor

Ecmax
e(t)

(t)

By def of i.e rate of change of angle

i(t)=d(t)/dt
t
(t)= i(t)
dt
t
= 0 2(fc+kVm(t))
dt
0
t
= 2fct+2kVm(t)dt
The freq modulated
wave is
0
e(t)=Ec sin(2fct+2kVm(t)dt)
t
= Ec sin[2fct+20f tcosm(t)dt]
=Ec sin[2fct+ f/fm sinmt]
0
= Ec sin[2fct+ mf sinmt]
mf modulation index

Freq spectrum of FM wave

From the expression of FM wave it is not

possible to tell what freq componets are

present
Solution is to use Bessel funtn
Using Bessel function eq1 can be expressed as
V=Vc{J0(mf)sinct
+ J1(mf)[sin(c+m)t-sin(c- m)t]
+ J2(mf)[sin(c+2m)t-sin(c- 2m)t]
+ J3(mf)[sin(c+3m)t-sin(c- 3m)t]
+..]
So FM wave contains carrier and infinite number
of sidebands

Graphical representation of
bessel function
FM has infinite no of

J0(mf)

Jn(mf)
side band separated by fm,
1st 2nd
0.5
2fm,
J coeficient decreases to 0 0as
1 2 3
mf
mf increases
In AM as the modulation depth increases, increases
the side band power and hence total power, In FM
total power transmitted always remain constant
Carrier component of FM wave disappear at 2.14

Bandwidth of FM wave
BW=2(f+fm)

Power in FM
Peak voltage of spectral component

Enmax=Jn(mf)Ecmax
En=Jn(mf)Ec
2
Pn=En /R
PT=P0+2(P1+P2+P3+)
In terms of rms voltages

2 /R+ E2
2 /R+E3
2 /R+]
PT=E02/R+2 [E1
2
2
2 /R[J1(mf)+J2(mf)+..]
2
2
=J0(mf)Ec/R+2Ec
2
2 (mf)+J2
2 (mf)+..]
=Ec2 /R[J0(mf)+2(J1
=Pc[J02(mf)+2(J1
2 (mf)+J2
2 (mf)+..]
=Pc

Phase modulation
Freq and phase are coming under angle modulation
In phase modulation phase of the carrier is varied
If carrier is

ec(t)=Vc sin(ct+ c)

(t)= c+ kVm(t)
K phase deviation constant
Vm(t)=Vm sinmt
kVm(t)=kVm sinmt
= sinmt
Peak phase deviation
(t)= c+ sinmt
c has no effect on modulation
Phase modulated wave can be written as
e(t)=Vc sin(ct+ m sinmt)

Advantages of FM
Amplitude of FM is const ,FM is independent of

depth
All the transmitted power in FM is useful
FM receivers can be fitted with amplitude
delimiters
It is possible to reduce the noise by incresing freq
deviation
Guard band is provided

Disadvantages of FM
A wider channel is req for FM (10 times)
FM transmitting and receiving equipment are

complex and costly

Area of reception is smaller

Generation of FM
The primary requirement of FM generator is

variable o/p freq with the variation

proportional to instantaneous ampt of
modulating signal
2 methods of FM generation
1)Direct method
2)Indirect method

Direct method
If an inductance or capacitance in a tank ckt

is varied than freq at o/p will vary

There are several devices with which it is
possible to change the capacitance as a result
of voltage change
Devices whose reactance can be varied as
based on voltage are FET and varactor diode

Basic reactance
modulator(see davis txtbook)
ib

Impedance z can be shown

purely reactive
Z=v/I
For impedance to be purely
reactive following condn
must be satisfied
ib<<iD
Zgd>>Zgs i.e Xc>>R
We know Vg=ibR
ib=V/Z=V/R-jXc
Vg=VR/R-jXc

C1

i
J1

Vg

iD
z
v

R1

 FET drain current is

id=gmVg
=gmVR/R-jXc
Z=V/id=R-jXc /gmR
=1/gm(1-jXc/R)
If Xc>>R above eq becoms
Z=-jXc/gmR
Z=jXc/gmR
=1/2fgmRC
=1/gmRC
Ceq=gmRC
Z=1/ Ceq
If the cond Xc>>R is not satisfied then will get
extra 1/gm term

Varactor diode
Varactor diode is

semiconductor diode
whose junction
capacitance varies linearly
with voltage
When reversed biased
depln region gets widen
and act as dielectric
constant
C= A/d

C2
To oscillator
tank ckt
Rfc
C3
D1

TX1

Af in

Disadvantages
These are not stable -since LC
FM requires high stability

Indirect method (Armstrong)

Oscillator

Buffer

90 deg phase
shifter

Combining
network

1st group of
multiplier

Balanced
modulator
Equalized audio

Af in

Audio equlaizer

Mixer

2nd group of
multiplierClass c pwe
amplifier

It is possible to generate FM through PM

FM detection technique
Various techniques of demodulation
Slope detector
Zero crossing detector
PLL(Phase Lock Loop)
Quadrature detection

Slope detection
In this FM demodulation is

performed by taking the time

derivative of FM signal followed by
envelop detection
FM signal is passed through limiter
The o/p of limiter is
V1(t)=V1 sin[2fct+1(t)]
The above eq is passed
through filter(slope filter)
The o/p of diffrentiator becomes
V2(t)=V1[2fct+d1/dt] cos(2fct
+1(t)]
o/p of envelop detector becomes
Vout (t)=V1[2fc+d1/dt]
=V1 2fc+V1 2kf Vm(t)

Zero crossing detector

FM signal is first passed

through limiter which

converts input signal to FM
pulse train
Pulse train V1(t) is then
passed through differentiator
whose o/p is used to trigger
monoshot
LPF is used to perform
averaging by extracting
slowly varying DC component
of signal
o/p of LPF is demodulated
wave

PLL
Vc(t)

For FM wave instantaneous freq is

Fi(t)=fc+kvm(t)
For vco instantaneous freq is
fvco= f0+kvcoVc(t)
Kvco freq deviation vco when loop is locked
fi=fvco
fc+kvm(t)= f0+kvcoVc(t)
Vc(t)=fc+kvm(t)-f0/kvco
Vc(t)=kvm(t)/kvco
Vc(t) vm(t)

Pulse communication

Analog and digital communication

Sampling Theorem: According to this theorem, signal
having minimum distortion can be reconstructed if
the rate of sampling in any of the pulse modulating
system will exceeds twice the maximum value of the
signal frequency, e.g. On the channels of the
standard telephones the range of audio frequency is
300 to 3400htz. In general, 8000 samples per
second is the worldwide standard for this system.
Types of Analog pulse communication
-Pulse amplitude modulation (PAM)
-Pulse time modulation (PTM)
-PWM, PPM

PAM
Signals to be Mixed

pulse train

modulating signal

Pulse Amplitude Modulated Signal

Signal

S/H

PAM

Multiplier

pulse AM signal

Pulse Amplitude Modulation

pulse AM signal

modulating signal

PAM is pulse modulation system in which

the signal is sampled at regular intervals
and
each sample is made proportional to
amplitude
Of signal at instant of sampling
PAM-FM

Demodulation PAM

PAM-FM

PLL

Diode
detector

LPF

Pulse time Modulation(PTM)

In this sample amplitude is kept

constant but one of the timing

characteristics of sample is varied and is
proportional to instantaneous amplitude
Adavantage of PTM over PAM???
Types of PTM
PWM
PPM

PWM

It is also called PDM, PLM

PPM
0

The amplitude and

width of pulse is kept
constant while the
position of each pulse
in relation to position of
a reference pulse is
varied by each
instantateneous
sampled value of
modulating wave
The disadvantage of
this is that if the
synchronization
between transmitter
and receiver is lost
than PPM fails but this

Digital communication
Advantages of digital communication

over Analog
-Greater noise immunity
-High security
-Error control codes which detects and
corrects the error
-Digital signal processors are used for
implementing
modulator/demodulators

Factors that influence the choice

of Digital modulation
Factors which influence digital

modulation
-Low BER at low received SNR
-Performs well in multipath and fading condition
-Occupies minimum BW
-Implimentation is Easy and cost effective

Modulation Performance measurement:

Power efficiency
BW efficiency
.

Power efficiency
It describes the ability to preserve the

fidelity of digital message at low power

levels
Power efficiency p is the measurement
of the favorable tradeoff fidelity and
signal power is made
p =Eb/N0
Ebenergy per bit
N0 noise power spectral density

BW efficiency
It describes the ability of modulation

scheme to accommodate data within a

limited BW
It is defined as ratio of o/p data rate per
hertz in given BW
B =R/B dimensions:bps/Hz
Shannon channel coding theorem
C=B log (1+S/N)
Then nBmax=C/B=log2 (1+S/N)

Problem1
If SNR of wireless communication link is

20dB and RF BW is 30KHz,determine the

maximum theoretical data rate that can be
transmitted
Soln
C=199.75 bps

The selection of the modulation scheme

is done according to power and
bandwidth efficiency and the channel
capacity

Digital modulation
PCM

PCM uses sampling technique but it is a

digital process i.e instead of sending the
pulse train by continuously varying one
of the parameters, PCM produces series
of bits corresponding to amplitude levels
of signal

Principle of generation of
PCM
It involves 4 steps

-Sampling
-Quantization
-Coding
-Synchronizing

4
3
2
1
0

L3
L2
L1
L0

011
010

001
000
L-0 100
L-1 101
L-2 110
L-3 111
t

Disadvantages
Choosing a discrete value near the analog signal for

each sample leads to quantization error, which

swings between -q/2 and q/2. In the ideal case (with
a fully linear ADC) it is uniformly distributed over this
interval, with zero mean and variance of q2/12.
Between samples no measurement of the signal is
made; the sampling theorem guarantees nonambiguous representation and recovery of the signal
only if it has no energy at frequency fs/2 or higher
(one half the sampling frequency, known as the
Nyquist frequency); higher frequencies will generally
not be correctly represented or recovered.

Types of PCM
DPCM
Delta PCM(Delta Modulation)
ADPCM

[http://www.andreasschwope.de/ASIC_s/Schnittstellen/Data_Lin
es/body_modulation.html]

DPCM
If the input is a continuous-time analog signal, it needs to be sampled first

so that a discrete-time signal is the input to the DPCM encoder.

Option 1: take the values of two consecutive samples; if they are analog
samples, quantize them; calculate the difference between the first one and
the next; the output is the difference, and it can be further entropy coded.
Option 2: instead of taking a difference relative to a previous input sample,
take the difference relative to the output of a local model of the decoder
process; in this option, the difference can be quantized, which allows a
good way to incorporate a controlled loss in the encoding.
Applying one of these two processes, short-term redundancy (positive
correlation of nearby values) of the signal is eliminated; compression
ratios on the order of 2 to 4 can be achieved if differences are
subsequently entropy coded, because the entropy of the difference signal
is much smaller than that of the original discrete signal treated as
independent samples.
DPCM was invented by C. Chapin Cutler at Bell Labs in 1950; his patent includes both

methods.[1]

Delta
Modulation
Delta modulation (DM or -modulation)is an analog-to

digital and digital-to-analog signal conversion technique

used for transmission of voice information where quality is
not of primary importance. DM is the simplest form of
differential pulse-code modulation (DPCM)
the transmitted data is reduced to a 1-bit data stream. Its
main features are:
the analog signal is approximated with a series of segments
each segment of the approximated signal is compared to
the original analog wave to determine the increase or
decrease in relative amplitude
only the change of information is sent, that is, only an
increase or decrease of the signal amplitude from the
previous sample is sent whereas a no-change condition
causes the modulated signal to remain at the same 0 or 1
state of the previous sample.

Principle of the delta PWM. The output signal (blue) is

compared with the limits (green). These limits correspond
to the reference signal (red), offset by a given value. Every
time the output signal reaches one of the limits, the PWM
signal changes state.

ADPCM
Adaptive DPCM (ADPCM) is a variant of DPCM

(differential pulse-code modulation) that varies the

size of the quantization step, to allow further
reduction of the required bandwidth for a given signalto-noise ratio.
Typically, the adaptation to signal statistics in ADPCM

consists simply of an adaptive scale factor before

quantizing the difference in the DPCM encoder.[1]
ADPCM was developed in the early 1970s at Bell Labs

for voice coding, by P. Cummiskey, N. S. Jayant, and

James L. Flanagan.[2]

Digital Modulation technique

{before this please refer to pp. 234

(bottom) of Rappaport: geometry of

modulation signals}
Non linear or constant envelop
Linear

Linear
In this amplitude of transmitted signal S(t)

varies linearly with modulating digital signal

m(t)
linear technique are BW efficient hence are
very attractive for use in wireless
communication system
e.g:BPSK,DPSK, QPSK,OQPSK,/4QPSK

Non linear or constant

envelop

Many mobile-radio communication

systems use nonlinear modulation

methods where the amplitude is constant
because:
-power efficient
-provides high immunity against signal
fluctuation due to Rayleigh fading
They occupy larger BW
In situations wherein BW efficiency is
important than power efficiency,
constant envelop is not suited
e.g BFSK,MSK,GMSK

ASK (Amplitude Shift Keying)

Digital signal
0

The digital signal is

used to switch the

carrier between
amplitude levels and is
referred as ASK or
OOK(on off keying) or
ICW(Interrupted
continuos wave)

carrier

ASK

ASK Generation
1 0

1 1

ASK-AM
1 0

1 1

AM

Product multiplier

ASK-AM

ASK demodulation
Non coherent detection
Coherent detection

Coherent detection
(synchronous)
ASK or OOK

LPF

Carrier
Recovery
ckt

Vout
Decision
ckt

Vout

fc

It simply retranslates the frequencies of the incoming waveform down to the base
band(info signal). This is done by multiplying or heterodyning the incoming ASK
waveform with a local oscillator matched to the carrier. The output of the
multiplier is,
Fb (t) {[cos(ct)]2} = +
The low pass filter will remove the cos (2ct) component. The LPF is generally not
only an LPF but an envelope detector. The decision circuit is an op-amp
comparator making a decision for logic 1 or 0.

Decision circuit
In the binary case, the decision circuit

compares the received signal with a

threshold at specific time instants

Distorted symbols

Decision
threshold

Clean symbols

Decision
Decision circuit
circuit

Decision time instant

FSK
In this carrier freq is
shifted in steps
corresponding to the
levels of digital
modulating signal
Here 2 carrier freq are
used one
corresponding to
binary 1 and other to
0
the carriers are
s1(t)=Ac cos(2f1t+0)

1 0 1

Osc 1
Inverter

osc2

Smallest freq separation

mf or
f2

f1

In general FSK signal may be represented as

Eb=1/2 Ac Tb
Eb energy per bit
Discontinuous FSK
Discontinuous FSK is represented as

Since Phase discontinuities leads to several problems such as

Spectral spreading and spurous transmissions, this type of FSK
generation is not used in regulated wireless system

The most common method for generating FSK

signal is to Freq modulate a carrier using

message signal (binary)
FSK may be represented as
SFSK(t)=2Eb/Tb cos[2fc+(t)]
=2Eb/Tb cos[2fct+2kVm(t)dt]
t

BW of FSK
Transmission BW is given by Carson rule

BT=2f+2B
=2(f+B)

FSK demodulation
Coherent Detection
Non coherent FSK

FSK Coherent Detection

It consits of 2 correlator
supplied With locally
generated coherent Ref
signal
Difference of correlator o/p
is
compared with threshold
Comparator
If difference >threshold
than
bit is binary 1

Carrier
(cos2fct)
Recovery
f1

Carrier
(sin2fct)
Recovery
f2

Noncoherent FSK
Receiver consits of pair of matched filter
Followed by envelop detector
Filter in upper path is matched to fL
Filter in lower path is matched to fH
The o/p of envelop detector is sampled
at t=kTb
their difference is compared with
a threshold, accordingly the comparator
Decides whether the received
Bit was a 1 or 0 n the signal.

t=kTb

MSK (Minimum shift keying)

MSk is a special type of CPFSK wherein the peak freq

deviation is th bit rate

In other words with modulation index 0.5(k fsk =(2f/Rb)
f peak RF freq deviation
Rbbit rate
Bits are separated in odd and even bits
Modulation index of 0.5 corresponds to minimum freq
spacing that allows 2 FSK signal to be coherently
orthogonal
2 FSK signals VH(t) and VL(t) are said to orthogonal if

MSK is mostly used in mobile communication due to

Good spectral efficiency, good BER ,constant envelope and

self synchronizing but not BW efficient

Example of MSK

VL
VH

PSK
PSK MODULATOR
BPSK
101

Ac max cos(ct+c)

Phase-shift keying (PSK) is a digital modulation scheme that conveys data

by changing, or modulating, the phase of a reference signal (the carrier wave).
Two common e.g are BPSK which uses two phases, and QPSK which uses four phases

BPSK
BPSK is the simplest form of PSK. It uses

two phases which are separated by 180

and so can also be termed 2-PSK.
In BPSK ,the phase of constant ampt
carrier signal is switched betwn 2 values
according to 0 and 1
2 phases are separated by 180 deg
If sinusoid carrier has ampt A c and
energy per bit Eb=1/2Ac2 Tb then BPSK
signal has form
SBPSK(t)=2Eb/Tb cos[2fct+c]
0<=t<=Tb
(1)

or
=2Eb/Tb cos[2fct++c]
0<=t<=Tb(0)
i.e.-2Eb/Tb cos[2fct+c]
0<=t<=Tb
(0)
1(t)= 2/Tb cos(2fct+ c)
Using this SBPSK=Eb 1(t),- Eb 1(t)}={1,0}
If m(t) is binary data then transmitted signal
represented as

SBPSK(t)=m(t) 2Eb/Tb cos[2fct+c]

BPSK receiver(

see rappaport,diagram is self-explanatory

Received SBPSK(t)=m(t)

2Eb/Tb cos[2fct+c+ ch]

Band pass filter is tune to 2fc
Freq divider is used to recreate the
cos[2fct+]

The o/p of multiplier is given as

m(t) 2Eb/Tb cos [2fct+]
=m(t) 2Eb/Tb [1/2+1/2 cos2
[2fct+]
This signal is passed through
integrator and dump ckt (LPF)
Bit synchronizer is used to sample the
integrator o/p at the end of bit
period
At the end of each bit period switch is
closed and o/p is fed to decision ckt

DPSK
In this input binary data

is differentially encoded
and then modulated
using BPSK
Differentially encoded
data {dk} is generated
frm binary seq{mk} by
complimenting the
modulo 2 sum of mk &
dk-1
Leave the dk
unchanged frm previous
symbol if mk=1
Dk toggles if mk=0

d k= mk

dk-1

DPSK modulation
It consist of 1 bit

delay element and

logic ckt
interconnected to
generate
differentially
encoded data

DPSK Receiver
DPSK

Integrator

Advantages of DPSK
Simple receiver ckt
Good power efficiency
Thresholds dk
detector

+
Logic mk=dk+dk-1
circuit
dk-1
Delay Tb

Cos(2fct)

Quadrature Phase Shift Keying

(QPSK)
Sometimes known as

quadriphase PSK or 4-PSK,

QPSK uses four points on the
constellation diagram,
equispaced around a circle.
With four phases, QPSK can
encode two bits per symbol
QPSK has twice the bandwidth
efficiency
QPSK can be interpreted as two
independent BPSK systems (one
on the I-channel and one on Q)

Constellation diagram for QPSK with

Gray coding. Each adjacent symbol
only differs by one bit.

QPSK
In QPSK phase of carrier takes on one of 4 equally

spaced values such as 0, /2, , 3 /2

Each phase correspond to pair of message bits
SQPSK= =2Es/Ts cos[2fct+(i-1)/2] 0<=t<=Ts
i=1,2,3,4
Ts symbol period and is equal to twice the bit period
Using trigonometric identity the above eq is rewritten as
SQPSK= 2Es/Ts cos[(i-1)/2] cos(2fct)
-2Es/Ts sin[(i-1)/2] sin(2fct)
If 1= 2/Ts cos(2fct) & 2= 2/Ts Sin(2fct) then
SQPSK= Es cos[(i-1)/2] 1 -Es sin[(i-1)/2] 2 for
i=1,2,3,4
Based on this representation a QPSK signal can be
depicted using 2-D constellation diagram with 4 points

QPSK constellation diagrams

QPSK Modulation
A unipolar Binary message

has bit rate Rb

1001
The bit stream m(t) is then
splits into 2 bit streams
mI(t) (in phase) and mQ(t)
(quadrature stream)
having bit rate Rs=Rb/2
11000110
2 binary seq are separatly
modulated by 1(t) &
2(t)
1010
2 BPSK are summed to
produce QPSK

cos2fc

sin2fc

QPSK receiver
BPF removes the out of band

noise and adjacent freq

Filter o/p is split in 2 parts
Each part is coherently
demodulated using in phase
and quadrature carriers
The o/p of integrator is
passed through decision ckt
to generate in phase and
quadrature binary stream

Combined linear and constant

envelop modulation technique
In modern communication digital data can be sent

by varying both the envelop and phase (or freq)

M-ary QAM

M-ary modulation

modulation technique in which base band data

is mapped into 4 or more RF carrier signals
In M-ary signaling scheme , 2 or more bits are
grouped to form symbol
n
M=2

Depending on whether amplitude , phase

or frq of carrier is varied we have

M- ary ASK
M- ary PSK
M- ary FSK

M- ary PSK
In MPSK ,the carrier phase takes on

one of M possible values

i=2(i-1) /M
i=1,2,3.M
The modulated waveform is
Si(t)=2Es/Ts cos(2fct + (i-1) 2/M]

The above eq in quadrature form

Si(t)=2Es/Ts cos[(i-1) 2/M] cos(2fct)
-2Es/Ts sin [(i -1) 2/M] sin(2fct)
By choosing orthogonal basics
Si(t)=Es cos[(i-1)/2] 1(t)
-Es sin [(i-1) /2] 2(t)

Bandwidth vs. Power

Efficiency

M- ary QAM(QAM)
In M-PSK ,amplitude of transmitted signal is kept

const hence circular constellation

By allowing ampt to vary along with phase this
modulation is called QAM
QAM is a modulation scheme which conveys data by
changing (modulating) the amplitude of two carrierConstellation diagram for rectangular 16-QAM
waves. These two waves, usually out of phase with
each other by 90 and are thus called quadrature
carriershence the name of the scheme is QAM.
Extensive use in digital microwave radio links
Constenstilation consits of square lattice of signal
points
The general form of QAM is
Si(t)=2Emin/Ts ai cos(2fct)+
2Emin/Ts bi sin(2fct)
i=1,2,M
Emin energy of signal with lowest amplitude

1(t)=2/Ts cos(2fct)
2(t)=2/Ts sin(2fct)

0<=t <=Ts
0<=t <=Ts
The coordinates of ith message points
are
aiEmin and biEmin, where {ai,bi} is an
element of LxL matrix
{ai,bi} =[(-L+1,L-1)
.
.

(-L+3,L-1) (L-1,L-1)
(-L+1,L-3) (-L+3,L-3)(L-1,L-3)
.
.
.
.
.
.
(-L+1,-L+1) (-L+3,-L+1).. (L-1,-L+1)]

Where L=M

For 16 QAM matrix

becomes
{ai,bi}=[(-3,3)

(-1,3)

(1,3)

(3,3)
(-3,1)
(-3,-1)

(-1,1)
(-1,-1)

(1,1) (3,1)
(1,-1) (3,-

1)
(-3,-3) (-1,-3)
3)]

(1,-3) (3,-