DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

LESSON PLAN

Name of the staff : Prof.S.MOHAN

Name of the subject : EC 6801 - WIRELESS COMMUNICATION
Academic year : 2019-2020
Semester : VIII SEMESTER
Course Objective :

Know the characteristic of wireless channel

Learn the various cellular architectures

Understand the concepts behind various digital signaling

schemes for fading channels

Be familiar the various multipath mitigation techniques

    Understand the various multiple antenna systems

Reason for
Teaching Assigned Attained
WEEK No Unit Date Hour Portions Planned deviation, if
Aids hours hours
any

Large scale path loss – CB

1
Path loss models

Free Space and TwoRay CB

1
models
8
1 Link Budget design PP

1 1 Small scale fading CB

Parameters of mobile CB
1
multipath channels

Time dispersion CB
1
parameters

1 Coherence bandwidth – CB
Doppler spread &
 Coherence time

Fading due to Multipath CB

1
time delay spread

Reason
Teaching Assigned Attained for
WEEK No Unit Date Hour Portions Planned
Aids hours hours deviation
, if any

flat fading – frequency CB

1
selective fading

Fading due to Doppler CB

spread – fast fading –
1 slow fading.

8
Multiple Access PP
2
techniques
2 2 FDMA CB

2 TDMA CB

2 CDMA CB

Capacity calculations– CB
2
Cellular concept

Frequency reuse - PP
2
channel assignment

Reason
Teaching Assigned Attained for
WEEK No Unit Date Hour Portions Planned
Aids hours hours deviation
, if any

2 hand off- interference CB

2 system capacity CB
8
3 2
- trunking & grade of
service
CB

Coverage and capacity PP

2
improvement.
 Structure of a wireless PP
3
communication link

Principles of Offset- CB
3
QPSK

3 p/4-DQPSK CB

3 Minimum Shift Keying CB

Reason
Teaching Assigned Attained for
WEEK No Unit Date Hour Portions Planned
Aids hours hours deviation
, if any

Gaussian Minimum Shift CB

3
Keying

Error performance in CB
3
fading channels

3 OFDM principle PP
8
4 3 Cyclic prefix CB

3 Windowing CB

3 PAPR PP

4 Equalisation CB

4 Adaptive equalization CB

Reason
Teaching Assigned Attained for
WEEK No Unit Date Hour Portions Planned
Aids hours hours deviation
, if any

Linear and Non-Linear CB

4
equalization 8
5 Zero forcing and LMS CB
4
Algorithms
 4 Diversity CB

4 Micro and Macrodiversity PP

Diversity combining PP
4
techniques

Error probability in fading CB

4
channels

4 diversity reception CB

4 Rake receiver, CB

Reason
Teaching Assigned Attained for
WEEK No Unit Date Hour Portions Planned
Aids hours hours deviation
, if any

5 MIMO systems PP

5 spatial multiplexing CB

5 System model CB

5 Pre-coding CB
8
6 5 Beam forming CB

5 transmitter diversity, CB

5 receiver diversity PP

Channel state CB
5
information

Teaching Assigned Attained Reason

WEEK No Unit Date Hour Portions Planned for
Aids hours hours
deviation
 , if any

capacity in fading CB
5
channels
8
7 5 non-fading channels PP

Subject incharge HoD Principal

UNIT-I WIRELESS CHANNELS

INTRODUCTION :
The mobile radio channel places fundamental limitations on the performance of
wireless communication systems.

The transmission path between the transmitter and the receiver can vary from
simple line of sight to the one that is severely obstructed by buildings, mountains and
foliage.

Unlike wired channels that are stationary and predictable, radio channels are
extremely random.

Even the speed of motion rapidly fades the signal level as a mobile terminal
moves in space.

Modeling the radio channel has been one of the most difficult parts of mobile
radio system design.

Large Scale Path Loss :

The mechanisms behind electromagnetic wave propagation are diverse and can
be attributed to reflection, diffraction and scattering.

Cellular radio systems operate in urban areas, there is no direct line of sight path
between the transmitter and the receiver, whereas the presence of high-rise buildings
causes severe diffraction loss.

Due to multiple reflection, the electromagnetic waves travel along different paths
of varying lengths.

The interaction between these waves causes multipath fading at a specific

location, and the strengths of the waves decrease as the distance between the transmitter
and receiver increases.

Propagation models that predict the mean signal strength for an arbitrary
transmitter-receiver (T-R) separation distance are useful in estimating the radio
coverage area of a transmitter and are called large scale propagation models.

Large scale model characterize signal strength over large T-R separation
distances (several l00s or 1000 meters)

Propagation models that characterize the rapid fluctuations of the received signal
strength over very short travel distances (a few wavelengths) are called small-scale or
fading models.
Path Loss Models :
Path loss or path attenuation is the reduction in power density of an
electromagnetic wave as it propagate through space.

Path loss is a major component in the analysis and design of the link budget of a
telecommunication system.

Path loss is the difference between the transmitter power and the received power.

Path loss is influenced by terrain contours, environment, propagation medium,

the distance between the transmitter and the receiver, and the height and location of
antennas.

Path loss models normally includes propagation losses caused by the normally
includes propagation losses caused by the natural expansion of the radio wave front in
free space, absorption losses, diffraction and also due to indoor models.

Indoor Propagation Models :

Characterize the radio propagation inside buildings.

Indoor radio channel differs from the traditional mobile radio channel in two
aspects.

1. distance covered are much smaller

2. variability of the environment is much greater.

Propagation within buildings are influenced by layout of the buildings.

construction materials and building type.

In general, indoor channels maybe classified either as Line-of-sight (LOS) or

obstructed (OBS) with varying degrees.
1.Partition Losses (Same Floor)
Buildings have a wide variety of partitions and obstacles which form the internal
and external structure.

Partitions that are formed as part of the building structure are called hard
partitions.

Partitions that may be moved and which do not spam to the ceiling are called soft
partitions.

2.Partition Losses (between Floors)

Losses between floors of a building are determined by the external dimensions
and materials of the building, type of construction used to create the floors and the
external surroundings.

Number of windows in a building and the presence of tinting impact the loss
between the floors.

3.Log-distance path Loss Model

Obey the distance power low

PL(dB) = PL(do)+ 10n.log (d/do) +Xo

Where Xo = normal random variable having standard deviation

n = surroundings and building type.

4.Ericsson Multiple Breakpoint Model

Obtained by measurements in a multiple floor office building.

Model has four breakpoints and considers both an upper and lower bound on the
path loss.

Ericsson model provides a deterministic limit on the range of path loss at a

particular distance.
5.Attenuation Factor Model
An in building site-specific propagation model that includes the effect of
building type as well as the variations caused by obstacles by sidle.

This model provides flexibility and reduce the standard deviation between
measured and predicted path loss in two different buildings are shown.

The attenuation factor model is

PL(d) dB = PL (do) + 10.n SF.log (d/do) + FAF + ∑ PAF

Where nSF =exponent value for the same floor.

FAF = Floor attenuation Factor.

PAF = Partition attenuation Factor.

Technique used for drawing a single ray between the transmitter and receiver is
called primary ray tracing.

For multiple floor separation,

PL (d) dB = PL (do) +10. nMF log (d/do) + ∑ PAF

Where nMF path loss exponent based for multiple floors.

Outdoor propagation Models :

Radio transmission in a mobile communication systems takes place often over
irregular terrain.

The terrain profile of the particular area needs to be taken for estimating path
loss.

The terrain profile may very from a simple curved earth profile to a highly
mountainous profile.

A number of propagation models are available to predict path loss over irregular
terrain.

There models aims to predict signal strength at a particular receiving point or a

specific local area (called sector).
Commonly used outdoor propagation models:

1) Longley Rice Model

2) Durkin’s model

3) Okumura model

4) Hata model

5) PCS Extension to Hata model

6) Walfisch and bertoni model

7) Wideband PCS Microcell Model

Longley-Rice Model :

Applicable to point to point communication systems in the frequency range from

40MHZ to 100GHZ over different terrains.

The median transmission loss is predicted using the path geometry of the terrain
profile and refractivity of the troposphere.

Geometric optics techniques are used to predict signal strengths withinthe radio
horizon.

Diffraction losses over isolated obstacks are estimated using fresnel-kirchoff

knife-edge models.

Forward scatter theory is used to make troposphere predictions over long

distances and far field diffraction losses in drouble horizon paths.

The Longley-Rice propagation prediction model is also referred to as the ITS

irregular terrain model.

This model calculate the large scale median transmission loss relative to free
space loss over irregular terrain for frequencies between 20MHZ and 10GHZ.

It takes input the transmission frequency, path length, polarization, antenna

heights, surface refractivity, effective radius of earth, ground condctivity, ground
dielectric constant and dimate.
Longley-Rice method operates in two models :

When a detailed terrain path profile is available, the path specific parameters can
be easily determined and the prediction is called a point-to-point mode prediction.

If the terrain path profile is not available to estimate the path specific parameters,
the prediction is called an area mode prediction.

Durkin’s Model :
Prediction is similar to Longley-Rice by Edward and Durkin as well as Dadson.

Predicts only large scale phenomena over irregular terrain and the losses caused
by obstacles in a radio path.

Execution consist of two parts :

First part accesses a topographic database of a proposed service area and

reconstructs the ground profile information along the radial joining the transmitter to
the receiver.

Assumption is that the receiving antenna receives all of its energy along the
radial, and therefore experiences no multipath propagation.

Second part, calculates the expected path loss along that radial.

After this is done, the simulated receiver location can be iteratively mored to
different locations in the service area to deducted the signal strength contour.

Okumura Model :
Wideley used models for signal prediction in urban areas.

applicable for frequencies in the range 150MHZ-1920MHZ and distances of

1Km to 1000Km.

Used for base station antenna heights ranging from 30m to 1000m.

The model can be expressed as

L50(dB) = LF + Amu(f,d) – G(hte)-G(hre) – GAREA

Where L50 = 50th percentile value of propagation path loss.

LF = Free space propagation Loss

 Amu = Median attenuation relative to free space

G(hte) = Basestation antenna height gain factor

G(hre) = Mobile antenna height gain factor

GAREA = Gain due to the type of environment.

Free space propagation model :

Used to predict received signal strength when the transmitter and receiver have a
clear, unobstructed Line-of-sight path between them.

Satellite and Microwave line-of-sight radio links typically undergo free space
propagation.

Large-scale radio wave propagation models, the free space model predicts that
received power decays as a function of the T-R separation distance raised to some
power.

The free space power received by a receiver antenna which is separated from a
radiating transmitter antenna by a distance d is given by the frils free space equation.

Pr(d) = Pt.Gt.Gr./ 2  (1)

(4/)2 . d2 . .L

Where Pt = Transmitted Power

Pr(d) = Received Power

Gt, Gr = Transmitter antenna gain, Receiver antenna gain

d = T-R separation, meters

L = System Loss factor (L/1)

y = Wavelength, meters
The gain of an antenna related to its effective aperture

G = 4 Ae (2)

The effective aperture Ae related to the physical size of the antenna and is
related to the carrier frequency by

/= C = 2/C  (3) w=2/f

f wc

Where f = carrier frequency, Hertz

wc = carrier frequency, radians per second

c = speed of light, m/s

L=1, indicates no loss in the system.

Losses are usually due to transmission line attenuation filter losses and antenna
losses in the system.

An isotropic radiator is an ideal antenna which radiates power with unit gain
uniformly in all directions and is often used to reference antenna gains.

The effective isotropic radiated power (EIRP) is designed as

EIRP = Pt.Gt  (4)

represents the maximum radiated power available from a transmitter in the

direction of maximum antenna gain.

In practice, effective radiated power (ERP) denotes the maximum radiated power
and antenna gains are given in dBi (gain w.r.t an isotropic antenna) or dBd (gain w.r.t a
half-wave-dipole)

The path loss, which represents signal attenuation, designed as the difference
between the effective transmitted power and the received power.
 The path loss for the free space model when antenna gains are included is given
by

[
PL(dB) = 10 log Pt = 10 log Gt.Gr.1/2 ]
 (5)
Pr 41/2. d2
When antenna gains are excluded, the antennas are assumed to have unity gain
and path loss is

PL(dB) = 10 log Pt = 10 log 1/2 [ ] 2

(6)
Pr (4- d)
The far field or fraunhofer region of a transmitting antenna is defined as the
region beyond the far field distance df.

 (7)
df = 2D2
1/

Where D = largest physical linear dimension of the antenna.

To be in the far field region, df >>D  (8)

df>>1/  (9)

Large scale propagation models use a close in distance do, known received
power reference point.

The received power Pr(d) at any distance d>do, related to Pr at do.

The reference distance must be chosen such that it lies in the far field region ie;
do>df.

Thees, the received power in free space at a distance d> do,

( )
Pr(d) = Pr (do) do 2
d> do >  (10)
d
In mobile radio systems, Pr are expressed interms dBm or dBW

( )  (11)
Pr(d) dBM = 10 log [pr(do)] + 20 log do
0.001 w do

Where Pr(dO) = Watts

For practical systems, the reference distance do using low-gain antennas in the 1-
2GHZ is choosen to be 1m in the indoor and 100m or 1km in outdoor environment.

Two-Ray Model/ Ground Reflection Model :

In a mobile radio channel, a single direct path between the base station and a
mobile is seldom the only physical means for propagation.

Hence the free space propagation model is inaccurate.

The two-ray ground reflection model is a useful propagation model that is based
on geometric optics and considers both the direct path and a ground reflected
propagation path between transmitter and receiver.

This model is accurate for predicting the large-scale signal strength over
distances of sevaral km for mobile radio systems that use tall towers, as well as for line-
of-sight microcell channels in urban environment.

The maximum T-R separation distance is atmost only a fewtens of km and the
earth maybe assured to be flat.

The total received field, Etot is a result of the direct line-of-sight component,
ELos and the ground reflected component. Eg.

ht is the height of the transmitter

hr is the height of the receiver

If Eo is the free space E-field (v/m)

at a reference distance fo from the

transmitter, then for d> do, the free

space propagation E-field is given by

 [
E(d,t) = Eo.do cos wc (t-d)]  (1) (d>do)
d c

Where /E(d,t)/ = Eo.do/d represents the envelope of the E-field at d meters from
the transmitter.

Two propagation waves arrive at the receiver the direct wave that travels a
distence d’ and the reflected wave that travels a distance d”

The E-field due to the line-of sight component at the receiver

[ ]
ELos (d;t) = Eo.do. cos wc (t-d’)  (2)
d’ c
The E-field for the ground reflected wave, which has a propagation distance of
d”
Eg(d”,tt) Eo.do cos [wcwcc (tt-dd”)]  (3)
d” c
According to laws of reflection in dielectrics
i = o.  (4)
Eg = Ei  (5)
Et = (1 + ) . Ei  (6)
where [= refelection coefficient for ground
The resultant E-field, assuming perfect horizontal E-field polarization and
ground reflection ie;
[ = -1 and E= 0
( EToT) = 1ELoS + Eg1  (7)
The electric field ETOT (d,t)
ETOT (d,t) = Eo.do cos [Wc (t-d1)] + (-1) Eo-do.cos [wc(t-d’)] (8)
d’ c d ” c
Using the method of images, the path difference A, between the line-of-sight and
the ground reflected paths.
 = d”-d1
= √(ht+hr)2+d2 - √(ht+hr)2+d2 (a)
When the T-R distance d is large compared to ht+hr
 = d”-d’ = 2 ht.hr ( b)
d
Once the path difference is known, the phase difference between the two E-field
components and the time delay Td between the arrival of the two components can be
computed using
  = 2π = wc (11) = c/f = c/f
c
c = wc =2πfc
 
Td =  =  (12)
c 2πfc
d becomes large, difference between d’ and d’’ becomes very small, and the amplitudes
ELOS = ELOS = Eg Eg and anddssdfsfd differ only in phase.

E0 . D0 = E E0 . D0 = E0 . D0 (13)
d d’ d’’
If the received E-filed is evaluated at t=d’’/c,
= E0 . D0 <  - E0 . d0
d’ d’’
= E0 . D0 [<  - 1]
d

d’

ht ht – hr
hr

ht d’’ hr
ht + hr

Method of images

Where d= distance over a flat earth between the bases of the transmitter and
receiver antennas.
E0.d0 2 E0.d0 2
ETOT(d) = (coscos  - 1)2 + .sm2  (15)
d d

E0.d0 2−2cos
ETOT(d) =  (16)
d

2.𝐸𝑜 .𝑑𝑜 Ѳ
ETOT(d) = .sin = (17)
d 2

This gives the exact received E-field for the two-ray ground reflection model.

-(15) may be simplified whenever sm (Ѳ/2) =Ѳ/2

Ѳ = 2 ht.hr < 0.3 rad (18)

2 d
 d > 20 ht.hr = 20 ht.hr (19)
3

ETOT (d) = 2.Eo.do . 2 ht.hr = k v/m (20)

d d d2

The received power at a distance d from the transmitter for the two-ray ground
bounce model.
Pr = Pt.Gt.Gr.ht2.hr2 (21)
d4

The path loss for the two-ray model,

PL(dB) = 40 log d- (10 log Gt + 10 log Gr + 20 log ht + 20 log hr) (22)

At small T-R separation distances, (8) is used to compute the total E-field..
Link Budget Design :
Radio propagation models are derived using a combination of analytical and
empirical methods.

Empirical approach is based on fitting curves or analytical expressions that

recreate a set of measured data.

Over time, classical propagation models are used to predict large-scale coverage
for mobile communication system design.

By using path loss models to estimate the received signal as a function of

distance becomes possible to predict the SNR.

(1) Log-distance path Loss Model :

Both theoretical and measurement based propagation models indicate that
average received signal power decreases. logarithmically with distance, in indoor or
outdoor channels.

The average large-scale path loss for an arbitrary T-R separation as a function of
distance by using path loss exponent.

PL (d) + d n
do
(or)
PL (dB) = PL(do) + 10nlog d
do
Where n = path loss exponent, indicates the rate at which the path loss increases
with distance.

do = close –in reference distance from transmitter

d = T-R Separation distance.

denotes the enremble average of all possible path loss free space reference
distance must be selected in appropriate manner for the propagation environments.

In large coverage cellular systems, 1km reference distances are commonly used.

Whereas in microcellular systems, small distances (100 mor 1m) are used.

2) Log-normal shadowing :
Measures signals that vastly differ from the average predicted value.
 Path loss PL(d) at a particular location is random and distributed log-normally.
dfgdfgdfgdfgdfgdfgdfgdfgfdgfgfg
𝑑
PL(d) [db] = PL (d) + X 𝜍 = PL (do) + 10n.log + X𝜍
𝑑𝑜

Pr (d) [dBm] = Pt [dBm] – PL(d) [dB]

Where X𝜍 = Zero mean Gaussian distributed describes the random shadowing effects
which occur over a large number of measurement locations having the same T-R
separation, but have different levels of cutter on the propagation path, is known as log
normal shadowing.

3) Determination of percentage of coverage Area

Due to random effects of shadowing, same locations within a coverage area will
be below a particular desired received signal thershold.

Computer now the boundary coverage relates to the percent of area covered
within the boundary.

For a circular coverage area having radius R from the base station, desired
received signal thershold.

Percentage of use service area

1
u(8) = Pr Pr 𝑟 > 8 . 𝑑𝐴
𝜋𝑅 2

1 2𝜋 𝑅
= 0 0
Pr [Pr(r)>8].r.dr.do
𝜋𝑅 2

Small scale fading:

Small scale fading or fading is used to describe the rapid fluctuations of the
amplitudes, phases of time or travel distance.

Fading is caused by interference between two or more versions of the transmitted

signal which arrive at the receiver at slightly different times.

these waves called multipath waves combine at the receiver antenna to give
resultant signal.

Multipath in the radio channel creates small-scale fading effects.

There important effects are

 Rapid changes in signal strength over a small travel distance

  Random frequency modulation due to varying Doppler shifts.
 Time dispersion caused by multipath propagation delays.

Multipath propagation:

the presence of reflecting objects and scatters in the channel creates a constantly
changing environment that dissipates the signal energy in amplitude, phase and time.

These effects result in multiple versions of the transmitted signal that phase and
amplitudes of the different multipath components cause fluenations in signals strength.

Multi path propagation often lengthens the time required for the baseband
portion of the signal to reach the receiver which cause smearing due to ISI.

Speed of the mobile:

Relative motion between the base station and the mobile results in random
frequency modulation due to different Doppler shifts on each of the multipath
components.

Doppler shift will be +ve or –ve depending on whether the mobile receiver is
moving toward or away from the base station.

Speed of surrounding objects:

If objects in the radio channel are in motion, they in duce a time varying Doppler
shifts on multipath components.

If the surrounding objects move at a greater rate than the mobile, then this effect
objects move at a greater rate than the mobile, then this effect dominates the small-scale
fading.

Coherence time defines the staticness of the channel, and is directly impacted by
the Doppler shift.

Transmission bandwidth of the signal

If the transmitted radio signal bandwidth > the bandwidth of the multipath
channel, the received signal will be distorted but will not fade much.

Bandwidth of the channel is quantified by the coherence bandwidth, is related

to the specific multipath structure of the channel.

Coherence bandwidth is a measure of the maximum frequency difference for

which signals are strongly correlate in amplitude.
 If transmitted signal has a narrow bandwidth the amplitude of the signal will
change rapidly.

Parameters of mobile multipath channels:

Parameters are derived from the power delay profile.

Power delay profiles are measured using the following techniques.

Power delay profile measurements over a local area in order to determine an

average small-scale power delay profile.

Time dispersion parameters:

The mean excess delay, rms delay spread and excess delay spread are
multipath channel parameters that can be determined from a power delay profile.

The time dispersive properties of wide band multipath channels are quantified by
their mean excess delay (𝜏) and rms delay spread.

The mean excess delay is the first moment of the power delay profile and is
defined to be

𝜏= 𝑘 𝑎𝑘2. 𝜏𝑘 = 𝑘 𝑝 𝑘𝜏 . 𝑘𝜏

𝑘 𝑎𝑘2 𝑘 𝑝(𝜏𝑘)

The rms delay spread is

𝜍𝜏 = 𝜏2 − 𝜏 2

Where

𝜏2 = 𝑘 𝑎𝑘2. 𝜏𝑘2 = 𝑘 𝑝 𝑘𝜏 . 𝑘𝜏2

𝑘 𝑎𝑘2 𝑘 𝑝(𝜏𝑘)

These delays are measured to the first detectable signal arriving at the receiver
𝜏0 = 0

The maximum excess delay is defined to be time delay which multipath energy
falls to XdB below the maximum.

In other words, the maximum excess delay is defined as 𝜏𝑥 − 𝜏 0, where

𝜏0 =first arriving signal and 𝜏𝑥 = maximum delay.

𝜏𝑥 sometimes called the excess delay spread.

Coherence Bandwidth:

Delay spread is caused by reflected and scattered propagation paths in the radio
channel.

Coherence bandwidth is a statistical measure of the range of frequencies over

which the channel considered to be flat.

Other words, it is the range of frequencies over which two frequency components
have strong potential for amplitude correlation.
1
Bc = [correlation function above 0,9]
500𝑐

Frequency correlation function is above 0.5.

1
Bc =
50𝑐

Doppler spread and Coherence Time:

There parameters describe the time dispersive nature of the channel in a small-
scale region.

Doppler spread Bd is the measure of the spectral broadening caused by the time
rate of change of the mobile radio.

It is defined as the rage of frequencies over which the received Doppler spectrum
is non-zero.

When a frequency fc is transmitted, the received signal spectrum, called the

Doppler Spectrum and have components in the range of fc –fd to fc-fd where fd =
Doppler shift.

Spectral broadening depends on fd, is a function of the relative velocity of the

mobile and direction of arrival of the scatter waves.

If the baseband signal bandwidth > Bd, effect of Doppler spread are negligible at
the receiver. This is a slow fading channel.

Coherence time Tc, characterize the time varying nature of the frequency
depressiveness of the channel.
1
Tc =
𝑓𝑚
 Coherence time is the time duration over which two received signals have a
strong potential for amplitude correlation.

If correlation function is above 0.5.

9
Tc =
16𝜋𝑓𝑚

Where fm= maximum Doppler shift, fm =v/𝑣/

Fading Effects due to Multipath Time Delay spread:

Depending on the relation between the signal parameters and the channel
parameters, different transmitted signals will undergo different types of fading.

Multipath delay spread bads to time dispersion and frequency selective fading.

Doppler spread leads to frequency dispersion and time selective fading.

Time dispersion due to multipath caused the transmitted signal to undergo either
flat or frequency selective fading.

Flat Fading:

If the mobile ratios channel has a constant gain and linear phase response over a
bandwidth > bandwidth of the transmitted signal, then the received signal undergo flat
fading.

In flat fading, the multipath structure of the channels is such that the spectral
characteristics of the transmitted signal are preserved at the receiver.

The strength of the received signal changes with time, due to fulenations in the
gain of the channel caused by multipath

Small-scale fading

(Based on multipath time delay spread)

Flat fading Frequency selective fading

1. BW of signal < BW of channel 1. BW of signal > BW of channel

2. Delay spread < symbol period 2. Delay spread > symbol period
 Small-scale fading

(Based on Doppler spread)

Fast fading Slow fading

1. High Doppler spread 1. Low Doppler spread

2. Tc < Symbol period 2. Tc > Symbol period

3. Channel variations faster than 3. Channel variations slower than

baseband signal variations Base band signal variations.

If the channel gain changes over time, a change of amplitude occurs in the
received signal.

Over time, the received signal r(t) varies in gain.

In a flat fading channel, the received bandwidth of the transmitted signal is > the
multipath time delay spread of the channel.

Flat fading channels are also referred to the narrowband channels.

Since the bandwidth of the applied signal is narrow compared to the channel flat
fading bandwidth.

Flat fading channels cause deep fades, and thus require 20/30 dB. more
transmitter power to achieve low bit error rates.

The most common amplitude distribution is Rayleigh distribution.

Rayleigh flat fading channel model assumes that the channel includes an
amplitude which varies in time according to the Rayleigh distribution.

A signal undergoes flat fading if

Bs << Bc

Ts >> 𝝈𝝉

Where Ts=reciprocal bandwidth, Bs=bandwidth of te transmitted modulation,

𝝈𝝉 =rms delay spread, Bc = coherence bandwidth.
 s(t) h(t, 𝝉) r(t) s(t) h(t, 𝝉) r(t)

Fast fading channel Frequency selective fading

Frequency selective fading:

If the channel posses a constant gain and linear phase response over a bandwidth
ie. smaller than the bandwidth of transmitted signal, then the channel creates frequency
selective fading on the received signal.

Channel impulse response has a multipath delay spread > the reciprocal
bandwidth of the transmitted message wave form.

When this occurs, the received signal includes multiple versions of the
transmitted wave form which are attenuated and delayed in time, and hence the
received signal is distorted.

Frequency selective fading is due to time dispersion of the transmitted symbols

within the channel.

Thus the channel includes intersymbol interference.

For frequency selective fading, the spectrum s(f) of the transmitted signal has a
bandwidth > Bc.

Frequency selective fading is caused by multipath delays which exceeds the

symbol period of the transmitted symbol.

Frequency selective fading channels are also known as wideband channels since
the bandwidth of the signal is wider than the bandwidth of the channel impulse
response.

A signal is flat if Ts ≥ 10𝝈𝝉 and is frequency selective if Ts < 10 𝝈𝝉

Fading Effects due to Doppler spread:

Fast fading:
 Depending on the transmitted baseband signal channels, a channel may be
classified as a fast or slow fading.

In a fast fading channel, the channel in pulse response changes rapidly within
the symbol duration

ie TS > TC BS < BD

This causes frequency dispersion due to Doppler spreading which leads to signal
distortion.

Fast fading occurs for very low data rates.

Slow Fading:

In a slow fading channel, the channel impulse response changes at a rate much
slower than the transmitted baseband signal.

In the frequency domain, Doppler spread of the channel < bandwidth of the
baseband signal.

TS << TC

BS >> BD

The velocity of the mobile and the baseband signaling determines whether a signal
undergoes fast or slow fading.

Ts

Flat slow fading Flat fast fading

𝝈𝝉 Frequency Frequency selective

Selective slow fading fast fading

Ts

Bs

Frequency selective Frequency selective

 Fast fading slow fading

Bc

Flat fast fading Flat slow fading

Bs

Bd

UNIT II – CELLULAR ARCHITECTURE

Introduction:

Multiple access schemes are used to allow many mobile users to share
simultaneously a finite amount of radio spectrum.

In wireless communication systems, it desirable to allow the subscriber to

send simultaneously information to the base station while receiving information from
the base station.

In conventional telephone systems, it is possible to talk and listen

simultaneously, called duplexing.

Duplexing is done using frequency or time domain techniques.

Frequency Division Duplexing (FDD) provides two distinct bands of

frequency or time domain techniques.

The forward band provides traffic from the base station to the mobile and the
reverse band provides traffic from the mobile to the base station.

In FDD, any duplex channel consists of two simplex channel and a device
called a duplex is used inside each subscriber unit and base station to allow
simultaneously bidirectional radio transmission and reception for both the subscriber
unit and base station on the duplex channel pair.

The frequency separation between each forward and reverse channel is

constant.

Time Division Duplexing (TDD)

It uses time instead of frequency to provide both a forward and reverse link.

In TDD multiple users share a single radio channel by taking turns in the time
domain.
 Individual user are allowed to access the channel in assigned time slots, and
each duplex channel has both a forward time slot and a reverse time slot to facilitate
bidirectional communication.

If the time separation between forward and reverse time slot is small, then
transmission and reception of data appears simultaneously to the users at both the
subscriber unit and on the base station side.

Frequency division multiple access, time division multiple access and code
division multiple access are three major access techniques used to share the available
bandwidth.

These techniques can be grouped as narrowband and wideband systems,

depending upon the available bandwidth allocated to the users.

Narrowband is used to relate the bandwidth of a single channel to the expected

coherence band width of the channel.

In this system, the available radio spectrum is divided into a large number of
narrow band channels.

The channels are usually operated using FDD.

To minimize interference between forward and reverse links on each channel,

the frequency separation is made as great as possible.

Wideband systems, the transmission bandwidth of a single channel is much

target than the coherence bandwidth if the channel.

Thus multipath fading does not vary greatly the received signal power and
frequency selective fades occurs in only a small fraction of the signal bandwidth.

A large number of txers are allowed to transmit on the same channel.

 Reverse Forward

Channel Channel

Frequency separation

Reverse Forward

Channel Channel

Time separation

Frequency Division Multiple Access (FDMA)

FDMA assigns individual channels to individual users.

Each user is allocated a unique frequency band or channel.

The channels are assigned on demand to user.

During the period of the call, no other user can share the same channel.

In FDD system, the users are assigned a channel as a pair of frequencies; one
frequency is used for forward channel while the other frequency of FDMA:

Carrier only one phone circuit at a time idle and cannot be used by other user to
increase or share capacity.

After the assignment of a voice channel, the base station and mobile transmit
simultaneously and continuously.

The bandwidth of FDMA channels are narrow(30hz in AMPS) as each channel

supports only circuit per carrier.

Is of the narrowband signal is large compared to the average delay speed.

code
 C C C C C
h h h h h
a a a a a
n n n n n
n n n n n
e e e e e
l l l l l
1 2 3 4 n

Non-linear effects in FDMA:

In FDMA, many channels share the same antenna at the base station.

Power amplifiers or the power combiners when operated at or near saturation

for maxim power efficiency are non-linear.

The non-linearities cause signal spreading in the frequency domain and

generate enter modulate frequencies.

In is undesired RF radiation which can interface with other channels.

Spreading of the spectrum results in adjacent channel interference.

The first analog cellular system, the Advanced Mobile System (AMPS) based
on FDMA/FDD.

A single user occupies a single channel while the call is in progress and the
single channel is actually two simplex channels which are frequency duplexed with
45mhz split.

Voice signal are sent on the forward channel from the base station to mobile
unit, and on the reverse channel from the mobile to base station.

In AMPS, analog narrowband frequency modular is used to modulate the

carrier.

Bt- 2Bquard

N=
Bc
Where Bt = Total Spectrum allocation

Bquard = Guard band allocated at edge of allocated

Bc = Channel bandwidth.

Time Division Multiple Access (TDMA)

TDMA system divides the radio spectrum into time slots and in each slot only
one user is allowed to either transmit or receive.

Each user occupies a cyclically repeating time slot.

TDMA systems transmit data in a buffer and burst method, thus the
transmission for any users is non-continuous.

The transmission from various users is interlaced into repeat ins frame
structure.

A frame consists of a number of slots.

Each frame is made up of a preamble. On information message and tail bits.

In TDMA/TDD, half of the time slots in the frame. Information message would
be used for the forward link channels and half for the reverse link channels.

In TDMA/TDD systems, an identical or similar frame structure would be solely

for either forward or reverse transmission.

In TDMA frame the preamble contains the address and synchronization

information that both the base station and subscribers use to identify each other.

Guard times are utilized to allow synchronization of the receivers between

different slots and frames.

Features of TDMA:

TDMA shares a single carrier frequency with several users, where each user
makes use of non-overlapping timeslots.

Data transmission for users occurs in burst. This results in low battery
consumption.

Because of discontinuous transmissions in TDMA, hand off process is much

simpler for a subscribes unit an enhanced link control.
 Mobile Assisted Hand off (MAHO) can be carried out by a subscribes by
listening on an idle slot.

TDMA uses different time slots for transmission and reception, duplexers are
not required. Even if FDD is used, a switch rather than a duplexer inside the subscriber
unit is required to switch between transmitter and receiver using TDMA.

Adaptive equalization is usually nessory in TDMA.

Guard time should be minimized.

High synchronization overhead is required.

Efficiency:

It is a measure of the percentage of transmitted data that contains information

an opposed to provide overhead fir the access schema.

Boh = Nr.br + Nt. bp + Nt.bg + Nr.bg.

Bt = Tf.R

Efficiency,

Ŋf = (1-boh/bt) * 100

Number of the channels in TDMA:

The number of the TDMA channel slots can be found by multiplying the
number of TDMA slots per channel by the number of channels.

N = m (Btot – 2 Bquard)
code

Cn

C2

C1
Code Division Multiple Access (CDMA):

In CDMA systems the narrowband message signal is multiplied by a large

bandwidth signal called spreading signal.

The spreading signal is a pseudo noise code. Sequence that has a chip rate
which is orders of magnitudes greater than the data rate of the message.

All users in CDMA systems use the same carrier frequency and may transmit
simultaneously.

Each user has its own pseudorandom code word which approximately
orthogonal to all other code words.

The receiver performs a time correlation operation to detect only the specific
desired code word.

All other code words appear as noise due to auto correlation.

For detection of the message signal the receiver needs to know the code word
by the transmitter.

Each user operates independently with no knowledge of other users.

code

C1

C2

code

Cn f

T
 In CDMA, the power of the multiple users at a receiver determines the noise
floor after decor elate.

If the power of each user within a cell is not controlled such that they do not
appear equal at the base station receiver. Then the near-far problem occurs.

The near-far problem occurs when many mobile users share the same channel.

In general the strongest received mobile signal will capture the demodulator at
base station.

To combat the near-far problem, power control used is most CDMA

implementation.

Power control is provided by each base station in cellular systems and assures
that each mobile within base station coverage area provides the same signal level to the
station.

This shows the problem of a nearby subscriber over powering the base station
receiver and drawings out the signals of faraway subscribers.

Features of CDMA:

Many users of CDMA systems share the same frequency either TDD or FDD
maybe used.

CDMA has soft capacity limit. Increasing the number of users in CDMA raise
the noise floor in linear manner.

The systems performance gradually degrades for all users as the number of
users is increased and improves as the number of users is decreased.

Multipath fading may be reduced because the signal is spread over a large
spectrum. If the spread spectrum bandwidth of the channels.

Channel data rates are very high in CDMA. A RAKE receiver can be used to
improve reception.

Since CDMA uses co-channels alls, it can use macroscopic spatial diversity to
provide soft handoff.

Self-jamming is a problem in CDMA.

 The near-far undesired user has a high deteated power.

Capacity Calculations:

Channel capacity for a radio system can be defined as the maximum numbers
of channels or users that can be provided in a fixed frequency band.

Radio capacity is a parameter which measures spectrum efficiency of a wireless

system.

This parameter is determined by the required carrier to interference ratio (C/I)

and the channel band with (BC).

In a cellular system, the interference at a base system receiver will come from
the subscriber unit in surrounding cells caller reverse channel interference.

For a particular subscriber unit, the desired base station will provide the desired
forward channel. While the surrounding co-channel base stations will provide forward
channel interference.

Cellular Capacity:

Cellular systems for mobile communications implement SDM each transmitter,

typically called base station (BS), covers a certain wire area called as cell. Cellular
concepts is a system level idea which calls for replacing a single, high power
transmitter (large cell) with many low power transmitter (small cell), each providing
coverage to only a small of serivce

2 3

4
5
Advantages of cellular services:

 Higher capacity
 Less transmission power
 Robustness
 Local interferences only.

Disadvantages of cellular services:

 Complex infrastructure needed

 Landover needed
 Frequency planning

Frequency Reuse Concept:

Each cellular base station is allocated a group of radio channels to be used

within a small geographic area called a cell. Base stations in adjacent cells are an
assigned channel group which contains completely different channels than neighboring
cells.

If a given set of frequencies can be received without increasing the interference

then the geographical areas covered by a single base station with high transmitting
power antennas can be divvied into small areas each allocated with subset of
frequencies.

The base station antennas are designed to achieve the desired coverage within a
particular cell. The design process of selecting and allocates channel groups for all of
the cellular base station within a system is called frequency reuse or frequency
planning.

The adjacent cells having same frequency is called co-channels cells.

Analysis of frequency reuse concept:

Consider a cellular system which has a total of a duplex channels available for
use. If each cells are allocated a group of K channels (K<S), and if the S channel are
divided among N cell into unique and disjoint channel groups which each have the
same number of channels.

S = KN
 The N cells which collectiouly use the complete set of available frequencies is
called a cluster. If the cluster is replicated „M‟ times within a system. Then the total
number of duplex channels, C can be used as a measure of capacity and is given by

C = MKN

C = MS [⁖ S = Kn]

The capacity of a cellular system is directly proportional to the number of times

a cluster is replicated in a fixed service area. The factor N is called the cluster size.

The frequency reuse factor of a cellular system is given by 1/N

D = 1/N

D/R = 3𝑁

Channel Assignment Strategies:

Channel assignment schema determines on assignment of channels to base

station such that frequency reuse is maximized for a given set of channels or
frequencies (F) and set of base stations (B) in coverage area.

Channel Assignment

Fixed channel Dynamic channel Hybrid channel

Simple flexible Random channel weighted carrier

Borrows Borrows DCA ordering ordering

Fixed channel Assignment Strategy (FCAS)

In this strategy each cell is pre-allocated to a pre-determined set of voice

channels. If all the channels in that cell are occupied. Then the excess call is blocked
and subscriber does not receive service.

Borrowing strategy:
 A cell is allowed to borrow channels from a neighboring cell if all the channels
are already occupied with no interference is called as borrowing strategy.

Simple borrowing:

Channel blocking performance suffers under heavy traffic conditions.

Flexible borrowing:

Some fixed channel set of cell is divided into two groups. One group for local
user only other group of channels for borrowing.

Dynamic channel Assignment Strategy (DCAS)

In DCAS voice channels are not pre-allocated to different cells permanently.

The serving base station requests a channel from MSC instead of each time a call
request is made. All channels are potentially available to cells and are assignment to
cells dynamically as cell arsives. Channels are temporally assigned for use in cells for
duration of call.

To avoid ECI any channel that in use one cell can only be reassigned
simultaneously to another cell in the system if the distance b/w two cells in large than
minimum reuse distance .

Random DCA:

Here available channels are randomly assigned which has per channel
utilization.

Channel ordering:

Here a cell can use any channel, but each has a different ordering. Channel with
the highest priority is selected for the cell.

Weighted carrier ordering:

Here cell develops “favorite” channels from post experience. This scheme
adapts faster to traffic changes from DCA and CO but needs more time to search for
highest priority channel that cause delay.

Hybrid channel Assignment:

It divided the total number of channels into two groups one of which is used for
fixed allocation to cells, which other is kept as central pool to be shared by all users.

Hand –off:
 As the mobile moves around, it is quite possible that it approaches the edges of
the cell. This is the point of which radio signal is to weak.

Hand-off is a technique used to continue the call establishment in mobile

communication when the subscriber users more towards one BTS to next BTS.

System designer must specify an optimum signal level at which to initiate

hand-off.

=Pr, hand-off – Pr, minimum usable. is too large unnecessary hand-offs

which burden the MSC may occur.

is too small they may be insufficient time to complete a hand-off before a

call is lost due to weak

R Level at point A
e
c
e
Hand-off Threshold
i
v
e
d
Minimum acceptable
S/R
S
/ Level at point B
R

Time

R
e
c Level at point B
e
Types of handoff:

1. Hard handoff
2. Soft handoff

Soft handoff:

Soft handoff is defined as abilities to select between instantaneous received

signals from a variety of base station.

Hard handoff:

Hard handoff, unlike channelized wireless systems that assign different radio
channels during a handoff, spread spectrum mobiles share the same channel is every
cell.

Handoff prioritization:

One of the ways to reduce the handoff failure rate is to prioritize handoff. Two
basic of handoff prioritization.

1. Guard channels handoff request

2. Queuing channels handoff request.

Guard channels:

It improves the portability of successful handoff by reasoning a fixed.

Advantages of guard channels:

It offers efficiency spectrum utilization when dynamic channel assignment

strategies.

It minimize the number of required guard channels by efficient demand based

allocation are used.

Disadvantages of guard channels:

 Only fewer channels are allocated to originating cells so its has disadvantage of
reducing the total carried traffic.

Queuing of handoff:

Queuing is a way of delaying handoff, MSC queues the handoff requests

instead of denying access if the candidate Bs is busy. Queuing new calls results in
increased handoff.

Cell Dragging:

When there is a line of sight radio path between the base station and subscriber
in urban environment then the cell dragging results from pedestrian users that provide a
new strong signal to the station.

Interference and system capacity:

Interference is the important factor, which degrades the performance of cellular

systems. Due to an undesired transmission interference on voice channels causes cross
talk where the subscriber hears interference in the background.

Interference from other mobiles residing at same cell is called intercell

interference. Interferences between different cells is called intercell interference.

 Mainly interference are due to the following factors:

 Another mobile in the same cell.
 A call conversion in a neighboring cell.
 Neighboring base stations operating in same frequency band.

Two major types of system-generated cellular interference are:

1. Co-channel interference

2. Adjacent channel interference

Co-channel interference:

Frequency reuse implies that in a given coverage areas are seven cells that use
the set of frequencies. These cells that use the set of frequencies. These cells are called
co-channels cell. Co-channel interference is defined as the interference between signals
from these cells.

When the each cell size is same and each base station transmit the same power
the co-channel interference ratio is independent of the transmitted power and the
function of the radius of the cell (R) and distance between contuse of the nearest co-
channel cells (D).

The parameter Q called co-channel reuse ratio is related to the cluster size.

ӨL = D/R = 3𝑁

Adjacent channel interfence:

It is caused by adjacent channels. This occurs when imperfect receive filters

allow nearby frequencies to leak in to the pass band. This problem enhance when the
adjacent channel user is transmitting in a close range.

S/I = (20)-n

Trunking and grade of service:

Concept of trunking allows a large numbers of users to share a relatively small

number of channels in a cell by providing access to each user on demand from a pool of
available channels

There are two types of trunked systems

1. Blocked calls cleared

2. Blocked calls delayed

Blocked calls cleared:

There is no queuing for call requests. For every user when requests service, it is
assumed there is no setup time and the user is given immediate access to a channel if
one is available. If no channels are available the requesting user is blocked without
access and is free to try again later.
𝐴𝑐
𝐶!
Pr(Blocking) = 𝐴𝑘
= Gos (Erlag B)
𝑘=0 𝐾 !

Blocked calls delayed:

Gos is defined as the probability that a call is blocked after waiting a specific
length of time in the queue.
 𝐴𝑐
Pr (delay > 0) = 𝐴 𝐴𝑘
𝑐+𝐶![1− ] 𝐶−1
𝐴 𝐶 𝑘=0 𝑘!

Pr [D>T] = pr[D>0] pr[D≥T] p[D>0]

𝐻
D avg = pr[delay>0] .
𝐶−𝐴

Coverage and capacity improvements:

Cellular systems are designed in such a manner they provide coverage to large
area as well as will accommodate large no. of users as possible. But in some case it
may be difficult to predict the need for network expansion. At this situation cellular
design techniques must be incorporated to accommodate more users as possible i.e.,
channel capacity must be improved. There are four basic capacity techniques available.

They are:

1. Cell splitting
2. Sectoring
3. Repeaters for extending range
4. Micro cell zone method

Cell splitting:

Cell splitting is the process of splitting a cell into smaller cells and the
frequencies are redistributed in such a way interference is eliminated. In this method
each cell is split into microcells which then own base station rescaling to the newer cell
sizer. This includes reduction in the antenna height as well the reduction in the
transmitted power.
 Cell splitting is very useful when the is cell congested, because smaller cells
provides way for more number of BS (Base Station), there by increasing thechannel
that can be used in a particular cell and increases the capacity. This concept helps in the
effective frequency reuse and Allows the system to grow without upsetting the channel
allocation. BS is usually placed at the cell boundaries. The cell also posses larger BS
thereby providing less hand off splitting is its cost, since BS are placed in a large
manner cost increases. But it provides the tradeoff between the channel capacity.

In order to understand the effect of interference let us consider a cell radius of

the new cells and reduced by half, then the transmitted power for the new cells is given
by,

Pr (old cell boundary) = PTR-n

When n – path loss exponent

𝑅
Pr (new cell boundary) = PT( )-n
2

𝑃𝑇1
Power of new cell PT2 =
2𝑛

If the pathloss exponent is taken as 3, then power of new cell will be

𝑃𝑇1
PT2 =
8

From the above equation, the transmitted power of new cell will be 9dB less
than the original transmit power. This avoids the interference effect in the cell splitting.

Sectoring

Omniolirectional Sectoring 600 Sectoring 1200

In sectoring, the cell have the same coverage space but instead of using a single
omnidirectional antenna within a cell, 3 (or) 6 directional antenna, are used so that
sectoring of 600, 900 or 1200 can be achieved. Sectoring methods reduces the D/R ratio
and keeps the R untouched. In sectoring technique each antenna will be radiating in a
specific region there by reducing the effort of interference.
 In sectoring the cells are divided into wedge shaped sectors each having its own
set of channels. These results in the increased number of user, thereby contributing to
the spectral efficiency. The main drawback of sectoring is it decreases the trunking
efficiency. It also results in increased of antennas as well as produces unnecessary
handoff‟s.

Repeaters

Repeaters are used for extending the range of the signal. They are mainly
incorporated in places which are hard to reach such as subways, underground buildings,
mountains, etc.,. repeaters does not have additional channels, they amplify the signal as
well as reradiate it to the unreachable locations. They are usually bidirectional in nature
i.e., can transmit and receive at the same time. The drawbacks of repeaters are since
they amplify the signal, the noise associated with the received signal is also reradiated.

Microcell zone method

Microcell zone method is used in order to overcome the drawback of sectoring.

Since cells are sectored in sectoring method it provides unnecessary handoff‟s. This is
eliminated in microcell zone method. In this method each cell in divided into zone or
microcells. Each zone are connected with the help of optical cable to the base station.
Important factor in microcell is, all the zones have the same channel frequency which is
offered by different directional antenna. This eliminates the need for hand off since the
user will be maintained under a single BS which are connected to each other. The
antenna‟s are usually placed at the edge of the zone, which are fed with low power.
This concept is useful when it is used in highway since it will eliminates the number of
call drops due to hand off.
UNIT III

DIGITAL SIGNALING FOR FADING CHANNELS

MODULATION TECHNIQUES

Modulation is the process of changing the parameters of the carrier signal, in accordance with
the instantaneous values of the modulating signal. Modulation may be analog or digital.

ANALOG MODULATION

Analog modulation is the process of converting an analog input signal into a signal that is
suitable for RF transmission. Here both information and carrier are analog signals.

Advantages of Analog Modulation:

1. Less tolerance by the term noise.

2. Flexibility with bandwidth.
3. You can rectify faulty components easily.
4. Easy to manipulate using mathematical formations and calculation.
5. Great lifespan.
6. Ambient weather Dependencies are low.

Disadvantages of analog system:

1. Not easy to implement.

2. Needed perfect receiver and Transmitter for specific communication scenario. If you move
into a new system and you want to change the analog signal you need to tune or change both
receiver and Transmitter.
3. No security for transmission data.
4. Can't be saved and transmit under urgency.

DIGITAL MODULATION

Digital modulation is the process of translating bits to analog waveforms that can be sent over
a physical channel. However the wireless channel is analog in nature so analog waveform of the data
has to be transmitted. For this reason, the digital modulator at the transmitter (TX) has to convert the
digital source data to analog waveforms. At the receiver (RX), the demodulator tries to recover the
bits from the received waveform.
METRICS FOR DIGITAL MODULATION

1. POWER EFFICIENCY (ɳp)

It is a measure of how much signal power should be increased to achieve a particular BER for
a given modulation scheme. It is the ability of a modulation technique to preserve the fidelity of the
digital message at low power levels.

Designer can increase noise immunity by increasing signal power. Power efficiency is a
measure of how much signal power should be increased to achieve a particular BER for a given
modulation scheme. i.e., signal energy per bit/noise power spectral density (Eb/N0).

2. BAND WIDTH EFFICIAENCY(ɳB)

It defines the ability to accommodate data within a limited bandwidth. It reflects how
efficiently the allocated band width is utilized. It is the ratio of the throughput data rate per Hertz in a
given bandwidth.

ɳB= R/B bps/Hz

Where, R is the data rate in bits per second.

B is the bandwidth occupied by the modulated RF signal.

3. SHANNON LIMIT

Shannon‟s channel coding theorem states that an arbitrarily small probability of error, the
maximum possible bandwidth efficiency is limited by the noise in the channel, and is given the
channel capacity formula.

Channel capacity (c)

ƞ𝒎𝒂𝒙 = = log2 (1 + S N)
Band width (B)

Where C is the channel capacity (in bps),

B is the RF bandwidth,

S / N is signal to noise ratio.

4. PROBABILITY OF ERROR

In digital communications it is desirable to minimize the average probability of bit error at the
receiver subject to the constraints on received power and channel bandwidth. The terms
probability of bit error (Pe) and bit error rate (BER) are used interchangeably. Advantage of
pass band transmission is optimum design of receiver so it reduces the average probability of
error in presence of additive white Gaussian Noise (AWGN).
SELECTION OF MODULATION SCHEME

Considerations are required in choice of selecting Modulation Scheme

1. High spectral efficiency

2. High power efficiency
3. Robust to multipath efforts
4. Low cost and ease of implementation

ADVANTAGHES OF DIGITAL COMMUNICATIONS

1. The digital communication has mostly common structure of encoding a signal so devices
used are mostly similar.
2. The Digital Communication's main advantage is that it provides us added security to our
information signal.
3. The digital Communication system has more immunity to noise and external interference.
4. Digital information can be saved and retrieved when necessary while it is not possible in
analog.
5. Digital Communication is cheaper than Analog Communication.
6. The configuring process of digital communication system is simple as compared to analog
communication system. Although, they are complex.
7. In Digital Communication System, the error correction and detection techniques can be
implemented easily.

DISADVANTAGHES OF DIGITAL COMMUNICATIONS

1. Generally, more bandwidth is required than that for analog systems.

2. Synchronization is required.
3. High power consumption (Due to various stages of conversion).
4. Complex circuit, more sophisticated device making is also drawbacks of digital system.
5. Introduce sampling error.
6. As square wave is more affected by noise, That‟s why while communicating through
channel we send sin waves but while operating on device we use square pulses.

STRUCTURE OF A WIRELESS COMMUNICATION LINK

Transceiver Block Structure

The goal of a wireless link is the transmission of information from an analog information
source via an analog wireless propagation channel to an analog information sink. The digitizing of
information is done only in order to increase the reliability of the link. The transmission can then add
redundancy in the form of a forward error connection code, in order to make it more resistant to error
introduced by the channel. The encoded data are then used as input to a modulator, which maps the
data to output wave form that can be transmitted. By transmitting there symbol on specific frequencies
or at specific times, different user can be distinguished.
 At the Rx, the signal is received by one or more antennas. If the channel is delay dispersing,
then an equalizer can be used to reverse that dispersion and eliminated inter symbol interference. Then
the single is demodulated and a channel decoder eliminates the errors that are present in the resulting
bit stream.

Block diagram of a radio link with digital Transmitter

Signaling

Information Source ADC Source coder Channel coder MUX

source

Local OSC

Tx filter Up converter Low pass filter Transmit D/A Base band

modulator

noise

Propagation
channel

Figure 3.1 Block diagram of a radio link with digital Transmitter

INFORMATION SOURCE

The information source provides an analog source signal and feed it into the source
ADC. The ADC band limits the signal from the analog information source and then converts
the signal into digital data.

SOURCE CODER

The source coder uses information on the properties of the source code in order to
reduce redundancy in the source signal. This reduces the amount of source data to be
transmitted and thus the required transmission time and bandwidth.
 CHANNEL CODER

The channel codes add redundancy in order to protect data against transmission errors.
This increases the data rate that has to be transmitted at interference. The channel coder often
uses information about the statistics of error sources in the channel to design a coder that are
well suited for the channel. Signaling adds control information for the establishing and ending
of communication users, synchronization etc...

MULTIPLEXER

The multiplexer combines user data and signaling information and combines the data
from multiple users. This is done by time multiplexing.

BASEBAND MODULATOR

The baseband modulator assigns the gross data bits, to complex transmit symbol in the
baseband. Spectral properties, inter symbol interference, peak to average ratio and other
properties of the transmit signal are determined in this step.

DIGITAL TO ANALOG CONVERTER

The transmitter digital to analog converter [DAC] generates a pair of analog, discrete
amplitude voltage corresponding to the real and imaging part the signal.

LOW-PASS FILTER

The analog low-pass filter in the transmitter side eliminates the spectral component
outside the desired transmission bandwidth. These components are created by the out-of-band
emission of a base band modulator.

LOCAL OSCILLATOR

The TX local oscillator provides an unmodulated sinusoidal signal corresponding to

one or the admissible center frequency of the considered system. The requirement for the
frequency stability depends on the modulation and multiple access method.

UP-CONVERTER

The up-converter converts the analog filtered baseband signal to a pass band signal by
mixing it with the local signal.

Block diagram of a digital receiver chain for mobile communication:

Rx filter Down Rx A/D Base band

fs
Base band filter
converter demodulator
Figure 3.2 Block diagram of a digital receiver chain for mobile communication:

The RF Tx filter eliminates out-of-band emission in the RF domain. Even if the low-pass filter
succeeded in eliminating all out-of band emission. The propagation channel attenuates the signal and
leads to delay and frequency dispersion.

The Rx filter performs a rough selection of the secured band. The bandwidth of the filter
corresponds to the total bandwidth assigned to a specific source.

LOW-NOISE AMPLIFIER

The low-noise amp amplifies the signal, so that the noise added by later components of the Rx
chain has less effect on the SNR.

RX LOCAL OSCILLATOR

The Rx local OSC provides sinusoidal signal corresponding to possible signals at the Tx local
OSC. The Rx down converter converts the required signal into baseband. The Rx low pass filter
provides a selection of desired freq bands for one specific user.

RX A / D CONVERTER

The RX A / D converter converts the analog signal into value that are discrete in time and
amplitude. The carrier recovery determines the frequency and phase of the causes of the received
signal.

BASEBAND DEMODULATOR

The baseband demodulator obtains soft decision data from digitized baseband data and hands
them over to the decoder. The symbol timing recovery uses demodulates data to determine an estimate
of the duration of symbol and uses it to fine-tune sampling intervals. The decoder uses soft estimates
from the demodulator to find the original source data.
SIGNALING RECOVERY

Signaling recovery identifies the parts of the data that represent signaling information and
control the subsequent de-multiplexer. The de-multiplexer separates the user data and signaling
information and resource possible time comparison of the Tx multiplier. The source decoder
reconstructs the source signal from the rules of source coding.

Simplified block:

Digital data source Base band Equivalent time-

modulator discrete LP channel

Digital data sink decoder Base band

demodulator

CLASIFICATION OF DIGITAL MODULATION TECHNIQUES

One possible classification of digital modulation techniques depends on whether receiver uses
coherent detection or not. These are:

1. Coherent Techniques
2. Non- Coherent Techniques

Digital Modulation Technique

Coherent Non-Coherent
 Figure 3.3 Hierarchy of Digital Modulation Technique

1. Coherent Techniques

Receiver is equipped with phase recovery circuit, i.e., locally-generated carrier at the Rx, is
synchronized in frequency and phase to received carrier. Thus the detection is done by correlating
noisy signal and locally generated carrier. Thus the coherent detection is a synchronous detection.

Examples:

 BPSK
 QPSK
 M-ARY PSK
 M-ARY QAM
 M-ARY FSK

2. Non-Coherent techniques

Used whenever it is impractical to maintain carrier-phase synchronization. The transmitter and

receiver are not required to be synchronized in both carrier phase and bit timing.

Examples:

 M-ary ASK
 M-ary FSK
 BFSK
 DSPK

Here no need to exactly recover phase but only need to recover phase difference another
possible classification of digital modulation techniques is

1. Binary Techniques
2. M-ary Techniques

1. Binary Techniques:
 The information bits {0, 1} are represented by one signal each, namely s1(t) and s2(t). The
symbol duration (T) is equal to the bit duration, T= Tb. The bandwidth is proportional to 1/Tb.

2. M-ary Techniques:

The information bits {0, 1} are grouped into M distinct n-bit symbols, represented by M
signals, namely S1(t),S2(t),….,Sm(t). Where M=2n. The symbol duration (T) is equal to the bit
duration, T=nTb. The bandwidth is proportional to 1/nTb. Instead of transmitting one bit at a time,
M-ary two or more bits are transmitted simultaneously. It reduces channel bandwidth.

OVERVIEW OF MODULATION

QUADATURE PHASE SHIFT KEYING (QPSK)

Principle:

In QPSK, as with BPSK, information carried by the transmitted signal is contained in the
phase. It provides efficient utilization of channel bandwidth. It has twice the bandwidth efficiency of
BPSK, since 2 bits are transmitted in a single modulation symbol. QPSK is equivalent to two coherent
BPSK systems working in parallel and using two carriers that are in phase quadrature. In quadrature
phase shift keying, 4 possible phase locations are used at a time. Two data bits can thus be transmitted
simultaneously. One of the data bits produces the in-phase (I) component and the other data bit
produces the Quadrature (Q) component.

QPSK SIGNAL REPRESENTATION

𝜋 3𝜋
The phase of the carrier takes on 1 of 4 equally spaced values, such as 0, 2 , 𝜋 and 2 . The
QPSK signal for this set of symbol states may be derived as

2𝐸𝑠
SQPSK = 𝑇𝑠
cos [2πfc+(2i-1)π/4] 0 ≤ t ≤ Ts

2𝐸𝑠
Or 𝑇𝑠
cos [2πfc+(i-1)π/2 ] 0 ≤ t ≤ Ts

E = Es= 2Eb

i.e., Ts = 2Tb

Es = 2Eb represents the signal energy per symbol is twice the signal energy per bit.

Where, Es - Transmitted signal energy per symbol.

Es - Bit energy

Tb - Bit duration
 Ts is the symbol duration and is equal to twice the bit period.

fc = nc/T

fc = carrier frequency

Quadrature phase-shift keying(QPSK) is a form of phase shift keying (PSK) using four phase state,
normally 90 degrees apart (π/4, 3π/4, 5π/4 and 7π/4).

QPSK SIGNAL SPACE REPRESENTATION

The above equation can be rewritten as follows by using trigonometric identity,

2 2
SQPSK(t)= cos[(2i-1)π/4]cos(2πfct)- sin[(2i-1)π/4]sin (2πfct)
𝑇𝑠 𝑇𝑠

Where, i = 1, 2, 3, 4

Orthogonal basis functions ϕ1 (t) and ϕ2 (t) can be

2
ϕ1 (t) = cos (2πfct) 0 ≤ t ≤ Ts
𝑇𝑠

2
ϕ2 (t) = sin (2πfct) 0 ≤ t ≤ Ts
𝑇𝑠

Where, ϕ1(t) and ϕ2(t) are two orthogonal basis function.

Equation can be rewrite as follows,

2 2
SQPSK = 𝐸𝑆 cos [(2i-1) π/4] cos (2πfct) - 𝐸𝑠 sin [(2i-1) π/4] sin (2πfct)
𝑇𝑠 𝑇𝑠

Then the 4 signals in the set can be expressed in terms of the basis signals as

SQPSK(t) = { 𝐸𝑠 cos [(2i-1) π/4] ϕ1 (t) - 𝐸𝑠 sin [(2i-1) π/4] ϕ2 (t)}

Where, i = 1, 2, 3, 4

The four message points and the associated signal vectors are defined by

SQPSK = [ 𝐸𝑠 cos (2i-1) π/4 - 𝐸𝑠 cos (2i-1) π/4], i = 1, 2, 3, 4

The carrier frequency fc equals nc/T for fixed integer nc. Each possible value of the phase corresponds
to a unique bit (pair of bit) as shown. We many choose the foregoing set of phase value to represent
the Gray – encoded set of dibits: 10,00,01 and 11, where only a single bit is changed from one di-bit
to the next. The elements of single vector have their values summarized in table
 Gray Coded Input Phase of QPSK Coordinate of message points
Dibit (Radians)
S11 S12

10 π/4 𝐸 𝐸
+ 2
- 2
𝐸 𝐸
00 3π/4 - 2
- 2

𝐸 𝐸
01 5π/4 - 2
+ 2

𝐸 𝐸
11 7π/4 + +
2 2

Signal space characterization

Phase2

Decision
Region Z1
Region Z2 boundary

𝐸/2

Message Message
point m2 (00) point m1 (10)
Phase1
0 𝐸/2
− 𝐸/2

Message Message
point m3 (01) point m4(11)

Region Z4
Region Z3
− 𝐸/2

Signal Space Diagram for QPSK

Constellation diagram
 Distance between adjacent points is 2𝐸 s

1 Symbol = 2 bit

Es = 2Eb

QPSK Transmitter

Polar NRZ Encoder

Here incoming data sequence is first given into Polar non return to zero level encoder for polar
conversion. Incoming symbols are represented by + 𝐸 b and - 𝐸 b by and for 1 and 0 respectively.

Next these binary waves are divided by de-multiplexer, which separate the incoming binary
wave into two components namely odd and even components. This two binary waves amplitudes are
represented by a1 (t) and a2 (t) for si1 and si2 respectively.

Product Multiplier

Then this two binary waves with amplitudes a1 (t) and a2 (t) are multiplied with orthogonal basis
functions ϕ1 (t) and ϕ2 (t). Product modulator 1by the signal Si1 with ϕ1 (t) then it gives output as Si1 ϕ1
(t). Product modulator 1by the signal Si2 with ϕ2 (t) then it gives output as Si2 ϕ2(t).

2
Φ1(f)= 𝑇
cos (2πfct)
a1(t)
x

Binary data Polar non return- QPSK

to-zero level De-multiplexer Ʃ signal
sequence
encoder

a2(t) 2
Φ2(f)= 𝑇
cos (2πfct)
QPSK Transmitter
QPSK Waveform

Then output of both product modulator 1 and 2 are given into summer which
sums the both signal then it gives output as follows

S(t) = si1ϕ1(t) + si2ϕ2(t)

Before that we have to know that output of each product modulator is BPSK output only.

Summary

After adding the output of both product modulators 1 and 2 we can get QPSK output from this
we may know that by adding two BPSK signal we may get QPSK. Two successive bits are combined
two form distinct bits.

2π
Phase shift =
Number of symbol

2𝜋
=
4
= 90 o

M = 4 so QPSK is called as M-ary PSK.

The dimensionality of a modulation is defined by the number of basis functions used. That makes
QPSK as two dimensional signals. Not because it sends two bits per symbol, but because it uses two
independent signals (a sine and a cosine) to create the symbols.
 QPSK Coherent Detection
It consists of a pair of correlates with a common input from QPSK transmitter output.
Correlators consist of product modulator and integrator. Product modulators take common inputs from
QPSK transmitter output and locally generated carrier as we used in QPSK transmitter.
Low pass filter operation are actually performed in integrated which evaluates the area under
the signal produced at the multiplier output. The integration is performed for the bit interval 0 ≤ t ≤ Tb
with the sample value x1 and x2.
The decision is made by decision device

 If x1˃0, decision is made in favor of symbol 1 for the in-phase channel output, but
x1˂0, decision is made in favor of symbol 0.
 If x2˃0, decision is made in favor of symbol 1 for the quadrature channel output, but
x2˂0, decision is made in favor of symbol 0.
Finally these two binary sequences at in-phase and quadrature channel outputs are added in a
multiplexer to reproduce the original binary sequence at the transmitter.

ϕ1(t) = 2𝜋 cos(2πfct) Threshold = 0

((2πfct)
X1
𝑇𝑏
Decision
𝑑𝑡
0 Circuit

Recovered
Signal s(t)

BPF Carrier Symbol Multiplexer

Recovery Recovery
Circuit System Recovered
signal
90 Degree
Phase Shift
X2
𝑇𝑏 Decision
𝑑𝑡 Circuit
0

Threshold = 0
ϕ1(t) = 2𝜋 sin(2πfct)
((2πfct)
PROBABILITY OF ERROR
In QPSK system, the received signal x(t) is defined by
x (t) = si(t) + w(t), {0≤t≤T
i=1, 2, 3, 4
Where w(t) = sample function of White Gaussian noise process with zero mean and power spectral
density N0/2. Then the components x1 and x2 in x(t) are defined by follows
𝑇
X1 = 0
𝑥(𝑡) ϕ1 (t) dt

Substitute equation

𝑇 2
= 0
𝐸 cos ((2i-1)π/4) 𝑇
cos (2πfct) dt

= 𝐸cos ((2i-1)π/4) + w1

𝐸
X1 =± 2
+ w1

𝑇
And X2 = 0
𝑥(𝑡) ϕ1 (t) dt

Substitute equation

𝑇 2
= 0
−𝐸 sin ((2i-1)π/4) 𝑇
sin (2πfct) dt

= - 𝐸sin ((2i-1)π/4) + w2 (⁖sin (2i -1)π/4=1/ 2)

𝐸
X1 =± 2
+ w2

As we know that QPSK is equivalent to two BPSK working in parallel manner with the following
property.

 The single energy per bit is E/2.

 The noise spectral density is n0/2.

Then the average probability of bit error in each channel of QPSK is

𝐸
P‟ = 1/2erfc [ /𝑁0 ] = 1/2erfc [ 𝐸/2𝑁0]
2

The average probability of a correct decision by decision rule resulting from combined action of the
two channels working together is

Pc = (1-p‟) 2 = [1-1/2 erfc [ 𝐸/2𝑁0]2

= 1- erfc [ 𝐸/2𝑁0] + 1/4 erfc2 [ 𝐸/2𝑁0]

The average probability of symbol error for QPSK is

Pe = 1 - pc

= erfc [ 𝐸/2𝑁0] – 1/4 erfc2 [ 𝐸/2𝑁0]

By ignoring quadratic equation in RHS since e/2N0 >1, we may get

 Pe, QPSK = erfc [ 𝐸/2𝑁0]

QPSK transmits two bits per symbol; the transmitted energy per symbol is twice the single energy per
bit i.e.

E = Es = 2Eb

Substitute equation

Pe, QPSK = erfc [ 2𝐸𝑏/2𝑁0] = erfc [ 𝐸𝑏/𝑁0]

Characteristics of QPSK

1. BER

Average BER can be defined as ratio of total number of erroneous bits to total number of bits
transmitted.

No of erroneous bit
Average BER =
Total no. of bits transmitted

BER = 1/2 pe = 1/2 ( 𝑒𝑟𝑓𝑐 (𝐸𝑏/𝑛0))

BER = Q 2𝛾𝑏

Probability of error for QPSK is same as for BPSK.

2.Bandwidth

In QPSK odd and even components are baseband signals. So one bit period for both of these
signals are equal to 2tb. So bandwidth is

Bandwidth = 2(1/2Tb)

BW = fb

From these we may know that bandwidth of QPSK is half of the BPSK.

PSD

A)The PSD of QPSK can be derived as follows

P QPSK = Es/2 [(sin π (-f –fc) Ts/ π (-f – fc)Ts)2 + (sin π (-f –fc) Ts/ π (-f – fc) Ts) 2]

E2 = 2Eb

Substitute equation

Where, Tb is bit duration

By replacing symbol period is by bit period Tb,

P QPSK = 2Es/2 [(sin 2π (-f –fc) Ts/ 2π (-f – fc)Ts)2 + (sin 2π (-f –fc) Ts/ 2π (-f – fc) Ts) 2]

= Eb [(sin 2π (-f –fc) Ts/ 2π (-f – fc)Ts)2 + (sin 2π (-f –fc) Ts/ 2π (-f – fc) Ts) 2]

Since f > fc, by considering the first term only, P QPSK becomes.

= Eb (sin 2π (-f –fc) Ts/ 2π (-f – fc)Ts)2

Let fs = -f –fc. Then

P QPSK = Eb (sin 2π(fs)Tb/2π(fs)Tb)

P QPSK = 4Eb sinc2 (Tf)

B) Depending on bits sent during the signaling interval, the in-phase components equals +g(t)
or –g(t). The g(t) denotes the symbol shaping function. It can be expressed as follows

𝐸
g(t) = { 𝑇
0≤t≤T

0 Otherwise

This is normalized with respect to 4Eb.

1.2

0.8

0.6 MSK
QPSK
0.4

0.2

0
0.25 0.5 0.75 1

Power spectra of QPSK and MSK single

π/4 QPSK
The π/4 shifted QPSK modulation is a quadrature phase shift keying technique which offers a
compromise between OQPSK and QPSK in terms of the allowed maximum phase transitions.

PRINCIPLES
 In π/4 QPSK, the maximum phase change is limited to ±1350 as compared to 1800 for QPSK.
Hence, the band-limit π/4 QPSK single preserves the constant envelope property better than base-
limited QPSK, but is more susceptible detected. Which envelope variations than OQPSK.When
differently encoded, π/4 QPSK is called π/4DQPSK.

π/4 QPSK TRANSMISSION TECHNIQUES.

Serial to parallel converter (s/p)

 The input bit stream is partitioned by a serial-to-parallel (s/p) converter.

 It splits data streams into two parallel data streams as ml.k and mq,k.
 Sysmbol rate equal to half that of the incoming bit rate i.e Rs=Rb/2.

Signal mapping circuit

The kth in-phase and quadrature phase, Ik and Qk are produced at the output of the signal mapping
circuit. Time is in the following range kT ≤ t ≤ (k+1)T and the kth in-phase and quadrature phase can
be determined from previous values Ik-1 and Qk-1 as well as Өk

Qk is a function of ϕk which is a function of input symbols mlk and mqk

Ik and Qk represent rectangular pulses over one symbol duration having amplitudes given by

Ik = cos Өk = Ik-1 cos ᵠk - Ik-1 sin ᵠk

Qk = sin Өk = Ik-1 sin ᵠk - Qk -1 sin ᵠk

Where Qk = ᵠk-1 + ϕk; Өk and Өk-1 are phase of the kth and (k-1)st symbol.

Modulator

The in-phase and quadrature bit streams Ik & Qk are then separately modulated by 2 carriers which are
in quadrature with one another, to produce the π/4 QPSK waveform given by

S π/4 QPSK (t) = I(t) cos(wct) – Q(t) sin(wct)

𝑛−1 𝑛−1
I(t) = 𝑘=0 Ik − p(t-KTs – Ts/2) = 𝑘=0 cos ᵠkp(t-KTs – Ts/2)
𝑛−1 𝑛−1
Q(t)= 𝑘=0 Qk − p(t-KTs – Ts/2) = 𝑘=0 sin ᵠkp(t-KTs – Ts/2)

Where, p(t) is pulse shape and Ts is symbol period.

 Cos 𝜔ct

LPF I(t)

ml.k lk I(t)
mk Serial to Signal Amplifier
parallel Ʃ
mapping Q(t
convertor ) π/4 QPSK
Qk Q(t)
M signal
q.k LPF

-sin 𝜔ctt

QPSK transmission Techniques

All possible states

Constellation diagram of π/4 QPSK

Carrier phase shift corresponding to various input bit pairs

Information bits ml,k , mQ,k Phase shift φk

11 π/4
01 3π/4
00 -3π/4
10 -π/4

 Both Ik and Qk are passed through raised cosine rolloff pulse shaping filters before modulation,
in order to reduce the bandwidth occupancy.
  Peak amplitude of the wave forms I(t) and Q(t) can take one of the five possible values, 0, +1,
-1, +1/ 2, -1/ 2

Summer

It adds two I(t) and Q(t) yo give π/4 QPSK signal.

Amplifier

After summer, the π/4 QPSK signal is then given into Amplifier. The signal is amplified by non-
linear amplifier with greater efficiency. From the above discussion it is clear that the information in a
π/4 QPSK signal is completely contained in the phase difference φk of the carrier between two
adjacent symbols. Since the information is completely contained in the phase difference, it is possible
to use non-coherent differential detection even in the absence of differential encoding.

π/4 QPSK Detection

Due to simple hardware implementation, differential detection is often employed to demodulate π/4
QPSK signals. In an AWGN channel , the BER performance of a differentially detected π/4 QPSK
is about 3 dB inferior to QPSK, while coherently detected it π/4 QPSK has the error performance as
QPSK.

Types of Detection Techniques

There are Various types of detection techniques that are used for the detection are π/4 QPSK
signals.

1. Baseband differential detection,

2. IF differential detection,
3. FM discriminator detection.
4.
1. Baseband Differential Detection

The incoming π/4 QPSK signal is quadrature demodulated using two local oscillator signals.
Local oscillator, carrier frequency has the same frequency as the unmodulated carrier at the
transmitter, but not necessarily the same phase. It is important to ensure the local receiver oscillator
frequency is the same as the transmitter carrier frequency, and that it does not drift.

If φk = tan-1〔Qk/Ik〕 is the phase of the carrier due to the kth data bit, the output wk and z k
from the two low poss filters in the in-phase and quadrature arms of the demodulator can be expressed
as
Wk = cos (φk - ɤ)
Zk = sin (φk - ɤ)
where ɤ is a phase shift due to noise, propagation, and interference.
The phase ɤ is assumed to change much slower than φk so it is essentially constant.

In phase Channel
wk xk
 Block diagram of a base band differential detector
The two sequences wk and zk are passed through a differential decoder which operates on the
following rule.
Xk = wk wk-1 + zk zk-1
yk = zk wk-1 - wk zk-1
The output of the differential decoder can be expressed as follows by substituting the equations,
wk = cos (ɸk - ɤ )
zk = sin (ɸk - ɤ )
xk = cos(ɸk - ɤ) cos(qk-1) + sin(ɸk - ɤ) sin(ɸk-1- ɤ)
Since cos (A-B) = cosAcosB + sinAsinB
Above equation becomes,
xk = cos(φk - φk-1)
yk = sin(φk - ɤ) cos(φk-1- ɤ) + cos(φk - ɤ) sin(φk-1- ɤ)
= sin(φk - φk-1)

Decision Device
The output of the differential decoder is applied to the decision circuit.
SI = 1, If xk > 0 or SI = 0, If xk < 0
SQ = 1, If yk > 0 or SQ = 0, If yk < 0
Where SI and SQ are the decoder bits in the in-phase and quadrature arms, respectively.
 Multiplexer selects any one of SI and SQ.

Advantages baseband differential detection

1. Hardware implementation is simple.

2. Non-Coherent detection is possible since information is completely contained in
phase difference φk of carrier.

Drawbacks baseband differential detection

Any drift in the carrier frequency will cause a drift in the output phase which will lead to BER
degradation.

2. IF Differential Detector

IF differential detector shown in figure 3.17 avoids the need for a local oscillator by using a
delay line and two phase detectors.

LPF Decision
device

Sampled at maximum
output for every TS
BPF Ts Multiplex

Modulated IF 90 Sampled at maximum Demodulated

0
input signal output for every TS output

LPF Decision
device

IF Differential detector for π/4 QPSK

IF Modulator

The received signal is converted to IF and is band pass filtered.

BPF

The band pass filter is designed to match the transmitted pulse shape, so that the carrier phase is
preserved and noise power is minimized the effect of ISI and noise, the bandwidth of the filters are
chosen as 0.57/ TS

Differential Decoder
 The received IF signal is differentially decoded using a delay line and two mixers. The
bandwidth of the signal at the output of the differential detector is twice that of the baseband signal at
transmitter end.

Decision device

The output of the differential decoder is applied to the decision circuit.

SI = 1, If xk > 0 or Si = 0, If xk< 0

SQ = 1, If yk> 0 or SQ= 0, If yk<0

where SI and SQ are the detected bits in the in-phase and quadrature arms, respectively.

3. FM Discriminator

Figure 3.18 shows a block diagram of an FM discriminator detector for π/4 QPSK.

BPF

The input signal is first filtered using a band-pass filter is matched to the transmitted signal.

Limiter

The filtered signal is then hard-limited to remove any develop fluctuations. Hard-limiting
preserves the phase changes in the input signal and hence no information is lost.

FS Discriminator

The FM discriminator extracts the instantaneous frequency deviation of the received signal.

Integrate and dump circuit

Signal for FS Discriminator is passed in integrate and dump. The demodulated signal is
integrated over each symbol period gives the phase difference between two sampling instants.

Four level threshold device

The phase difference is then detected by a four level threshold comparator to obtain the original
signal. The phase difference can also be detected using a modulo-2π phase detector. The modulo-2π
phase detector improves the BER performance and reduces the effect of click noise.

Parallel to serial convertor

Demodulated signal is passed through parallel to serial converter.

FM Discriminator for π/4 QPSK

/4 QPSK BPF Limiter FM Discriminator
ignal
 Demodulated
waveforms

FM Discriminator detector for π/4 DQPSK demodulator

Disadvantages of FM Discriminator

 Occupies large bandwidth.

 Poor bandwidth efficiency.

OFFSET QPSK (STAGGERED QPSK)

Necessity of Offset QPSK

In QPSK, the amplitude is ideally constant. However, when QPSK signals are pulse shaped,
they lose the constant envelop property which results in

1. Regeneration of side-lobes
2. Spectral widening

QPSK signals is amplified only using linear amplifiers, which are less efficient. A modified form of
QPSK, called offset QPSK (OQPSK) or staggered QPSK.

Examining the QPSK waveform we may make the following observations

1. The carrier phase changes by ±180 degrees whenever both the in-phase and
quadrature components of the QPSK signal changes whenever the in-phase or
quadrature component changes sign.
2. The carrier phase changes by ±90 degrees whenever the in-phase or quadrature
component changes sign.
3. The carrier phase is unchanged when neither the in – phase nor the quadrature
component changes sign.

These shifts in carrier phase can result in changes in the carrier amplitude, thereby causing
additional symbol errors on detection. The extent of amplitude fluctuations exhibited by
QPSK signals may be reduced by using offset QPSK, where the bit stream responsible for
generating the quadrature component is delayed by half a symbol interval with respect to the
bit stream responsible for generating the in-phase component.

Definition

OQPSK (offset QPSK) is a special version of QPSK in which the transmitted signal has no
amplitude modulation. It results to 180o shifting in the phase. If the two bit streams are offset by a ½
interval, then the amplitude fluctuations are minimized since the phase never changes by 180 o as its
occurred in QPSK when debit 01 changes to 10. This modulation is called as offset QPSK.
 Signal Representation

Offset QPSK is obtained from QPSK by delaying the odd bit stream by half a bit interval with
respect to the even bit stream.

OQPSK can be represented as follows

SQPSK(t) = 2𝐸 s/Ts cos[(2i-1)π/4] cos(2πfct) - 2𝐸 s/ Ts sin[(2i-1)π/4] sin(2πfct)

Above signal representation is similar to QPSK except for time alignment of even and odd bit
streams.

Principles of offset QPSK

In QPSK signaling, the bit transitions of the even odd bit stream occur at the
same time instants, but in OQPSK signaling, the even and odd streams, m I(t) and mQ(t) are offset in
theft relative alignment by one bit period(half-symbol period). This is shown in the waveform.

Offset QPSK Block diagram

In OQPSK, the incoming signal is divided in the modulator into two portions I and Q which
are then transmitted shifted by a half symbol duration, so that I and Q channel signals do not transition
at the same time.

The result of this sample change is that phase shifts at any one time are limited and hence
offset QPSK is more “constant-envelop” than straight QPSK.

I Channel

Rb/2
Set1 Pulse X
Amplitude shaping

Informati ˷
on bit ƩƩ
stream Serial to Cos (2πfcTs)
parallel
converter π/2 QPSK
delay (or)
OQPSK
R1=Rb/2
Cos(2πfcTb carrier
+π/2)
Set1 Pulse Tsdelay X
Amplitude shaping
Rb/2
Q Channel

Offset QPSK
 d1(t)

+1 d0 d6

0 t

d2 d4

-1

-T 0 T 3T 5T 7T

dQ(t)

+1

d1 d5 d1 t

d3

-1

0 2T 4T 6T 8T

Time offset of QPSK

Elimination of 180o phase transitions


Due to time alignment of m1(t) and mQ(t) in QPSK, phase transition occur only
once every Ts = 2Tb seconds.
 In QPSK , if there is a change in the value of both m1(t) and mQ(t), then there
will be a maximum of phase shift of 180o
 In OQPSK signaling, bit transitions(and hence phase transitions) occur every Tb.
 Since the transitions instants of mI(t) and mQ(t) are offset, at any given time only
one of the two bit streams can change values. This implies that the maximum
phase shift of the transmitted signal at any given time is limited to ±90o.
 Hence, by switching phases more frequently (i.e., every Tb instead of 2Tb)
OQPSK signaling eliminates 180o phase transitions.
Probability of error
Probability of error is same as for QPSK
Advantage of offset QPSK
1. Offset QPSK offers better performance in satellite applications.
2. It performs better than QPSK in presence of phase jitter.
Application of offset QPSK
1. In high power amplifiers and for certain satellite applications.
2. Mobile communication systems.
Differential QPSK
DQPSK transfers the data by mapping the two bits digital signal to one of the four modulated
phase pattern. With incoming data stream in group of two bits, the modulator selects the
phase from 0, π/2,π/4,3π/4 and add current phase with previous phase to form a
differential encoded signal. The In-phase and Quadrature-phase (I and Q) signals are obtained
by multiplying the to the cosine and negative sine. Both I and Q signals are transmitted and
received by pulse shaping filter gT(t) at modulator and gR(t) at demodulator to maximize the
SNR. The original data is recovered at phase comparator, which accepts current and delayed I
and Q signals as input.

DQPSK modulator block diagram

DQPSK demodulator block diagram

MINIMUM SHIFT KEYING (MSK):

 Minimum shift keying (MSK) is a special type of continuous phase-frequency shift keying
[CPFSK] where in the peak frequency deviation is equal to ¼ the bit rate. In other words, MSK is
continuous phase FSK with a modulation index of 0.5

A modulation index 0.5 corresponds to the minimum frequency spacing that allows two FSK
signals to be coherently orthogonal. The name minimum shift keying implies the minimum frequency
separation (i.e., Bandwidth) that allows orthogonal detection.

Different ways of interpretations are:

1. The first interpretation is as CPFSK with a modulation index.

Hmod = 0.5 ,fmod = 1/4T

This implies the phase changes by ± π/2 during a 1-bit duration.

2. Alternatively, we can interpret MSK as offset QAM (OQAM), basis pulses are sinusoidal
half-waves extending over a duration of 2TB.
g(t) = sin (2π fmod(t + TB)) gR (t, 2TB)

Due to the use of smoother basis functions the spectrum decreases faster than regular
OQPSK.

16𝑇𝐵 cos ⁡
(2𝜋𝑓 𝑇𝐵
S(f) = ( )
𝜋2 1−16 𝑓 2 𝑇𝐵2

MSK is only a binary modulation format, white OQPSK transmits 2 bits per symbol duration.
MSK is sometimes referred to as fast FSK, as the frequency spacing used is only half as much
as that used in conventional non-coherent FSK.

MSK is a spectrally efficient modulation scheme and is attractive for use in mobile radio
communication systems.

Properties of MSK:

(i) Constant envelope

(ii) Spectral efficiency
(iii) Good BER performance
(iv) Self – synchronizing capability
 Power spectrum of MSK is obtained by frequency shifting the magnitude squared of the
fourier transform of the base band pulse shaping function.

MSK has lower spectral efficiency when the energy bandwidth is 90% (1.29 bit/s/Hz),
but still performs well when the energy bandwidth is 99% (0.85 bit/s/Hz).

MSK modulator:

Multiplying a carrier signal with cos [nt/2T] produces two phase-coherent signals at fc +
X(t)
1/4 T and fc – 1/4 T. X(t)
BPF fc+1/4T X(t)

- +
Cos 2πfct MSK signal
ml(t
)
S(t)

Cos 2πfct
+ +
BPF fc-1/4T
 MSK modulator

These two FSK signals are separated using two narrow band filters and appropriately combined to
form the in-phase and quatrature carrier components x (t) and y(t) respectively.

These carriers are multiplied with the odd and even bit streams, m I(t) and mQ(t), to produce the MSK
modulated signal S(t).

MSK demodulator:

The received signal s(t) in the absence of noise and interference is multiplied by the respective
in-phase and quadrature carriers x(t) and y(t).

The output of the multipliers are integrated over two bit periods and dumped to a decision
circuit at the end of each two bit periods.

Based on the level signal at the output of the integrator, the threshold detector decides whether
the signal is a 0 or a 1.

Threshold=0

integrator Decision device

Ml(t)
T=2(k+1)
X(t) T

Smsk(t) X(t)

integrator Decision device

MQ(t)

T=2(2k+2)T
Threshold=0

MSK demodulator

The output data streams are mi(t) and mq(t), which are combined to obtained the demodulation signal.

Different demodulator structures correspond to different interpretation:

The different interpretations of MSK are not useful for gaining insights into the modulation
scheme, but also foe building demodulator.

(i) Frequency Discriminator: MSK is a type of FSK, it is straight forward to

checkWhether the instantaneous frequency is larger or smaller the carrier frequency.
The instantaneous frequency can be sampled in the middle of the bit, or it can be integrated
over bit duration in order to reduce the effect of noise.
 (ii) Differential Detection: the phase of the signal changes by +π/2 or -π/2 over a 1- bit
duration, depending on the bit that was transmitted. The receiver, require the
differential encoding of the transmitted signal to estimate bit errors.
(iii) Matched filter reception: when considering MSK as OQPSK and multi plus
modulation and MSK is modulation format with memory. The bit duration is
suboptimum.
Memory can be a memory likelihood sequence estimation.

Advantages
(i) Constant envelope
(ii) Spectral efficiency
(iii) Self synchronization
(iv) Good BER performance

Disadvantages
(i) The generation and detection of MSK signal is slightly complex.
(ii) Phase jitter is present because of incorrect synchronization.
(iii) The bandwidth requirement of MSK is high i.. 1.5 fb whereas fb in QPSK.

GAUSSION MINIMUM SHIFT KEYING [GMSK] :

While MSK has many advantages, its special efficiency is rather low. The drawback is
eliminated by GMSK [Gaussian minimum shift keying]

GMSK is CPFSK [Continuous Phase Frequency shift keying] with modulation index h mod=0.5,
the difference is the basis pulses are Gaussian pulses.

𝑔(t) = gG (t, TB, BGT)

So, the sequence of transmit phase pulse is

∞
PD(t) = 𝑖=−∞ 𝑏𝑖 𝑔(t-iTB) = bi* 𝑔*(t)

1
Mobile radio channels have an irreducible error rate due to mobile velocity the GMSK has irreducible
error rate is less than that produced by the mobile channel, there is no penalty is using GMSK.

GMSK is most attractive for its excellent power efficiency (due to the constant envelope) and
its excellent spectral efficiency.

Desirable properties can be achieved by passing a non-return-to-zero (NRZ) binary data stream
through a baseband pulse-shaping filter whose impulse response is defined by a Gaussian function.
The resulting method of binary frequency modulation is naturally referred to as Gaussian shift keying.

The probability of error of GMSK is given by

2ɤ𝐸𝑏
Pe = Q
𝑁0

Where

Eb= bit energy

No = power spectral density

ɤ = constant related to BT (Bandwidth – bit duration product)

GMSK transmitter and receiver GMSK transmitter:

To generate a GMSK is to pass a NRZ message bit stream through a Gaussian baseband filter
having a response, followed by a FM modulator.

NRZ Gaussian low FM GMSK

data pass filter modulator output

GMSK transmitter

GMSK is used in a variety of analog and digital implementation for the US Cellular Digital
Packet Data [CDPD] and Global System for Mobile [GSM]

GMSK Receiver:
 GMSK signal can be detected using orthogonal coherent detectors (or) with simple non-
coherent detectors.

LPF

π/2

Loop
- filter 𝜋 +
+
Modulated Demodulate
if input π/ d signal
signal
2

LPF

Clock
recovery

GMSK Receiver

Carrier frequency is recovered by Buda‟s method, where the sum of the two discrete frequency
components contained at the output of a frequency decide. It is divided by four.

Bit error rate

The Bit error probability is given by

2𝛾𝐸𝑏
𝑃𝑒 = 𝑄
𝑁0

Where 𝛾is a constant related to BT

B is bandwidth

T is message symbol duration

Eb is transmitted bit energy and

N0 is noise spectral density

0.68 𝑓𝑜𝑟 𝐺𝑀𝑆𝐾 𝑤𝑖𝑡𝑕 𝐵𝑇 = 0.25

𝛾=
0.85𝑓𝑜𝑟 𝑠𝑖𝑚𝑝𝑙𝑒 𝑀𝑆𝐾 𝑤𝑖𝑡𝑕 𝐵𝑇 = ∞
Advantages
(i) It has narrow power spectral.
(ii) It has sharp cutoff characteristics.
(iii) It reduces adjacent channel interference

Modulation Technique For large value of Fb/N0

For coherent binary PSK, For large value of Eb/N0

1 1 Pe = 1/4⎾ (for whereat BPSK)
Pe= 2 1 − 1+⎾
For coherent binary FSK, Pe = 1/2⎾ (for coherent FSK)
1 1
Pe= 2 1 − 2+⎾
For differential binary PSK [DPSK] Pe = 1/2⎾ (for DPSK)
⎾
Pe= 2(1+⎾)

For non-coherent orthogonal binary PSK Pe=1/⎾

⎾ (for non-coherent orthogonal binary FSK)
Pe= 2(1+⎾)

𝐸𝑏
⎾=
𝑁0

Methods for the computation of Error Probability

Error Probability for coherent Receivers

Coherent receivers compensate for phase rotation of the channel by means of carrier recovery.
Furthermore, the channel gain α is assumed to be known, and absorbed into the received signal, so
that in the absence of noise, r = s holds. The probability the symbol sy is mistaken for symbol sk
that has Euclidean distance djk from sj (pairwise error probability) is given as

𝑑 2 𝑗𝑘 𝐸
Prpair (sj,sk) = Q =Q (1 − 𝑅𝑒 𝜌𝑗𝑘 ) ----------------------(1)
2𝑁0 𝑁0

Where the Q-function is defined as

1 ∞ 𝑡2
Q(x) = 𝑥
exp − 𝑑𝑡
2𝜋 2

This is related to the complementary error function:

1 𝑥
Q(x) = 𝑒𝑟𝑓𝑐( 2)
2

And
1
Erfc(x) = 2 exp⁡
(−𝛾𝑏/2)
It is found by computing the probability that the noise is large enough to make the received signal
geometrically closer to the point sk in the signal space diagram, even though the signal sj was
transmitted.

Error Probability for Coherent Receivers – Binary Orthogonal Signals

The modulation formats can be viewed as binary orthogonal signals – most prominently, binary
frequency shift keying (FSK) and binary pulse position modulation (PPM). The decision boundary: if
a received signal point falls into the shaded region, then it is decided thata+1 was transmitted,
otherwise a-1 was transmitted.

Defining the signal-to-Noise Ratio (SNR) for one symbols as 𝛾𝑠 = 𝐸𝑠/𝑁𝑜, we get

Prpair (sj,sk) = Q( 𝛾𝑠(1 − 𝑅𝑒{𝜌𝑗𝑘 }))

= Q ( 𝛾𝑠)

Error Probability for Differential Detection

Differential detection is an attractive alternative to coherent detection, because it renders

irrelevant the absolute phase of the detected signal. The receiver just compares the phases (and
possibly amplitudes) of two subsequent symbols to recover the information in the symbol; this phase
difference is independent of the absolute phase.

If the phase rotation introduce by the channel is slowly time varying (and thus effectively the
same for two subsequent symbols), it enters just the absolute phase, and thus need not be taken into
account in the detection process.

For differential detection of Phase Shifting Keying (PSK), the transmitter needs to provide
differential encoding. For binary symmetric PSK, the transmit phase ϕi of the ith bit is
𝜋
+ 𝑖𝑓 𝑏 𝑖 =+1
Φi = ϕi-1 + 𝜋
2
− 𝑖𝑓 𝑏 𝑖 =−1
2

Comparison of the difference between phases on two subsequent sampling instances determines
whether the transmitted bit bi was +1 or -1.

For Continuous Phase Frequency Shift Keying (CPFSK), such differential encoding can be
avoided. Remember that in the case of MSK (without differential encoding), the phase rotation over a
1-bit duration is ±π/2.

For binary orthogonal signals, the BER for differential detection is

1
BER = 2 exp(-γb)

For 4-PSK with Gray-coding, it is

1 1
BER = QM (a,b) - I0(ab) exp (- 2 𝑎2 + 𝑏 2 )
2

Where
 1 1
a= 2𝛾𝐵 (1 − ), b = 2𝛾𝐵 (1 + )
2 2

and QM(a,b) is Marcum‟s Q-function:

∞ 𝑎 2 +𝑏 2
QM(a,b) = 𝑏
𝑥 exp − I0(ax) dx
2

Whose series representation is given by Marcum‟s Q-function.

Error probability in Flat-Fading Channels

Average BER- Classical Computation Method

In fading, channels the received signal power (and thus the SNR) is not constant but
changes as the fading of the channel changes. In many cases, we are interested in the BER in a fading
channel averaged over the different fading states. For a mathematical computation of the BER in such
a channel, we have to proceed in three steps:

1. Determine the BER for any arbitrary SNR.

2. Determine the probability that a certain SNR occurs in the channel – in other words,
determine the pdf of the power gain of the channel.
3. Average the BER over the distribution of SNRs

In an AWGN channel, the BER decrease approximately as the SNR increases:

For the binary modulation formats, a 10-dB SNR is sufficient to give a BER on the order of 10-4, for
15dB the BER is below that in a fading channel the BER decreases only linearly with the (Average)
SNR. At first glance, this is astonishing: sometimes fading leads to high SNRs, sometimes it leads to
low SNRs, and it could be assumed that high and low values would compensate for each other. The
important point here is that the relationship between (instantaneous) BER and (instantaneous) SNR is
highly nonlinear, so that the cases of low SNR essentially determine the overall BER.

Error Probability in Fading Channels

For AWGN channels, the advantages of the alternative representation of the Q – function are
rather limited. They allow a simpler formulation for higher order modulation formats, but do not
exhibit significant advantages for the modulation formats that are mostly used in practice. The real
advantage emerges when we apply this description method as the basis for computations of the BER
in fading channels.

The average over the pdf of the SNR pdfɤ (ɤ), we have now seen that the alternative
representation of the Q function allows us to write the SER in the generic form:
Ѳ2
SER(ɤ) = Ѳ1
𝑓1 Ѳ exp⁡
(−ɤ𝑓2 (Ѳ)) d Ѳ−

Thus, the average SER becomes

∞
SER = 0
𝑝𝑑𝑓ɤ (ɤ) SER (ɤ) dɤ
 ∞ Ѳ
= 0
𝑝𝑑𝑓ɤ (ɤ) Ѳ 2 𝑓1 Ѳ exp⁡
(−ɤ𝑓2 (𝜃)) d 𝜃 dɤ
1

Ѳ2 ∞
= Ѳ1
𝑓1 Ѳ 0
𝑝𝑑𝑓ɤ (ɤ) exp(−ɤ𝑓2 (𝜃)) dɤ d 𝜃

Frequency Dispersion

Consider error due to frequency dispersion. For FSK, it is immediately obvious how
frequency dispersion leads to error: random frequency Modulation (FM) leads to a frequency shift of
the receiver signal, and can push a bit over the decision boundary. Assume that a+1 was sent (i.e., the
frequency fc +fmod). Due to the random FM effect, the frequency fc+ fmod finst is received. If this is
smaller than fc, the receiver opts for a-1.Note that instantaneous frequency shifts can be significantly
larger than the maximum Doppler frequency even though the statistics of the random FM are
determined by the Doppler spectrum of the channel. Consider the following equation for the
instantaneous frequency:
𝑑𝑟 𝑡
𝐼𝑚 (𝑟∗ )
𝑑𝑡
Finst(t) = |𝑟(𝑡)|2

In deep fading dips lead to large shift in the instantaneous frequency, and thus higher error probability.

A somewhat different interpretation can be given for differential detection. The detection assumes that
the channel does not change between two adjacent symbols.

However, if there is a finite Doppler, then the channel does change-remember that the Doppler a
spectrum gives statistical description of channel changes. Thus, a nonzero Doppler effect implies a
wrong reference phase for differential detection. If this effort is strong, it can lead to erroneous
decision. Also in this case it is true that channel changes are strongest near fading dips.

Delay Dispersion

In contrast to frequency dispersion, delay dispersion has great importance for high-data-rate systems.
This becomes obvious when we remember that the errors in un-equalized systems are determined by
the ratio of symbol duration that is disturbed by Inter Symbol Interface (ISI) to that of the un-
disturbed part of the symbol.

The maximum excess delay of a channel impulse response is determined by the environment, and
independent of the system: let us assume in the following a maximum excess delay of 1𝜇𝑠. In a
system with a symbol duration of 20 𝜇𝑠, the ISI can disturb 5% of each symbol, while it can disturb
20% if the symbol duration is 5 𝜇𝑠.

Many theoretical and experimental investigations have shown that the error floor due to delay
dispersion is given by the following equation:
𝑆
𝐵𝐸𝑅 = K(𝑇 𝜏 )2
𝐵

Where 𝑆𝜏 is the rms delay spread of the channel. Just as for frequency dispertion, errors mainly occur
near fading dips. An interpretation of this fact in terms of group delay, which reaches its largest value
near fading dips.
Equation is only valid if the maximum excess delay of the channel is much smaller than the symbol
duration, and the channel is Rayleigh fading.

ORTHOGONAL FREQUENCY DIVISION MULTIPLEXING [OFDM]:

OFDM is a modulation scheme. It is suitable for high-data-rate transmission in delay-

dispersive environments. It convert a high-rate data stream into a number of low-rate streams that are
transmitted over parallel, narrowband channels that can be easily equalized.

Advances in hardware for digital signal processing made OFDM a realistic option for wireless
communication systems. OFDM is used for Digital audio broadcasting (DAB), digital video
broadcasting (DVB), and wireless local area networks [LANs].

Principal of OFDM:

Principle – OFDM splits the information into N parallel streams which are then transmitted
by modulating N distinct carries (called Sub-carriers or tones). Symbol duration on each subcarrier
thus becomes targerby a subcarriers they have to be orthogonal.

In FDMA [Conventional Frequency Division Multiple Access], large frequency spacing

between carriers, due to this wastes of precious spectrum by the carriers, whereas in OFDM much
narrower spacing of subcarriers can be achieved.

Let subcarriers be at the frequencies, fn = nW/N

Where
n – is an integer
N – different carriers.
W – total available bandwidth
𝑁
Total available bandwidth, W = 𝑇𝑠

Where
Ts – symbol duration.
Carrier spacing

FDMA:

OFDM:
Assume, modulation on each of the subcarriers is pulse amplitude modulation (PAM) with rectangular
pulses. Subcarriers are mutually orthogonal, the relationship is
𝑖+1 𝑇𝑠
𝑖𝑇𝑠
exp 𝑗2𝜋 𝑓𝑛𝑡 𝑑𝑡 = 𝛿𝑛𝑘

In time, due to the rectangular shape of pulses, the spectrum of each modulated carrier has a
sin (𝑥)
shape.
𝑥

The spectra of different modulated carriers overlap, but each, carrier is in the spectral nulls of
all other carriers. Therefore, the data stream of any subcarriers will not interface at the receiver during
demodulation process.

Transceiver implementation:

OFDM can be constructed in two ways.

(i) Analog construction

(ii) Digital construction

(i)Analog construction:

S/P conversion [serial to parallel conversion]: Data streams are first split into N parallel data
streams each of which has a lower data rate.

Modulator: We have a number of local oscillators. It oscillators at a frequency

𝑛𝑊
𝑓𝑛 = 𝑁

Each of the parallel data streams then modulates one of the carriers. Actual implementation
hardware effort of multiple local oscillators is too high. Let the complex transmit symbol at time
instant i on the nth carrier be cn, i.

Receiver
Transmitter
Channel
Ck0 Ck0
The transmit signal is
∞ ∞ 𝑁−1
𝑆 𝑡 = 𝑖=−∞ 𝑆𝑖 𝑡 = 𝑖=−∞. 𝑛=0 𝐶𝑛,𝑖 𝑔𝑛 (𝑡 − 𝑖𝑇𝑠 )

Where

gn(t) – is the basis pulse

cn,i - is the nth carrier
TS - is the symbol duration.
The frequency – shifted rectangular pulse.

1
𝑔𝑛 𝑡 = exp(𝑗2𝜋 𝑇𝑡𝑠 ) ; for 0<t<Ts
𝑇𝑠

; otherwise

Let us consider the signal only for i=0, and sample it at instances tk=kTs/N.
1 𝑁−1 𝑘
𝑆𝑘 = 𝑆(𝑡𝑘 ) = 𝑛=0 𝐶𝑛,0 exp(𝑗2𝜋𝑛 𝑁 )
𝑇𝑠

This is nothing but the inverse discrete Fourier transform of the transmit sysmbols.

Therefore, the transmitter can be realized by performing an Inverse Discrete Fourier

Transform (IDPT) on the block of transmit symbols. [the block size must equak the number of
subcarriers].

Receivers: we can reverse the process of modulation done at the transmitter.

The received signal is demodulated by using locally generated subcarriers. Then, each signal
is serially converted by parallel to serial (p/s) conversion. The result is an estimate 𝐶𝑛 of the original
data Cn.

Requirements of analog construction of OFDM:

It would require multiple local oscillators each of which has to operate with little phase noise
and drift in order to retain orthography between the different subcarriers.

It would require serial to parallel converter and parallel to serial converter.

Disadvantages:
  It is less efficient than digital
 Slow in process, hardware complexity is too high.
 Low sensitivity to time and frequency dispersion.

(ii)Digital Construction:

The data streams are divided into blocks on N symbols. This block of N data symbols are
subjected to an Inverse Fast Transformations (IFFT) and then transmitted.

Transmitter
Channel
Ck,0

Inverse Parallel to
S(t)
Serial to Ck,1 Fast serial
Data
parallel Fourier
source H
conversion
(IFFT)
Hs(t)
Ck,n-1 Transform

𝐶 k,0
FFT Serial to
parallel
𝐶 k,1 [FastFouri
Parallel to
Data er
serial
sink Transform
conversion
]

𝐶 k,n-1

(K+1) T

Receiver

The input to IFFT is made up of n samples [the symbols for the different subcarriers], and therefore
the output from the IFFT also consists of N values.

These N temporal sample values to be transmitted using parallel to serial converter, one offer
the other.

Receiver:
 We can reverse the process the received is to parallel signal converted into seriual using S/P
converter. Then perform FFT on this vector, this resulted signal is an estimate of Cnoriginal data
from τn.

Advantages of digital construction:

(i) It is much simpler and cheaper.

(ii) It have fast process.
(iii) Highly efficient “butterfly structures” of FFT are used.
(iv) Low frequency dispersion.

Cyclic prefix [CP]:

Delay dispersion is OFDM leads to a loss of orthogonality between subcarriers and thus to
inter carrier interference [ICI]

The negative effects can be eliminated by a special type of guard interval called the Cyclic
Prefix [CP]. The performance can be achieved in frequency selective channels.

Cyclic Prefix:

CP means creating a cyclically extended guard interval whereby each symbol sequence is
preceded by a periodic extension of the sequence itself.

Specifically, the last V samples of the symbol sequence are repeated at the beginning of the
sequence being transmitted.

S[-k] = S [N –k] for k= 1,2,…..v

This condition is called cyclic prefix. It is needed for frequency selective channels AWGN
[Additive White Gaussian Noise Channels].

Principle of the cyclic prefix:

Let us first define a new base function for transmission,

𝑔𝑛 𝑡 = exp(𝑗2𝜋 𝑊𝑁 𝑡)for –Tcp< t<𝑇𝑠

𝑊
Where - is the carrier spacing
𝑁

𝑁
𝑇𝑠 = 𝑊

Ts – is symbol duration

Ts - 𝑇𝑠 + 𝑇𝑐𝑝

The base function means that for duration 0<t<𝑇𝑠 the “normal” OFDM symbol is transmitted.
During time - 𝑇𝑐𝑝 < t < 0, a copy of the last part of the symbol is transmitted.
 Sk
Cyclic prefix

-Ncp N-Ncp
K
-1 N-1

𝑁
gn(t) = gn [t+𝑊 ]. This pretended part of the signal is called the “cyclic prefix” shown in above
diagram.

When transmitting any data stream over a delay – dispressive channel the arriving signal is the
linear convolution of the transmitted signal with the channel impulse response. The CP converts this
Linear Convolution into a cyclical convolution. During the time 0<t<τmax, where τmaxis the maximum
excess delay of the channel.

The received signal suffers from ISI [Inter symbol Interference], as echoes of the last part of
the preceding symbol interfere with the desired symbol. This ISI is eliminated by discarding the
received signal during this time interval.

During the remainder of the symbol, we have cyclical ISI. It is the last part of the current (not
the preceding) symbol that interfaces with the first part of the current symbol.

The block diagram of an OFDM system with cyclic prefix[CP]

The original data stream is serial to parallel (s/p) converted.

each block of n data symbol is subjected to an IFFT and then the last NTcp/Ts samples are
preponded.
 Ck,0

Inverse
Serial to Ck,1 Fast
Data Parallel Addition of
parallel Fourier
source cyclic prefix
conversion to serial
(IFFT)

Transform Conversio
Ck,n-1
n S(t)

𝐶 k,0 Channel
1-tap
equalizer Fast Serial
Serial to Fourier Stripping
Data to
parallel 1-tap of cyclic
source Transform H
conversion 𝐶 k,1 equalizer Parallel prefix
1 (FFT)
Conversio Hs(t)
𝐶 k,, N-1
1-tap n
,n-1 equalizer

The resulting signal is modulated onto a (single) carrier and transmitted over a channel, which distorts
the signal and adds noise.

At the receiver, the received signal is partitioned into books. For each block the cyclic prefix
(CP) is removed and the remainder is subjected to an FFT.

The resulting samples are equalized by means of one-tap equalization that is division by the
complex channel attenuation on each carrier.

Performance in frequency – selective channels:

The cyclic prefix (cp) converts a frequency – selective channel into a number of parallel flat-
fading channels. This is positive in the sense that it gets free from ISI that plagues TDMA and CDMA
systems.

If a subcarrier is in a fading dip, then error probability is very high, and dominates the BER
[Bit Error Rate] of the total system for high SNRs.

Performance of OFDM can be improved by any of the following approaches:

 (1) Coding across the different tones: such coding helps to compensate for fading dips on
one subcarrier by a good SNR in another subcarrier.
(2) Spreading the signal over all tones: each symbol is spread across all carriers. So, SNR is
the average of all tones over which it is spread.
(3) Adaptive modulation: if the transmitter knows the SNR on each of the subcarriers, it can
choose its modulation alphabet and coding rate adaptively.

Thus, low SNR, the transmitter will send symbols using stronger encoding and a smaller
modulation alphabet, The power allocated to each subcarrier can be varied.

Channel estimation:

OFDM systems requires an estimate of the channel transfer function or the channel impulse response.
the good channel estimators are

(i) Pilot symbols – which are mainly suitable for an initial estimate of the channel.
(ii) Scattered pilot tones whichhelp to track changes in channels over time.
(iii) Eligen value decomposition – based methods – which can be used to reduce the
complexity of the first two methods.

Major problems of OFDM:

The major problems of OFDM is

(i) Peak-to-average power ratio (PAR)

(ii) Intercarrier interference [ICI]
(iii) Inter symbol interface[ISI]

Peak-to-average power ratio(PAR):

The peak amplitude of the emitted signal can be considerably higher than the average
amplitude. PAR occurs in OFDM signal is the superposition of N sinusoidal signals on different
subcarriers.

Due to PAR, non-linear distortions and decreased spectral efficiency is OFDM system. So, we
use PAR reduction techniques, these are:

(a) Coding for PAR reduction

(b) Phase adjustments
(c) Correction by multiplicative function
(d) Correction by additive function

Intercarrier interference:

The cyclic prefix provides an excellent way of ensuring orthogonality of the carriers in delay
dispersive environment. ICI can be eliminated of OFDM.

Inter symbol interference:

 In a digital communication system, the received waveform spreads into neighboring symbols
and produces inter symbol interference, this results in irreducible errors that are caused in the detected
signal. ISI occurs modulation pulses are spread in time into adjacent symbols.

PEAK-TO-AVERAGE POWER RATIO [PAPR]:

ORIGIN OF THE PEAK-TO-AVERAGE RATIO PROBLEM:

One of the major problems of OFDM is that the peak amplitude of the emitted signal can be
considerably higher than the average amplitude. This issue originates from the fact that an OFDM
signal is the superposition of N sinusoidal signals on different subcarriers.

On Average the emitted power is linearly proportional to N. sometimes the signal on the
subcarriers add up constructively so that the amplitude of the signal is proportional to N and the power
thus goes with N2.

The contributions to the total signal from the different subcarriers can be viewed as random
variables (they have quasi random phases, depending on the sampling time as well as the value of the
symbol with which they are modulated).

If the number of subcarriers is large the central limit theorem to show that the distribution of
the amplitudes of in-phase components is Gaussian, with a standard deviation 𝜍 = 1 2 (and
similarly for the quadrature components) such that mean power is unity. Since both in-phase and
quadrature components are Gaussian, the absolute amplitude is Rayleigh-distributed.

Knowing the amplitude distributions it is easy to compute the probability that the
instantaneous amplitude will be above a given threshold and similarly for power is 6dB above the
average power.

PEAK-TO-AVERAGE – RATIO REDUCTION TECHNIQUES:

Some of the promising approaches are:

Coding for PAR reduction

Phase adjustments

Correction by multiplicative function

Correction by additive function

1. CODING FOR PAR REDUCTION:

These code words only a subset of size 2K is acceptable in the sense that its PAR is lower than a
given threshold. Both the transmitter and the receiver know the mapping between a bit combination
of length K and the codeword of length N that is chosen to represent it which has a admissible
PAR.

The transmission scheme is thus the following:

 Parse the incoming bitstream into blocks of length K.

Select the associated codeword of length N.

Transmit this codeword via the OFDM modulator.

The coding scheme can guarantee a certain value for the PAR. It also has some coding gain is
smaller than for code that are solely dedicated to error correction.

2. PHASE ADJUSTMENTS:
This scheme first defines an ensemble of phase adjustment vectors ∅ l, l= 1,….,L, that are
known to both the transmitter and receiver; each vector has N entire {∅}. The transmitter than
multiplies the OFDM symbol to be transmitted cn by each of these phase vectors to get:
𝑐𝑛 l = cnexp[j (∅n)l]
and then selects:
𝑙 = argmin (PAR ( 𝑐𝑛 l))

The vector which gives the lowest PAR. 𝑐𝑛 𝑙 is then transmitted, together with index 𝑙 .The
receiver can then undoes phase adjustment and demodulate the OFDM symbol.

This method has the advantage that the overhead is rather small.

3. CORRECTION BY MULTIPLICATIVE FUNCTION:

Another approach is to multiply the OFDM signal by a time-dependent function whenever the
peak is very high. The simplest example for such an approach is the clipping we mentioned in
the previous subsection: if the signal attains a level Sk>A0 ist is multiplied by a factor A0/Sk.
In other words; the transmit signal becomes:
𝑆𝑘 −𝐴0
𝑠 𝑡 = S(t) 1 − 𝑛 𝑚𝑎𝑥 0, 𝑆𝑘

4. CORRECTION BY ADDITIVE FUNCTION:

In a similar sprit we can an additive instead of multiplicative correction function.

The correction function should be smooth enough nit to introduce significant out-of-band
interference. Furthermore the correction function acts as additional pseudo-noise and thus
increases the BER of the system.

When comparing the different approaches to PAR reduction we find that there is no single
“best” technique. The coding method can guarantee a maximum PAR value but requires
considerable overhead and thus reduced throughput.

The phase adjustment method has a smaller overhead (depending on the number of phase
adjustment vectors), but cannot give a guaranteed performance. Neither of these two methods
leads to an increase in either ICI or out-of-band emissions.
 WINDOWING

Introduction
The digital signals are infinite in nature and sufficiently large that the dataset cannot be
manipulated as a whole. Such a large set of signals are difficult to analyze statistically, because
statistical calculation require all points to be available for analysis. In order to avoid these
problems, typically to analyze small subset of the total data, through a process called
windowing.

In communication windowing function is a mathematical function that is zero – valued

outside of some chosen interval and is the process of taking and is the process taking a small
subset of large dataset for processing and analysis.

The nonlinear distortion of the OFDM signal significantly increases the level of the out
of band radiation. The OFDM signals consist of number of unfiltered sub-carriers. Therefore
the out-of band spectrum decreases rather slowly with the speed depending on the number of
subcarriers.

UNIT IV

MULTIPATH MITIGATION TECHNIQUES

Introduction:

In wireless communication systems requires signal processing technique to improve the link performance in
radio environment. The multipath propagation and Doppler spread have a negative impact on the bit error
rate of any modulation technique. Equalization, diversity and channel coding are three techniques to improve
received signal quality.

EQUALIZATION

Introduction

Equalization compensates for intersymbol interface (ISI) created by multipath within time dispersive channels.
If the modulation bandwidth exceeds the coherence bandwidth of the radio channel, ISI occurs and
modulation pulses are spread in time into adjacent symbols. An equalizer within a receiver compensates for
the average range of expected channel amplitude and delay characteristics. Equalizers must be adaptive since
the channel is generally unknown and time varying.

Intersymbol interface (ISI) caused by multipath in band limited (frequency selectively) time dispersive
channels distorts the transmitted signal, causing bit errors at the receiver. ISI has been recognized as the
major obstacle to high speed data transmission over wireless channels. Equalization is a technique used to
fight against intersymbol interference.

ADAPTIVE EQUALIZER

In radio channels, a variety of adaptive equalizers can be used to cancel interference while providing
diversity. The mobile fading channel is random and time varying, equalizers must track the time varying
characteristics of the mobile channel, and thus are called adaptive equalizers.

The general operating models of an adaptive equalizer include:

 (i) Training and

(ii) Tracking.

A known, fixed-length training sequence is send by the transmitter so that the receiver’s equalizer may adapt
to a proper sending for minimum bit error rate (BER) detection.

The training sequence is typically a pseudorandom binary signal is a fixed, prescribed bit pattern. Immediately
following this training sequence, the user data (which may or may not include coding bits) is sent, and the
adaptive equalizer at the receiver utilizes a recursive algorithm to evaluate the channel and estimate filter
coefficients to compensate for the distortion created by multipath in the channel.

Tracking

The training sequence is designed to permit an equalizer at the receiver to acquire the proper filter
coefficients in the worst possible channel conditions. As user data are received, the adaptive algorithm of the
equalizer tracks the changing channel. As a consequence, the adaptive equalizer is continually changing its
filter characteristics overtime. When an equalizer has been properly trained, it is said to have converged.

The time span over which an equalizer converges is a function of the equalizer algorithm, the
equalizer structure, and the time rate of change of the multipath radio channel. Equalizers require periodic
retraining in order to maintain effective ISI cancellation.

BLOCK DIAGRAM OF ADAPTIVE EQUALIZER

An equalizer is usually implemented at baseband or at IF in a receiver. The block diagram of a

communication system with an adaptive equalizer in the receiver is shown.

If x(t) is the original information signal, and f(t) is the combined complex baseband impulse response of the
transmitter, channel, and the RF/IF sections of the receiver, the signal received by the equalizer may be
expressed as

y(t) = x(t) x f* (t) + nb(t)

where f*(t) denotes the complex conjugate of f(t),

nb(t) is the baseband noise at the input of the equalizer, and

x denotes the convolution operation.

F(t)=combined impulse response of transmitter, multipath

radio channel, and receiver RF/IF

Original baseband
message x(t) Radio channel
Modulator Transmitter
 BLOCK DIAGRAM OF ADAPTIVE EQUALIZER

If the impulse response of the equalizer is heq(t), then the output of the equalizer is

d(t) = x(t) f*(t) heq(t) + nb(t) heq(t)

= x(t) g(t) + nb(t) heq(t)

If the channel is frequency selective, the equalizer enhances the frequency components with small amplitudes
and attenuates the strong frequencies in the received frequency spectrum in order to provide a flat,
composite, received frequency response and linear phase response.

For a time-varying channel, an adaptive equalizer is designed to track the channel variations is approximately
satisfied.

Classification of Equalization Techniques

Equalization techniques can be subdivided into two general categories linear and nonlinear
equalization. These categories are determined from how the output of an adaptive equalizer is used for
subsequent control (feedback) of the equalizer. In analog signal d (t) is processed by the decision making
device in the receiver. The decision maker determines the value of the digital data bit being received and
applies a slicing or thresholding operation (a non-linear operation) in order to determine the value of d (t).

Linear Equalizer
 If d(t) is not used in the feedback path to adapt the equalizer, the equalization is linear.

Non - Linear Equalization

If d(t) is fed back to change the subsequent outputs of the equalizer. Many filter structures are used to
implement linear and nonlinear equalizers.

CLASSIFICAATION OF EQUALIZER:

EQUALIZER

Nonlinear
Linear

ML symbol
DFE MLSE
Detector

Transversal Lattice Transversal

Transversal Lattice
channel Est

ero forcing Gradient RLS LMS Gradient RLS LMS

LMSB RLS RLS

RLS Fast RLS Fast RLS

Fast RLS Square Root RLS Square Root RLS

Linear Equalizers:

Linear equalizer can be implemented as an FIR filter, otherwise known as the traversal filter. This type
of equalizer is the simplest type available.

In this equalizer, the current and past values of the received signal are linearly weighted by the filter
coefficient and summed to produce the output. If the delays and the tap gains are analog, the continuous
output of the equalizer is sampled at the symbol rate and the samples are applied to the decision device.

The most common equalizer structure is a linear traversal equalizer (LTE). A linear transversal filter is made up
of tapped lines, with the tapings spaced a symbol period (Ts) apart.

Assuming that the delay elements have unity gain and delay Ts, the transfer function of a linear traversal
equalizer can be written as a function of the delay operator exp(-jωTs) or Z-1.

.
Delay elements

Ts Ts Ts Ts
Y(t)+nb(t)

ClockZ

Taps

BASIC LINEAR TRANSVERSAL EQUALIZER STRUCTURE:

LINEAR TRANSVERSAL EQUALIZER

This simplest LTE uses only feed forward taps, and the transfer function of the equalizer filter is a polynomial
in Z-1. This filter has many zeroes but poles only at z=0, and is called a finite impulse response (FIR) filter, or
simply a traversal filter.

Yk+N1-1 Yk Yk+N2
Yk+N1 Yk-N2
z-1 z-1 z-1 +1 z-1
If the equalizer has both feed forward and feedback taps, its transfer function is a rational function of Z -1, and
is called an infinite impulse response (IIR) filter with poles and zeroes
𝑁2
𝑑k = 𝑛=−𝑁1 (𝑐𝑛 ∗)𝑦𝑘−𝑛

Where cn* represents the complex filter coefficients or tap weights,

𝑑 k is the output at time index k,

yi is the output received signal at time t0 + iT,

t0 is the equalizer starting time, and

N=N1 + N2 + 1 is the number of taps.

The values N1 and N2 denote the number of taps used in the forward and reverse portions of the equalizer,
respectively. The minimum mean squared error E[|e(n)|2] that a linear transversal equalizer can be achieved .
𝜋
𝑇 𝑁0
E [|e (n) |2] = 2𝜋 𝑇
−𝜋 𝑖𝜔𝑇 )|2 +𝑁 0
𝑑𝜔
𝑇
|𝐹(𝑒

Where 𝐹(𝑒 𝑖𝜔𝑇 ) is the frequency response of the channel, and N0 is the noise power spectral density.

Lattice Equalizer

In lattice structure the input signal Yk is transformed into set of N intermediate forward fn(k) and backward
error signals nn(k). Those two signal are used as input to the tap multiplier and are used to calculate the
updated co-efficient.

At each stage of the lattice is characterized by the recursive equations:

𝑓1 𝑘 = 𝑏1 𝑘 = 𝑦 𝑘
𝑛

𝑓𝑛 𝑘 = 𝑦 𝑘 − 𝐾𝑖 𝑦 𝑘 − 𝑖 = 𝑓𝑛−1 𝑘 + 𝐾𝑛−1 𝑘 𝑏𝑛−1 (𝑘 − 1)

𝑖=1

bn k = y k − n − k i y(k − n + 1)
i=1
 = 𝑏𝑛−1 𝑘 − 1 + 𝐾𝑛−1 𝑘 𝑓𝑛−1 (𝑘)

where Kn(k) is the reflection coefficient for the nth stage of the structure

The backward error signal bn are then used as input to the tap weight

The output of the equalizer is given by

𝑁

𝑑= 𝑐𝑛 𝑘 𝑏𝑛 𝑘
𝑛=1

Advantages of lattice equalizer

(i) Numerically stable

(ii) Faster convergence

(iii) Unique structure allows the dynamic assignment of the most efficient

length.

(iv) If the channel is not time dispersive, only a fraction of stages are used.

Disadvantages of lattice equalizer

(i) Structure is more complicated

(ii) For severe distortion channel, lattice equalizer is not preferable one.

f1(k) f2(k) fN(k)

Σ Σ

Y K1 KN-1

b2(k)
bN

Z-1 Σ Z-1 Σ
b1(k-1)
b1(k)
C1
C2 CN

𝑑1
NONLINEAR EQUALIZATION:

Nonlinear equalizers are used in applications where the channel distortion is too server for a linear
equalizer to handle, and are commonplace in practical wireless systems.

Linear equalizers do not perform well on channels which have deep spectral null in the passband. Three very
effective nonlinear methods have been developed which offer improvements over linear equalization
techniques and are used in most 2G and 3G system. They are

1. Decision Feedback Equalization (DFE)

2. Maximum Likelihood Symbol Detection
3. Maximum Likelihood Sequence Estimation (MLSE)

DECISION FEEDBACK EQUALIZATION (DFF):

The basic idea behind decision feedback equalization is that once an information symbol has been detected
and decides upon, the ISI that it includes on future symbols can be estimated and subtracted out before
detection of subsequent symbols.

The DFE can be realized in either the direct transversal from or as a lattice filter. The direct from is shown in
figure. It consists of a feed forward filter (FFF) and a feedback filter (FBF). The FBF is driven by decision on the
output of the detector, and its Coefficients can be adjusted to cancel the SIS on the current symbol from past
detected symbols.

The equalizer has N1+N2+1 taps in the feed forward filter and N3 taps in the feedback filter, and its
output can be expressed as:
𝑁2 ∗ 𝑁3
𝑑k = 𝑛 =−𝑁1 𝐶𝑛 𝑌𝑘−𝑛 + 𝑖=1 𝐹𝑖 𝑑𝑘−𝑖

Where 𝐶𝑛∗ and 𝑌𝑛∗ are taps gain and the inputs, respectively, to the forward filter,

𝐹𝑛∗ are tap gains for the feedback filter, and

di(i<K) is the previous decision made on the detected signal.

 Yk+N1-1 Yk Yk+N2+1
Yk+N1 Yk-N2
z-1
z-1 z-1 z-1

c-N1 c-N1+1 C0 CN2-1 CN2

d^k
dk
+ + + +

Feed Forward Filter (FFF)

dk-1
dk-N3 z-1 z-1

FN3 FN3-1 F1

+ +

Feed Back Filter (FBF)

DECISION FEEDBACK EQUALIZATION (DFF):

𝑑k is obtained using Equation. Then, dk along with previous decisions dk-1, dk-2.....are feedback into the
equalizer, and 𝑑k+1is obtained using Equation. The minimum mean squared error a DFE can achieved

𝑇 𝜋 𝑇 𝑁
E[|e(n)|2]min = exp {2𝜋 −𝜋 𝑇
𝑖𝑛 [|𝐹(𝑒 𝑖𝜔𝑇 0)|2 +𝑁0] 𝑑𝜔}

It can be shown that the minimum MSE for a DFE in Equation is always smaller than that of an LTE in
Equation unless |𝐹(𝑒 𝑖𝜔𝑇 )| is a constant, a DFE has significantly smaller minimum MSE than an LTE.
 Thus, a DFE is more appropriate for severely distorted wireless channels. The lattice implementation
of the DFE is equivalent to a transversal DFE having a feed forward filter of length N1 and a feedback filter of
length N2, where N1>N2.

PREDICTIVE DFE

Another form of DFE is called a predictive DFE. It also consists of a feed forward filter (FFF) as in the
conventional DFE.

However, the feedback filter (FBF) is driven by an input sequence formed by the difference of the
output of the detector and the output of the feed forward filter.

Hence, the FBF here is called a noise predictor because it predicts the noise and the residual ISI
contained in the signal at the FFF output and subtracts from it the detector output after some feedback delay.

The predictive DFE performs as well as the conventional DFE as the limit in the number of taps in the
FFF and the FBF approach infinity. The FBF in the predictive DFE can also be realized as a lattice structure. The
RLS lattice algorithm can be used in this case to yield fast convergence.

MAXIMUM LIKELIHOOD SEQUENCE ESTIMATION (MLSE) EQUALIZER:

The MSE based linear equalizers are optimum with respect to the criterion of minimum probability of
symbol error when the channel does not introduce any amplitude distortion.

These equalizers use various forms of the classical maximum likelihood receiver structure. Using a channel
impulse response simulator within the algorithm, the MLSE tests all possible data sequence (rather than
decoding each received symbol by itself), and choose the data sequence with the maximum probability as the
output. An MLSE as national requirement, especially when the delay spread of the channel is large.

The MLSE can be viewed as a problem in estimating the state of a discrete-time finite state machine,
which in this case happens to the radio channels with coefficients fk and with a channel state which at any
instant of time is estimated by the receiver based on the L most recent input samples. The block diagram of a
MLSE receiver based on the DFE is shown in.

Requirements of MLSE

The MLSE is optimal in the sense that it minimizes the probability of a sequence error. The MLSE
requires knowledge of the channel characteristics in order to compute the metrics for making decisions. The
MLSE also requires knowledge of the statistical distribution of the noise computing the signal.

Thus, the probability distribution of the noise determines the form of the metric for optimum
demodulation of the received signal. Notice that the matched filter operates on the continuous time signal,
whereas the MLSE and channel estimator relay on discretized (nonlinear) sample.

Received Signal
Sample +
Feed Back Ʃ Decision Device
Filter
Output
Decision
-
 THE STRUCTURE OF A MAXIMUM LIKELIHOOD SEQUENCE

ESTIMATOR (MLSE) WITH AN ADAPTIVE MATCHED FILTER:

Channel output
Estimated data
Z(t) {zn}
y(t) sequence
Matched Filter MLSE
{an}

{si}
Delay

+
Channel
+
Estimator
-

ALGORITHMS FOR ADAPTIVE EQUALIZATION:

Since an adaptive compensates for an unknown and time-varying channel, it requires a specific
algorithm to update the equalizer coefficients and track the channel variations.
 A wide range of algorithm exists to adapt the filter coefficients. The development of adaptive algorithms is a
complex undertaking, and it is beyond the scope of this text to delve into great detail on how this is done. The
performance of an algorithm is determined by various factors which include:

Rate of convergence - This is defined as the number of iterations required for the algorithm, in
response to stationary inputs, to converge close enough to the optimum solutions. A fast rate of convergence
allows the algorithm to adapt rapidly to a stationary environment of unknown statistic. Furthermore, it
enables the algorithm to track statistical variations when operating in a non-stationary environment.

Misadjustment - this parameter provides a quantitative measure of the amount by which the final
value of the mean square error, average over an ensemble of adaptive filters, deviates from the optimal
minimum mean square error.

Computational complexity – This is the number of operations required to make one complete iteration
of the algorithm.

Numerical properties – When an algorithm is implemented numerically, inaccuracies are produced

due to round-off noise and representation error in the computer. These kinds of errors influence the stability
of the algorithm.

In practice, the cost of the computing platform, the power budget, and the radio propagation
characteristics dominate the choice of an equalizer structure and its algorithm.

The choice of algorithm, and is corresponding rate of convergence, depends on the channel data and
coherence time.

The maximum excepted time delay spread of the channel dictates the number of taps to the equalizer
design. An equalizer can only equalize over delay intervals less than or equal to the maximum delay within the
filter structure.

Three classic equalizer algorithms are discussed below. These includes the 1. Zero forcing (ZF)
algorithm,

2. The least mean squares (LMS) algorithm, and

3. The recursive least squares (RLS) algorithm.

ZERO FORCING ALGORITHMS:

In a zero forcing equalizer, the equalizer coefficients Cn are chosen to force the samples of the
combined channel and equalizer impulse response to zero at all but one of the NT spaced sample points in the
tapped delay line filter.

By letting the number of coefficients increase without bound, an infinite length equalizer with zero ISI at the
output can be obtained.

When each of the delay elements provide a time delay equal to the symbol duration T, the frequency
response Heq(f) of the equalizer is periodic with a period equal to the symbol rate 1/T. the combined response
of the channel with the equalizer must stratify NY Quist’s first criterion.

Hch(f)Heq(f) = 1, |f| <1/2T

Where Hch(f) is the folded frequency response of the channel.

ISI equalizer is simply an inverse filter which inverts the folded frequency response of the channel. Thus
infinite length equalizer is usually implemented by a truncate length version.

Disadvantages

1. The zero forcing equalizer has the disadvantages that the inverse filter may excessively amplify noise at
frequencies where the folded channel spectrum has high attenuation.

2. The ZF equalizer thus neglects the effect of noise altogether, and os not often used for wireless links.
However, it performs well foe static channel with high SNR such as local wired telephone lines.

LEAST MEAN SQUARE ALGORITHM:

A more robust equalizer is the LMS equalizer where the criterion used is the minimization of the mean
square error (MSE) between the desired equalizer output and the actual equalizer output. Using the notation
developed in algorithm can be readily understood.

The prediction error is given by

ek =dk - 𝑑k= xk - 𝑑k

And from Equation

ek = xk -𝑌𝑘𝑇 𝑤𝑘 = xk -𝑤𝑘𝑇 𝑦𝑘

To compute the mean square error |ek|2 at time instant k, Equation is squared to obtain

ξ = E*𝑒𝑘∗ ek]

The LMS algorithm seeks to minimize the mean square error given in Equation. For a specific channel
condition, the prediction error ek is dependent on the tap gain vector wN,

so the MSE of an equalizer is a function of wv. Let the cost function J(wN) denote the mean square
error as a function of tap gain vector wN in order to minimize the MSE zero.

𝜕
J( wN) = -2𝑝𝑁 + 2RNN𝑊𝑁 =0
𝜕𝑊𝑁

Simplifying Equation

2RNN𝑊𝑁 = 𝑝𝑁

Since the error is minimized and is made orthogonal to the projection related to the desired signal Xk.
when Equation is satisfied the MMSE of the equalizer is

Jopt= J(𝑊𝑁 ) = E[Xk𝑋𝑘∗ ] - 𝑃𝑁𝑇 𝑊𝑁

To obtain the optimal tap gain vector 𝑊𝑁 , the normal equation in must be solved interactively as the
equalizer converges to an acceptably small value of Jopt. There are several ways to do this and many variants of
the LMS algorithm have been built upon the solution of equation. One obvious technique is to calculate.

𝑊 = R-1NN𝑝𝑁
 However inverting a matrix requires O(N3) arithmetic operations.

Advantages

The advantages of these methods which directly solve Equation is that only N symbol inputs are required to
solve the normal equation. Consequently a long training sequence is not necessary.

The LMS algorithm is the simplest equalization algorithm and required only 2N+1 operation per
iteration.

The filter weights are updated by the equations given below. Letting the variable n denote the sequence of
iterations, LMS is compute iteratively by

𝑑k (n) = 𝑤𝑁𝑇 (n)YN(n)

ek(n) = Xk(n) -𝑑k (n)

WN (n+1) = WN (n) - α𝑒𝑘∗ (n)YN(n)

Where the subscript N denotes the number of delay stages in the equalizer and α is the step size
which controls the convergence rate and stability of the algorithm.

The LMS equalizer maximize the signal to distortion ratio its output within the constraints of the
equalizer filter length. If an input signal has a time dispersion characteristic that is greater than the
propagation delay through the equalizer then the equalizer will be unable to reduce distortion.

Disadvantage

The convergence rate of the LMS algorithm is slow due to the fact that the there is only one parameter the
step α that controls the adaptation rate. To prevent the adaptation from becoming unstable the value of α is
chosen from
𝑁
0< α<2/ 𝑖=1 𝜆𝑖

Where 𝜆𝑖 is the ith eigenvalue of the covariance matrix RNN. Since 𝑁 𝑇

𝑖=1 𝜆𝑖 =𝑌𝑁 (n)YN(n), the step size α
can be controlled by the total input power in order to avoid instability in the equalizer.

RECURSIVE LEAST SQUARES ALGORITHM:

The convergence rate of the gradient-based LMS algorithm is very slow, especially when the eigen
values of the input covariance matrix RNN have a very large speed, 𝜆 max/λmin>>1.

In order to achieve faster convergence, complex algorithms which involve additional parameters are used.
Faster converging algorithms are based on a least squares approach, as opposed to the statistical approach
used in the LMS algorithm.

That is rapid convergence relies on error measures expressed In terms of a time average of the actual received
signal instead of a statistical average. This leads to the family of powerful, albeit complex adaptive signal
processing techniques known as recursive least squares (RLS), which significantly the convergence of adaptive
equalizer.

The least square error based on the time average is defined as

 𝑛 𝑛−𝑖 ∗
J(n) = 𝑖=1 𝜆 𝑒 (i, n)e(i, n)

Where 𝜆 is the weighting factor close to 1, but smaller than 1, e*(i, n) is the complex conjugate of e(i,
n), and the error e(i, n) is

e(i, n) = x(i) -𝑌𝑁𝑇 𝑖 WN(n) 0≤i≤n

And

YN(i) = [Y(i), y(i-1),…….y(i-N+1)]T

Where YN(i) is the data input vector at time I, and WN(n) is the new tap gain vector at time n.

e(i, n) is the error using the new tap gain at time n to test the old data at time i, and

J(n) is the cumulative squared error of the new tap gains on all the old data.

To obtain the minimum of least square error J(n) the gradient of J(n) in equation is set to zero

𝜕
𝜕𝑊𝑁
J(n) =0

Using equations it can be shown that

RNN(n)𝑤𝑁 (n) = PN(n)

Where 𝑤𝑁 is the optimal tap gain vector of the RLS equalizer,

𝑛
RNN(n) = 𝑖=1 𝜆
𝑛−1
𝑦𝑁∗ (i)𝑦𝑁𝑇 (i)
𝑛 𝑛−1 *
PN(n) = 𝑖=1 𝜆 x (i)yN(i)

The matrix RNN(n) in equation is the deterministic correlation matrix of input data of the equalizer y N(i)
and PN(i) in equation is the deterministic cross-correlation vector between inputs of the equalizer yN(i) and the
desired output d(i), where d(i) = x(i). To compute the equalizer weight vector 𝑤𝑁 using equation it is required
to compute R-1NN(n).

From the definition of RNN(n) in equation it is possible to obtain a recursive equation expressing RNN(n)
in terms of RNN(n-1)

RNN(n) = λ RNN(n-1)+yN(n)𝑦𝑁𝑇 (n)

Since the three terms in equation are all N by N matrices a matrix inverse lemma can be used to
𝑇−1 𝑇−1
derive a recursive update for 𝑅𝑁𝑁 in terms of the previous inverse 𝑅𝑁𝑁 (n-1)
𝑇−1 𝑇 𝑇−1
𝑇−1 1𝑇−1 𝑅𝑁𝑁 (n−1)yN(n)𝑦𝑁 (n)𝑅𝑁𝑁 (n−1)
𝑅𝑁𝑁 (𝑛) = 𝜆 [𝑅𝑁𝑁 (n-1) - 𝜆+𝜇 (𝑛)

Where

𝜇(𝑛) = 𝑦𝑁𝑇 (n) R-1NN(n) yN(n)

Based on these recursive equations the RLS minimization leads to the following weight update
equations:
 WN(n) = wN(n-1) + KN(n)e*(n,n-1)

Where
𝑇−1
𝑅𝑁𝑁 (n−1)yN(n)
KN(n) = 𝜆+𝜇 (𝑛)

The RLS algorithm may be summarized as follows:

1. Initialize w(0) = K(0) = x(0) = 0, R-1(0) = 𝛿𝐼 nn, where INN is an N*N identity matrix, and 𝛿 is a large
positive constant.

2. Recursively computes the following:

𝑑(n) = wT(n-1)y(n)

e(n) = x(n)-𝑑(n)

𝑅 −1 𝑛−1 𝑦(𝑛)
k(n) = 𝜆+𝑌 𝑇 𝑛 𝑅 −1 𝑛−1 𝑦(𝑛)

1
𝑅 −1 (n) = [𝑅 −1 𝑛 − 1 -k(n)yT(n)𝑅 −1 𝑛 − 1 ]
𝜆
w(n) =w(n-1) +k(n)e*(n)
In equation 𝜆 is the weighting coefficient that can change the performance of the equalizer. If a
channel is time-invariant, 𝜆 can be set to one. The value of 𝜆 has no influence on the rate of convergence, but
does determines the tracking ability of the RLS equalizers. The smaller the value of 𝜆 then better the tracking
ability of the equalizer. However, if 𝜆 is too small the equalizer will be unstable.

DIVERSITY TECHNIQUES:
Introduction
Diversity is a powerful communication receiver technique that provides wireless link improvement at
relatively low cost. Unlink equalizations, diversity requires no training overhead since a training sequence is
not required by the transmitter.
Diversity is a method used to develop information from several signal transmitted over independent
fading paths. It exploits the random nature of radio propagation by finding independent signal paths for
communications.
In such a system, the receiver is provided with multiple copies of the same information signal which
are transmitted over two or more real or virtual communication channels. Thus the basic idea of diversity
is repetition or redundancy of information. In virtually all the applications, the diversity decisions are made
by the receiver and are unknown to the transmitter.
It is a very simple concept where if one path undergoes to a deep fade, another independent path
may have a strong signal. As there is more than one path to select from both the instantaneous and average
SNRs at the receiver may be improved.
Consider the simple case of an Rx with two antennas. The antennas are assumed to be far enough
from each other that small scale fading is independent at the two antennas. The Rx always chooses the
antenna that has instantaneously larger receive power.

Classification of fading
 There are two types of fading-small-scale and large-scale fading.
(i) Small-scale faders are characterized by deep and rapid amplitude fluctuations which occur as the
mobile moves over distances of just a few wavelengths. These faders are caused by multiple reflections from
the surrounding in the vicinity of the mobile.
In order to prevent deep fades from occurring, microscopic diversity techniques can exploit the rapidly
changing signal. For example the small-scale fading shown revels that if two antennas are separated by a
fraction of a meter, one may receive a null while the other receivers a strong signal. By selecting the best
signal at all times a receiver can mitigate small-scale fading effects.
(ii) Large-scale fading is caused by shadowing due to variations in both the terrain profile and the
nature of the surrounding. In deeply shadowed conditions the received signal strength at a mobile can drop
well below that of free space.By selecting a base station which is not shadowed when others are, the mobile
can improve substantially the average signal-to-noise ratio on the forward link. This is called macroscopic
diversity, since the mobile is taking advantage of large separations between the serving base stations.
Macroscopic diversity is also useful at the base station receiver. By using base station antennas that
are sufficiently separated in space the base station is able to improve the reverse link by selecting the antenna
with the strongest signal from the mobile.

MICRO DIVERSITY
The basic principle of diversity is that the Rx has multiple copies of the transmit signal, where each of
the copies goes through a statistically independent channel. For narrowband signals, small scale fading results
in a Rayleigh fading. In order to prevent deep fades from occurring microscopic technique can be used. The
diversity method that fight against small scale fading i.e. the fading created by interference of MPCs is called
as micro diversity.
The most five methods are:
1. Temporal diversity: Transmission of the transmit signal at different times
2. Spatial diversity: Several antenna element separated in space
3. Frequency diversity: Transmission of the signal on different frequencies
4. Angular diversity: Multiple antennas with different antenna pattern
5. Polarization diversity: Multiple antennas with different polarization.

TEMPORAL DIVERSITY
Principle
In the wireless propagation channel is time variant, signal that are received at different times are
𝟏
uncorrelated. For sufficient de-correlation, the temporal distance must be at least 𝟐𝛄 .
𝐦𝐚𝐱
where 𝛄𝐦𝐚𝐱 is the maximum Doppler frequency.
In a static channel, where neither transmitted (Tx), Rx, nor the IOs are moving, the channel state is the
same at all times. Temporal diversity cal be realized by the following ways:
1. Repetition coding
2. Automatic repeat request (ARQ)
3. Combining of interleaving and coding
1. REPETITION CODING
The signal is repeated several times, where the repetition intervals are long enough to achieve de-
correlation. This obviously achieves diversity, but is also highly bandwidth inefficient. Spectral efficiency
decreases by a factor that is equal to the number of repetitions.
Disadvantages
Retransmission occurs always
2. AUTOMATIC REPEAT REQUEST (ARQ)
The Rx sends a message to the Tx to ensure whether it received the data with sufficient quantity. If
the data is not transmitted successfully, then the transmission is repeated.
Advantages
1. The spectral efficiency of ARQ is better than that of repetition coding.
2. It requires multiple transmissions only when the first transmission occurs
in a bad fading state.
Disadvantages
1. It requires feedback channel.

3. COMBINING OF INTERLEAVING AND CODING

A more advanced version of repetition coding is forward error correlation coding with interleaving.
The different symbols of a codeword are transmitted at different time which increases the probability that at
least some of them arrive with a good SNR. The transmitted codeword can be reconstructed.

SPATIAL DIVERSITY (or) ANTENNA DIVERSITY

The basic principle of this type of diversity is selecting the best signal among all the signals received
from different branches at the received end.
A method of transmission or reception or both in which the effects of fading are minimized by the
simultaneous use of two or more physically separated antennas, ideally separated by one half or more
wavelengths.
Signal received from spatially separated antennas have uncorrelated envelope. This concept is also
used in base station.

Spatial Diversity

Selection based diversity Combining diversity

Maximal Ratio Combining

Selection diversity
( MRC)

Switched diversity
Equal gain Diversity (EGD)
Feedback diversity

Space diversity reception methods can be classified into four categories

1. Selection diversity
2. Switched diversity
3. Feedback diversity

1. SELECTION BASED DIVERSITY

 A block diagram of this method is similar to that shown, where m demodulators are used to provide m
diversity branches whose gains are adjusted to provide the same average SNR for each branch. As derived in
the receiver branch having the highest instantaneous best one sent to a single demodulator. The antenna
signals themselves could be sampled and the best one sent to a single demodulator. In practice, the branch
with the largest (S+N)/N is used since it is difficult to measure SNR alone. A practical selection diversity
systems cannot function on a truly instantaneous basis, but most bedesigned so that internal time constants
of the selection circuitry are shorter than the reciprocal of the signal fading rate.

G1

G2 Switching logic or
Demodulators output

m
antenna
Gm

Variable gain
GENERALIZED BLOCK DIAGRAM FOR SPACE DIVERSITY

(i) Received Signal Strength Indication – RSSI

The basic principle of this type of diversity is selecting the signal with largest instantaneous power
among all the signal received from different branches at the receiving end process.
It consist of Ng demodulator are used to provide Ng diversity branches whose gains are adjusted to
provide the same average SNR for each branch and Ng-1 are used to switch channel which contains signal with
largest instantaneous power.
It requires only one RF chain. The receiver branched having the highest instantaneous SNR is
connected to the demodulator.

Antenna A
Rx filter

RSSI
Antenna B Comparator Demodulator

RSSI
DERIVATION OF SELECTING DIVERSITY IMPROVEMENT:

Consider M independent Rayleigh fading channels available at a receiver. Each channel is called a
diversity branch further assume that each branch has the same average SNR given by
𝐸
SNR = ⎾ = 𝑁𝑏 𝛼 2
0
Where we assume 𝛼 2 =1.
If each branch has an instantaneous SNR = 𝛾𝑖 , then from equation the pdf of 𝛾𝑖 is
−𝛾 𝑖
1
P(𝛾𝑖 ) = ⎾ 𝑒 ⎾ 𝛾𝑖 ≥0
Where ⎾ is the mean SNR of each branch. The probability that a single branch has an instantaneous
SNR less than some threshold 𝛾 is
−𝛾 𝑖 −𝛾 𝑖
𝛾 𝛾 1
pr[𝛾𝑖 ≤ 𝛾] = 0 P(𝛾𝑖 ) d𝛾𝑖 = 0 𝑒 ⎾ d𝛾𝑖 = 1-𝑒 ⎾
⎾
Now the probability that all M independent diversity branches receive signals which are
simultaneously less than some specific SNR threshold 𝛾 is
pr[𝛾𝑖 , …….𝛾𝑀 ≤ 𝛾] = (1- 𝑒 −𝛾 ⎾)M = pM(𝛾)

pM(𝛾)in equation is the probability of all branches failing to achieve instantaneous SNR= 𝛾. If a signal
branch achieves SNR >𝛾 then the probability that SNR >𝛾 for one or more branches is given by
𝑑 𝑀
pr[𝛾𝑖 ≤ 𝛾] = 𝑑𝛾 𝑃𝑀 (𝛾) = ⎾(1- 𝑒 −𝛾 ⎾ )M-1𝑒 −𝛾 ⎾
Then the mean SNR, 𝛾 may be expressed as
∞ ∞
𝛾 = 0 𝛾 𝑝𝑀 (𝛾)d 𝛾 = ⎾ 0 𝑀𝑥 -e-x)M-1 e-xdx
Where x= 𝛾/⎾. Note that ⎾ is the average SNR for a single branch equation is evaluated to yield the
average SNR improvement offered by selection diversity
𝛾 1
⎾
= 𝑀𝑘=1 𝑘
The following example illustrates the advantage that diversity provides.
From equation, it can be seen that the average SNR in the branch which is selected using selection
diversity naturally increases, since it is always guaranteed to be above the specified threshold. Thus selection
diversity offers an average improvement in the link margin without requiring additional transmitter power or
sophisticated receiver circuitry.
(ii) Bit – error –rate –driven diversity
Here we transmit a training sequence i.e. a bit sequence that is known at the Rx. The Rx then
demodulates the signal from each received antenna element and compares it with the transmit signal. The
antenna whose associated signal results in the smallest BER is identified as best signal and used for the
subsequent reception of data signals.
If the channel is time variant, the training sequence has to be repeated at regular intervals and
selection of the best antenna has to done. The necessary repetition rate depends on the coherence time of
the channel.
Disadvantages
1. Rx more complex here it requires N RF chains and demodulators.
2. Spectral efficiency decreases due to training sequence has to be repeated
for every Nr times, so that the signal at all antenna elements can be
evaluated.
3. The variance of BER around its true mean decreases as the duration of the
 training sequence increases.
Antenna A

Rx Filter Demodulator

Correlation
Tx Sequence Comparison Detector
Antenna B
Correlation

2. SWITCHED DIVERSITY
In this the selection criterion of active diversity branch is monitored. If it falls below a certain
threshold, then the Rx switches to a different antenna. Switching only depends on the quality of the active
Rx Filter
diversity branch. Demodulator
It does not matter whether the other branch actually provides a better signal quality or not.
The parameters to be considered are switching threshold and hysteresis time.

3. FEEDBACK DIVERSITY (or) SCANNING DIVERSITY:

Scanning diversity is very similar to selection diversity except that instead of always using the best of
M signals the M signals are scanned in a fixed sequence until one is found to be above a predetermined
threshold. This signal is then received until it falls below threshold and the scanning process is again initiated.
Advantages
This method is very simple to implement – only one receiver is required.

Antenna
Control
Comparator Present
Threshold

Short-Term
Average

Receiver

BASIC FORM OF SCANNING DIVERSITY

Combining diversity

Combining diversity leads to better performance, as all available information is exploited. On the
downside it requires a more complex Rx than selection diversity. In most Rxs all processing is done in the
baseband. Thus the Rx with combining diversity needs to down – convert all available signals and combine
them appropriately in the baseband.

Two main methods:

1. Maximum Ratio Combining (MRC) weighs all signal copies by their amplitude. It can be shown that this is an
optimum combining strategy.

2. Equal gain combining (EGC) where all amplitude weights are the same (or no weighting but just a phase
correction)

1. MAXIMAL RATIO COMBINING

1 γ1

G1
2 γ2
Cophase and Sum Detector
G2

m γm
m
Gm
Antenna

Adaptive control
MAXIMAL RATIO COMBINER

In this method the signal from all of the M branches are weighted according to their individual signal
voltage to noise power rations and then summed. Here, the individual signals must be co-phased before being
summed which generally requires an individual receiver and phasing circuit for each antenna element.
Maximal ratio combining produces an output SNR equal to the sum of the individual SNR as explained.

This signal at the different branches is multiplied with weights Wn*

𝑁
𝑛−1 𝑊𝑛
∗ 𝑎𝑛 2
𝑃𝑛 𝑁
𝑤𝑛 2𝑛−1
Output SNR of the diversity combiner is the sum of the branch SNRs:
𝑁𝑟

𝛾𝑀𝑅𝐶 = 𝛾𝑛
𝑛−1
PDF can be expressed as
1 𝛾 𝑁 𝑟−1 𝛾
𝑝𝑑𝑓 𝛾 = (𝑁𝑟 −1)! 𝛾 𝑁 𝑟
exp⁡
(− 𝛾 )
γ = Nr γ
Modern DSP techniques and digital receivers are now making this optimal form of diversity practical.
Advantages
(i) Thus it has the advantages of producing an output with an acceptable
SNR even when none of the individual signals are themselves
acceptable.
 (ii) This technique gives the best statistical reduction of fading of any known
linear diversity combiner.
Disadvantages
(i) It requires individual receiver and phasing circuits for each antenna elements.
(ii) Needs high implementation cost.

2. EQUAL GAIN COMBAINING:

In certain case it is not convenient to provide for the variable weighting capability required for true
maximal ratio combining. In such case, the branch weights are all set to unity, but the signals from each
branch are co-phased to provide equal gain combining diversity.
This allows the receiver to exploit signals that are simultaneously received on each branch. The
possibility of producing an acceptable signal from a number of unacceptable inputs is still retained and
performance is only marginally inferior to maximal ratio combining and superior to selection diversity.

Antenna A Maximum ratio combining

Phase Correction

Measurement
Output
Σ Demodulator
Antenna B

Measurement

Advantages
(i) EGC is superior to selection diversity.
Disadvantages
(i) EGC performs worse than MRC by factor π/4.
(ii) EGC is inferior to MGC since interference and noise corrupted signals may be combined with high
quality signals.

FREQUENCY DIVERSITY:

Frequency diversity is implemented by transmitting information on one more than one carrier
frequency. The rationale behind this technique is that frequencies separated by more than the coherence
bandwidth of the channel will be uncorrelated and will thus not experience the same.
If the channels are uncorrelated the probability of simultaneous fading will be the product of the
individual fading probabilities.Frequency diversity is often employed in microwave line-of-sight links which
carry several channels in a frequency division multiplex mode (FDM).
Due to tropospheric propagation and resulting refraction, deep fading sometimes occurs. In practice,
1:N protection switching is provided by a ratio license wherein one frequency is nominally idle but is available
on a stand-by basis to provide frequency diversity switching for any one of the N other carriers (frequency)
being used on the same link, each carrying independent traffic. When diversity is needed, the appropriate
traffic is simply switched to the backup frequency.
Advantages
(i) This methods allow the transmission of information without wasting bandwidth.
(ii) Frequency diversity can be exploited by the system to make a more robust and decreases the
effects of fading.
Disadvantages
(i)This technique requires spare bandwidth
(ii) It also requires that there be as many receivers as there are channels used for the frequency
diversity.

PROARIZATION DIVERSITY:

At the base station space diversity is considerable less practical than at the mobile because the
narrow angle of incident fields requires large antennas spacing. The comparatively high cost of using space
diversity at the base station prompts the consideration of using orthogonal polarization to exploit polarization
diversity. While this only provides two diversity branches it does allow the antenna elements to be co-located.
Measured horizontal and vertical polarization paths between a mobile and a base station are reported
to be uncorrelated. The de-correlation for the signals in each polarization is caused by multiple reflections in
the channel between the mobile and base stations antennas.
Circular and linear polarized antennas have been used to characterize multipath inside building. When
the path was obstructed polarizations diversity was found to dramatically reduce the multipath delay spared
without significantly decreasing the received power.
While polarization diversity has been studied in the past, it has primarily been used for fixed ratio links
which vary slowly in time. Line-of-sight microwave links for example typically use polarization diversity to
support two simultaneous users on the same radio channel. Since the channel does not change much in such
a link there is little likelihood of cross polarization interference. As portable users proliferate, polarizations
diversity is likely to become more important for improving link margin and capacity.
Y

V2 V1

X
Multipath β

TIME DIVERSITY: Mobile Main beam

 Time diversity repeatedly transmits information at time spacings that exceed the coherence time of
the channel, so that multiple repetitions of the signal will be received with independent fading conditions
thereby providing for diversity.
One modern implementation of time diversity involves the use of the RAKE receiver for spread
spectrum CDMA, where the multipath channel provides redundancy in the transmitted message. By
demodulating several replicas of the transmitted CDMA signal, where each replica experience a particular
multipath delay the RAKE receiver is able to align the replicas in time so that a better estimate of the original
signal may be formed at the receiver.

MACRO DIVERSITY
The diversity methods that combat large – scale fading i.e. the fading created by shadowing effect is
called as macro diversity.
Shadowing is almost independent of transmit frequency and polarization, so that frequency diversity
or polarization diversity is not effective. If correlation distances for large – scale fading are order of tens or
hundreds of meters, then the spatial diversity can be used.
The simplest method for macrodiversity is the use of on – frequency repeaters that receive the signal
and retransmit an amplified version of it.
Advantages
(i) Uses of on-frequency repeaters that receive the signal and retransmit an amplified version of it.
(ii) In cellular application the two BSs should be synchronized.
(iii) For compensating large signal fading, macro-diversity is used.

RAKE RECEIVER:

In CDMA spread spectrum systems, the chip rate is typically much greater than the flat-fading
bandwidth of the channel. Whereas a conventional modulation technique requires an equalizer to undo the
intersymbol interference between adjacent symbols, CDMA spreading codes are designed to provide very low
correlation between successive chips.
Thus propagation delay spread in the ratio channel merely provides multiple versions of the
transmitted signal at the receiver. If these multipath components are delayed in time by more than chip
duration, they appear like uncorrelated noise at a CDMA receiver and equalizations is not required. The
spread spectrum processing gain makes uncorrelated noise negligible after dispreading.

However since there is useful information in the multipath components, CDMA receivers may
combine the time delayed versions of the original signal transmission in order to improve the signal-to-noise
ratio at the receiver.
A RAKE receiver does just this – it attempts to collect the time-shifted versions of the original signal by
providing a separate correlation receiver for each of the multipath signals. Each correlations receiver may be
adjusted in time delay so that a microprocessor controller can cause different correlation receiver to search in
different time windows for significant multipath.
The range of time delays that a particular correlateor can search is called a window. The RAKE receiver
shown is essentially a diversity receiver designed specifically for CDMA where the diversity is provided by the
fact that the multipath components are practically uncorrelated from one another when their relative
propagation delays exceed a chip period.

Z1
To explore the performance of a RAKE receiver assume M Correlators are used in a CDMA receiver to capture
the M strongest multipath components. A weighting network is used to provide a linear combination of the
Correlator output for bit detection. Correlator 1 is synchronized to the strongest multipath m 1. Multipath
component m2 arrives𝜏1 later than component m1 where 𝜏2 − 𝜏1 is assumed to be greater than a chip
duration. The second Correlator is synchronized to m2. It correlates strongly with m2but has low correlation
with m1. Note that if only a single Correlator is used in the receiver, once the output of the single Correlator is
corrupted by fading, the receiver cannot correct the value. Bit decisions based on only single correlation may
produce a large bit error rate. In a RAKE receiver if the output from one Correlator is corrupted by fading, the
others may not be discounted through the weighting process. Decisions based on the combinations of the M
separate decision statistic offered by the RAKE provide a form of diversity which can overcome fading and
thereby improve CDMA reception.
The M decision statistics are weighted to from an overall decision statistics as shown. The outputs of
the M Correlators are denoted as z1,z2,…..and zm. they are weighted α 1, α 2 ….and α mrespectively. The
weighting coefficients are based on the power or the SNR from each Correlators output. If the power or SNR is
small out of a particular Correlator, it will be assigned a small weighting factor. Just as in the case of a maximal
ratio combining diversity scheme, the overall signal z’ is given by
Z’ = 𝑀 𝑚 =1 α𝑚 𝑍𝑚

The weighting coefficients, α m are normalized to the output signal power of the Correlator in such a
way that the coefficients sum to unity as shown.

2
𝑍𝑚
αm= 𝑀 2
𝑚 =1 𝑍𝑚

As in the case of adaptive equalizer and diversity combining there are many ways to generate the
weighting coefficients. However due to multiple access interference, RAKE fingers with strong multipath
amplitudes will not necessarily provide strong output after correlation. Choosing weighting confidents based
on the actual outputs of the Correlators yields better RAKE performance.

Error probability in fading channels with diversity reception

We determine the symbol error rate (SER) in fading channel when diversity is used at the Rx.
Error probability in flat – fading channels
We can compute the error probability of diversity system by averaging the conditional error
probability over the distribution of the SNR.
∞
SER= 0
𝑝𝑑𝑓𝛾 𝛾 𝑆𝐸𝑅 𝛾 𝑑𝛾

The SER of BPSK in AWGN is

𝑆𝐸𝑅 𝛾 = 𝑄( 2𝛾 )
The analytical equation is
1−b 𝑁𝑟 −1 𝑁𝑟 − 1 + 𝑛 1+𝑏 n
SER = ( ) Nr 𝑛=0 ( )( 2 )
2 𝑛

𝛾
b= 1+𝛾

CHANNEL CODING
Channel coding protects digital data from errors by selectively introduces redundancies in the
transmission data. Channel codes that are used to detect errors are called error detection codes, while codes
that can detect and correct errors are called error corrections codes

BLOCK CODES AND FINITE FIELDS:

Block codes are a forward error correction (FEC) code that enables a limited number of errors to be
detected and corrected without retransmission. Block codes can be used to improve the performance of a
communications system when other means of improvements (such as increasing transmitter power or using a
more sophisticated demodulator) are impractical.
In block codes parity bits are added to block of message bits to make code words or code locks. Ina
blocks encoder, k information bits are encoded into n code bits. A total of n-k redundant bits are added to the
k information bits for the proposes of detecting and correcting errors. The block code is referred to as an (n,k)
code and the rate of the code is defined as Rc = k/n and is equal to the rate of information divided by the raw
channel rate.

SOME TECHNICAL PARAMETERS OF ERROR CONTROL CODING

DISTANCE OF CODE:

The distance between two code words is the number of elements in which two code words Ci and Cj
differ
d(Ci, Cj) = 𝑁
𝑙=1 𝐶𝑖 , l + Cj, l(modulo q)
Where d is the distance between the codes words and q is the total number of possible values of Ci and
Cj. the length of each code word are N elements or characters. If the code used is binary the distance is known
and the hamming distance. The minimum distance dmin is the smallest distance for the given code word set
and is given as
dmin = Min {d(Ci,Cj)}

WEIGHT OF A CODE:
 The weight of a code word of length N is given by the number of nonzero elements in the code word.
For a binary code, the weight is basically the number of ls in the code word and is given as
w(Ci) = 𝑁 𝑙=1 𝐶𝑖
CODE RATE
Consider an encoder that takes k information bits and add q redundant bits for a total of n=k+q bits
per codeword. The code rate is the fraction k/n and the code is called a (n,k) error – control code.

CODE EFFICIENCY
It is the ratio of message bits in a block to the transmitted bits for the block by the encoder.
Message bits in a block
code Efficiency =
Transmitted bits for the block

PROPERTIES OF BLOCK CODES:

LINEARITY- Suppose Ci and Cj are two code words in an (n,k) block code. Let α 1 and α 2 be any two
elements selected from the alphabet. Then the code the is said to be linear if and only if α 1C1 +α 2C2 is also a
code word a linear code must contain the all-zero code word Consequently, a constant-weight code is
nonlinear.

SYSTEMATIC:
A systematic code is one in which the parity bits are appended to the end of the information bits. For
an (n, k) code the first k bits are identical to the information bits and the remaining n-k bits of each code word
are linear combinations of the k information bits.

CYCLIC:
Cyclic codes are a subset of the class of linear codes which satisfy the following cyclic shift property: If
C = [Cn-1, Cn-2,…..C0] is a code word of a cyclic code, then [cn-2,Cn-3,…C0,Cn-1], obtained by a cyclic shift of the
elements of C,is also a codes possess a considerable amount of structure which can be exploited to greatly
simplify the encoding and decoding operations.

EXAMPLES OF BOLCK CODES:

HAMMING CODES:
Hamming codes were among the first of the nontrivial error correction codes. These codes and their
variations have been used for error control in digital communications systems. There are both binary and non-
binary Hamming codes. A binary Hamming code has the property that
(n, k) = (2m – 1, 2m – 1- m)
Where k is the number if information bits used to form a n bit code word and m is any positive integer.
The number of parity symbols are n-k = m.

HADAMARD CODES:
Hadamard codes are obtained by selecting as code words the rows of a Hadamard matrix. A
Hadamard matrix A is a N x N matrix of l s and 0s such that each row differs from any other row in exactly N/2
locations. One row contains all zeros with the remainder containing N/2 zeros and N/2. The minimum
distance for these codes is N/2.
For n=2 , the Hadamard matrix A is
 0 0
A=
0 1
In addition to the special case considered above when N=2m (m being a positive integer), Hadamard
codes of other block lengths are possible, but the codes are not linear.

GOLAY CODES:
Golay codes are linear binary codes with a minimum distance of seven and an error correction
capability of three bits. This is a special one of a kind code in that this is the only nontrivial example of a
perfect code. (Hamming codes and some repetition codes are also perfect). Every code word lies within
distance three of any code word thus making maximum likelihood decoding possible.

CYCLIC CODES:
Cyclic codes are a subset of the class of linear codes which satisfy the cyclic property as discussed
before. As a result of this property these possess a considerable amount of structure which van be exploited.
A cyclic code can be generated by using a generator polynomial g(p) of degree (n-k). The generator
polynomial of an (n, k) cyclic code is factor of pn+1 and has the general form
g(p) = Pn-k+gn-k-1Pn-k-1+…….+g1P+1
A message polynomial x(p) can also be defined as
X(p) = xk-1pk-1+ ……+x1p+x0
Where (xk-1,……x0) represents the k information bits. The resultant code word c(p) can be written as
C(p) = x(p)g(p)
Where c(p) is a polynomial of degree less than n.
Encoding for a cyclic code is usually performed by a linear feedback shift register based on either the
generator or parity polynomial.

BCH CODES:
BCH cyclic codes are among the most important block codes, since they exist for a wide range of rates,
achieve significant coding gains, and can be implemented even at high speeds. The block length of the codes is
n=2m-1 for m ≥3, and the number of errors that they can correct is bounded by t < (2m-1)/2. The binary BCH
codes can be generalized to create classes of non-binary codes which use m bits per code symbol. The most
important and common class of non-binary BCH codes is the family of codes known as Reed-Solomon codes.
The Reed-SOLOMON code in US cellular Digital Packet Data (CDPD) uses m=6 bits per code symbol.

REED_SOLOMON CODES:
Reed-Solomon (RS) are non-binary codes which are capable of correcting errors which appears in
bursts and are commonly used in concatenated coding systems. The block length of these codes is n=2 m-1.
These can be extended to 2m or 2m +1. The number of parity symbols that must be used to correct e errors is
n-k= 2e. the minimum distance dmin = 2e+1. RS codes achieve the largest possible dmin of any linear code.

MULTIPLE ANTENNA TECHNIQUES

MIMO SYSTEM:
 MIMO system consists of several antenna elements plus adaptive signal
processing at both transmitter and receiver.

Multiple Antenna Types

SISO – single input single output means that the transmitter and receiver of the
ratio system have only one antenna

Tx Ry

SIMO – single input multiple outputs means that the receiver has multiple
antennas while the transmitter has one antenna.

Tx Ry

MISO – Multiple input single output means that the transmitter has multiple
antennas while receiver has one antenna.

Tx Ry

MIMO – Multiple inputs multiple outputs means that the both the transmitter
and receiver have multiple antennas.

Tx Ry

TYPES OF MIMO SYSTEM:

 Single user MIMO

 Open loop MIMO

SINGLE USER MIMO:

 When the data rate is to be increased for a single UE, this is called single user
MIMO. h11

1 1
h21
h h12
2
1

2 2
h22

BE UE

MULTIUSER MIMO:

When the individual streams are assigned to various to various users, this is
called multi user MIMO.
h11

1 1
h h21 h
h12
2 2 UE
1 1
2 2
h22 h UE
2
1

BE

OPEN LOOP MIMO VS CLOSE LOOP MIMO:

Commonly used MIMO technology has most often been in reference to open
loop MIMI techniques. Closed loop MIMO techniques also known as Transmitter
Adaptive Antenna (TX-AA) technique. Ω are simply referred to by the industry as
beam foxing.

OPEN LOOP MIMO:

 With open loop MIMO, the communication channel does not utilize explicit
information regarding the propagation channel. Common open loop MIMO techniques
includes space time transmit diversity, special multiplexing (SM) and collaborative
uplink MIMO.

SPACE TIME TRANSMITS DIVERSITY (STTD) MIMO:

STTD is a method of transmit diversity used in UMTS third generation cellular

systems. STTD utilities space time block code in order to explicit redundancy in
multiply transmitted versions of a signal. The same data is coded and transmitted
through different antennas.

SPATIAL MULTIPLEXING MIMO:

Spatial multiplexing is transmission technique in MIMO. Wireless

communication to transmit independent and separately encoded data signals so called
streams, from each of the multiple transmit antennas. Therefore, the space dimension is
Reused, or multiplexed, more than one time. SM delivers parallel streams of data to
CPE by exploiting multipath.

UPLINK COLLABRATIVE MIMO:

Collaborative MIMO is comparable to regular spatial Multiplexing where data

streams are transmitted from multiple antennas on the same device. With this technique
two device can Collaboratively transmit on the same sub-channel which can increase
the uplink capacity

SYSTEM MODEL:

A narrow band point to point communication system employing n transmit and m

receive antennas.

h11

X y1
X2

X2

1
TX X2 Y2 Ry
X2

Xn Ym

hmn
 The transmitter and receiver are equipped with multiple antenna elements. The
transmit stream go through a matrix channel which consists of multiple receive
antennas at the receiver. Then the receiver gets the received s/l vectors by the multiple
receive antenna and decodes the received signal vectors into the original information

Received signal vectors into the original information.

𝑦1 𝑕𝑖1 … … … . . 𝑕𝑖𝑛 𝑥1 𝑁1
. . . .
. = . . + .
. . . .
𝑦𝑚 𝑕𝑚 1 … … … … 𝑕𝑚𝑛 𝑥𝑛 𝑁𝑚

𝑦 = H𝑥 + 𝑁

Where 𝑦 = MX1 receive vector

𝑥 = Represents the NX 1 – dimensional transmitted symbol

H = Channel state

The transmit power constraint is given by

𝑛 ∗
𝑖=1 𝐸[ 𝑥𝑖 𝑋𝑖 ] = p Function of MIMO system:

MIMO includes following functions

 Beam forming
 Transmit precoding
 Receiver Shaping
 Spatial multiplexing
 Diversity coding.

PRECODING:

Precoding is a generalization of beam forming to suffer multilayer bans mission

in multi antenna wireless communication.

In general R symbols/s input data stream can be split into r-parallel. Independent
data streams, producing r-tuples at a rote R/r symbols/s. The actual input to the antenna
is generated through linear transformation on 𝑥 as
 𝑥 = 𝑀𝑥 `

Where M is an n*r fixed matrix. This operation is called transmit Precoding.

𝑥 = 𝑀𝑥 𝑦 = 𝑀𝑥 + 𝑁 𝑦 = 𝐹𝑦

Modulated

Symbol 𝑦

𝑥 𝑥 𝑦
P
r
e
CLASSIFICATION
c OF PRECODING:
o
d  Precoding for single user MIMO
i  Precoding for multi user MIMO
n
g

PRECODING
f FOR SINGLE USER MIMO:
o
r In
SU-MIMO, a transmitter equipped with multiplex antennas communicates
with a receiver that has multiple antennas. Most classic Precoding results assume
s
recommended
i
slow.
n
PRECODING FOR MULTI USER MIMO:
g
l
In
MU-MIMO a multiantenna transmitter communicates simultaneously with
e
multiple receivers. This is known as SDMA.
u
BEAM
s FORMING:
e
r Beam forming
is a technique that focuses radio signals directly on the target
antenna thereby improving range and performance by limiting interference.
M
I Beam forming is the method used to create the radiation patterns of an antenna
array.MIt can be applied in all antennas array system as well as MIMO systems. Beam
O
Forming is exactly analogous to frequency domain analysis of time signals.
 P
r
e
c
 x1 Y1

C1

X2 Y2
C2

x
Y3
X3
C3

It is no longer possible to transform the MIMO channel into non-interfering

SISO channels, Since the decoding complexity is exponential in r, we can keep the
complicity low by keeping r small.

A transmit strategy where the input contrarian‟s matrix has limit rank is called
beam forming. This corresponding to the producing matrix being just a column vector
M=C, the team forming vector.

Spatial mateched filtering yields a single SISO AWGW channel as follows

𝐶 𝐻
𝑦= y
𝐶 𝐻

𝐶 𝐻 𝐶 𝐻
= H𝐶x+ 𝑁
𝐶 𝐻 𝐶 𝐻

= 𝐻 𝐶 x+ 𝑁

SNR= 𝐶 *H* 𝐶 E[XX*]

TYPES OF BEAM FORMING:

Beam forming is divided into two groups. Are

Phased Array System with a finite number of fixed predefined patterns.

Adaptive Array Systems with an infinite number of patterns adjusted to the

scenario in real times.
 Switched beam former Adaptive beam former

Switched beam former electrically calculate the DOA (Direction Of Array

Arrival) and switch on the fixed beam. The user only has the optimum signal strength
along the center of the beam.

Adaptive beam former deals with the problem and adjusts the beam in real time
to the moving UE. The complexity is high.

ADVANTAGES OF BEAM FORMING:

It increases the received signal again, by making signals emitted from different
antennas add up constructively and to reduce the multipath fading channel effect.

TRANSMITTER & RECEIVER DIVERSITY:

In transmit diversity there are multiple antennas available at the transmitter and
the transmitted signal s(t) is sent over the i th antenna with a branch weight αi transmit
diversity is desirable in system such as Cellular system where more space, power,
processing capabilities is available on the transmit side versus the receive side the path
𝑗𝜃𝑖
again associated with the i th antenna is 𝑟𝑖𝑒 and the signals transmitted over all
antennas are added “in the air”, which leads to s received signal given by
𝑀 𝑗𝜃𝑖
r (t) = 𝑖=1 𝛼𝑖 𝑟𝑖𝑒 s(t)

Transmitter diversity produces the same diversity agains as receiver diversity.

TRNASMITTER DIVERSITY WITH CHANNEL STATE INFORMATION:

Channel state information is available at TX which knows the channel perfectly.

This knowledge might be obtained from feedback from the RX.

Here we find that there is a complete equivalence between transmitter diversity

and receive diversity. The optimum transmission scheme linearly weighs signals
transmitted from different antenna elements with the conjugates of the channel to a
single receive antenna. This approach is known as maximum ratio transmission.

TRANSMISSION DIVERSITY WITHOUT CHANNEL STATE

INFORMATION:
 CST is not available at the TX. We cannot simply transmit weighted copies of
the same signal from different transmit antennas, because we cannot know how would
they add up at the RX. It is equally likely for the addition of different components to be
costrutive or destruaive, in other words we would just be adding up MPS with random
process phase which results in Rayleigh fading.

 Delay Diversity
 Phase Sweeping Diversity
 Space Time Coding
RECEIVER DIVERSITY:

Receiver Diversity is a well-known technique to improve the performance of

wireless communications in fading channels.

Consider a single user system model where in the received single is a sum of the
desired signal and noise

Yn(t) = hn(t) + Nn(t) n=I,……,L

U(t) is the unit power, hn(t) == channel, Nn(t) = = noise

1 𝑇
P= 𝑕𝑛 (𝑡) 2 𝑢(𝑡) 2dt
𝑇𝑠 0

2 1 𝑇𝑠
= 𝑕𝑛 (𝑡) 𝑢(𝑡) 2 dt
𝑇𝑠 0

2
P= 𝑕𝑛
𝑕𝑛 2
E{|n(t)2} = 𝜍 2 , SNR(𝛾𝑛 ) =
𝜍2

1
𝛾𝑛 ≈ e-𝛾𝑛 𝑇
⎾

𝐸{|𝑕𝑛 |2 } 𝑃0
⎾ = E{𝛾𝑛 } = =
𝜍2 𝜍2

2 1 2
𝑕 SNR = LSNR. 𝑕
𝐿

Array gain Diversity gain

1 2 1 𝐿
𝑕 = 𝑙=1 𝑕𝑛[1] 2
𝐿 𝐿
 RCr Diversity Transmit Trans&receiver

CHANNEL STATE INFORMATION

Based on requirement of amount of CSI, algorithm for MIMO can be

categorized.

ALGORITHM CSI@TX CSI@RX ADVANTAGE DISADVANTAGE

Full CSI at TX TX has RX has High efficiency Difficult to obtain

and full CSI at perfect perfect full CSIT
RX. knowledge knowledge of
of channel. channel
Average CSI TX knows RX has all Easy to archive Require calibration
and full CSIR only average information no reciprocity
CSI

No CSI & full TX does not RX learns the Archived Require calibration
CSIR use any CSI instant without F/B
anuous cs
from training
sequence
Noisy CSI RX learned _ _ Any received
the CS training pulse will
perfectly be affected

CAPACITY IN FADING & NON FADING CHANNELS:

Non-fading channel:

By Shannon theorem, the information theoretic capacity equation for normal

AWGN channel is

Cshannon = log2 (1+ γ |H|2)

 For MIMO, the channel is represented as

H= [ 𝑈*] W

Ʃ DIAGONAL MATRIX

Received signal is then

γ = Hs + n

γ = W ƩU*S+n

Multiplication of the transmit data vector by matrix U and the received signal
vector by diaogonlizes the channel

W* γ = W*WƩU*𝑆 + W*n

γ = Ʃ𝑆 +𝑛

Since U and W are Unitary Matrices.

The capacity of channel H is thus given by the sum of the capacities of the Eigen
modes of the channel.
𝑅𝑛 𝑃𝑘
C= 𝑘=1 log 2 1+ 𝜍𝑘 2
𝜍𝑛 2

𝛾
C = log 2 𝑑𝑒𝑡 𝐼𝑁𝑟 + 𝐻𝑅𝑠𝑠𝐻 ∗
𝑁𝑡

Where INr is the identity matrix

γ is the mean SNR per RX branch

Rss is the correlation matrix

FADING CHANNEL:

We assume a describe time channel with stationary and ergodic time varying
gain 𝑔(𝑖). The channel power gain g(i) follows a given distribution. The channel g(i)
can change at each time I, either as an 11D processor with same correlation overtime.
𝑔(𝑖) n(i)

W Encoder y[i] Decoder w

 SNR;
𝑠
γ [i] = g[i] 0≤ γ[i] < ∞
𝑁0𝐵

C= 𝑠∈𝑠 𝐶𝑠 𝑝 𝑆
∞
C= 0 γ
𝐶𝑝 γ 𝑑γ
∞
C= 0
𝐵 log2 (1+ γ) p(γ) d γ

FADING CHANNEL CAPACITY WITH RECEIVER CSI;

∞
C= 0 γ
𝐶𝑝 γ 𝑑γ
∞
C= 0
𝐵 log2 (1+ γ) p(γ) d γ
∞ γ 𝑚 −1 𝑘!
E[C] = 0
𝑙𝑜𝑔 2 1+ 𝜆 𝑘=0 𝑘+𝑛−𝑚 !
𝑁𝑡

It is the exact expression for the ergodic capacity.

 QUESTION BANK

EC6801 WIRELESS COMMUNICATION

UNIT I WIRELESS CHANNELS

Large scale path loss – Path loss models: Free Space and Two-Ray models -Link Budget
design – Small scale fading- Parameters of mobile multipath channels – Time dispersion
parametersCoherence bandwidth – Doppler spread & Coherence time, Fading due to Multipath
time delay spread – flat fading – frequency selective fading – Fading due to Doppler spread –
fast fading – slow fading.
UNIT II CELLULAR ARCHITECTURE
Multiple Access techniques - FDMA, TDMA, CDMA – Capacity calculations–Cellular
conceptFrequency reuse - channel assignment- hand off- interference & system capacity-
trunking & grade of service – Coverage and capacity improvement.
UNIT III DIGITAL SIGNALING FOR FADING CHANNELS
Structure of a wireless communication link, Principles of Offset-QPSK, p/4-DQPSK,
Minimum Shift Keying, Gaussian Minimum Shift Keying, Error performance in fading
channels, OFDM principle – Cyclic prefix, Windowing, PAPR.
UNIT IV MULTIPATH MITIGATION TECHNIQUES
9 Equalisation – Adaptive equalization, Linear and Non-Linear equalization, Zero forcing and
LMS Algorithms. Diversity – Micro and Macrodiversity, Diversity combining techniques,
Error probability in fading channels with diversity reception, Rake receiver,
UNIT V MULTIPLE ANTENNA TECHNIQUES
MIMO systems – spatial multiplexing -System model -Pre-coding - Beam forming -
transmitter diversity, receiver diversity- Channel state information-capacity in fading and non-
fading channels.

TEXTBOOKS: 1. Rappaport,T.S., “Wireless communications”, Second Edition, Pearson

Education, 2010. 2. Andreas.F. Molisch, “Wireless Communications”, John Wiley – India,
2006.

REFERENCES: 1. David Tse and Pramod Viswanath, “Fundamentals of Wireless

Communication”, Cambridge University Press, 2005. 2. Upena Dalal, “ Wireless
Communication”, Oxford University Press, 2009. 3. Van Nee, R. and Ramji Prasad, “OFDM
for wireless multimedia communications”, Artech House, 2000

UNIT I: WIRELESS CHANNELS

Part-A
1. What are the propagation mechanisms of EM waves?
The four propagation mechanisms of EM waves are
i. Free space propagation
ii. Reflection
iii. Diffraction
iv. Scattering

2.What is the significance of propagation model?

The major significance of propagation model are:
i. Propagation model predicts the parameter of receiver.
ii. It predicts the average received signal strength at a given distance from the
transmitter.

3.What do you mean by small scale fading?

Rapid fluctuations of the amplitude, phase as multipath delays of a radio signal over a
short period of time is called small scale fading.

4.What are the factors influencing small scale fading?

The factors which influence small scale fading are:
Multipath propagation, Speed of the mobile, Speed of surrounding objects and the
transmission bandwidth of the signal.

5.When does large scale propagation occur?

Large scale propagation occurs due to general terrain and the density and height of
buildings and vegetation, large scale propagation occurs.

6. Differentiate the propagation effects with mobile radio.

Slow Fading Fast Fading

Slow variations in the signal strength.
Mobile station (MS) moves slowly.
It occurs when the large reflectors and
diffracting objects along the transmission
paths are distant from the terminal.
Eg. Rayleigh fading, Rician fading and
Doppler shift
Rapid variations in the signal strength.
Local objects reflect the signal causes
fast fading.
It occurs when the user terminal (MS)
moves for short distances.

7. Define Doppler shift.

If the receiver is moving towards the source, then the zero crossings of the
signal appear faster and the received frequency is higher.The opposite effect occurs if
the receiver is moving away from the source. The resulting chance in frequency is
known as the Doppler shift (fD).
FD = fr – f0 = -f0V/C
Where f0 -> transmission frequency
fr -> received frequency

8. Differentiate time selective and frequency selective channel.

 The gain and the signal strength of the received signal are time varying means
then the channel is described as time selective channel. The frequency response of the
time selective channel is constant so that frequency flat channel. The channel is time
invariant but the impulse response of the channel show a frequency-dependent
response so called frequency selective channel.

9. Define coherence time and coherence bandwidth.

Coherence time is the maximum duration for which the channel can be
assumed to be approximately constant. It is the time separation of the two time
domain samples. Coherence bandwidth is the frequency separation of the two
frequency domain samples.

10. What do you mean by WSSUS channels?

In multipath channels, the gain and phase shift at one delay are uncorrelated with
another delay is known as uncorrelated scattering of WSSUS.

11. What is free space propagation model?

The free space propagation model is used to predict received signal strength,
when unobstructed line-of-sight path between transmitter & receiver. Friis free space
equation is given by,

The factor (λ/4πd)2 is also known as the free space loss factor.

12. Define EIRP.

EIRP (Equivalent Isotropically Radiated Power) of a transmitting system in a given
direction is defined as the transmitter power that would be needed, with an isotropic
radiator, to produce the same power density in the given direction.
EIRP=PtGt
Where Pt-transmitted power in W
Gt-transmitting antenna gain

13. Explain path loss.

The path loss is defined as the difference (in dB) between the effective
transmitted power and the received power. Path loss may or may not include the effect
of the antenna gains.

14. What is intrinsic impedance and Brewster angle?

Intrinsic impedance is defined by the ratio of electric to magnetic field for a
uniform plane wave in the particular medium.
Brewster angle is the angle at which no reflection occurs in the origin. Brewster angle
is denoted by θB as shown below,
 15. What is scattering?
When a radio wave impinges on a rough surface, the reflected energy is spread
out in all directions due to scattering.

16. Define radar cross section.

Radar Cross Section of a scattering object is defined as the ratio of the power
density of the signal scattered in the direction of the receiver to the power density of
the radio wave incident upon the scattering object & has units of squares meters

17. Name some of the outdoor propagation models?

Some of the commonly used outdoor propagation models are
i. Longely-Rice model
ii. Durkin’s model
iii. Okumura model.

18. Define indoor propagation models.

The indoor propagation models are used to characterizing radio propagation
inside the buildings. The distances covered are much smaller, and the variability of the
environment is much greater for smaller range of Transmitter and receiver separation
distances. Features such as lay-out of the building, the construction
materials, and the building type strongly influence the propagation within the building.

19. Mention some indoor propagation models?

Some of the indoor propagation models
are:
i. Long –distance path loss model
ii. Ericession multiple break point model
iii. Attenuation factor model.

20. What are merits and demerits of Okumara’s model?

Merits:
Accuracy in parameter prediction.
Suitable for modern land mobile radio system.
Urban, suburban areas are analyzed. Demerits:
Rural areas are not analyzed.
Analytical explanation is not enough.

21. List the advantages and disadvantages of Hata model?

Advantages: Suitable for large cell mobile system. Cell radius on the order of
1km is taken for analysis.
Disadvantages: Not suitable for PCS model. This model does not have any path
specific correction.
 22. What is the necessity of link budget?
The necessities of link budget are:
i. A link budget is the clearest and most intuitive way of computing the required
Transmitter power. It tabulates all equations that connect the Transmitter
power to the received SNR
ii. It is reliable for communications.
iii. It is used to ensure the sufficient receiver power is available.
iv. To meet the SNR requirement link budget is
calculated.

UNIT II CELLULAR ARCHITECTURE

Part-A

1. What is meant by frequency reuse?

If an area is served by a single Base Station, then the available spectrum can be divided
into N frequency channels that can serve N users simultaneously. If more than N users are to be
served, multiple BSs are required, and frequency channels have to be reused in different
locations. Since spectrum is limited, the same spectrum has to be used for different wireless
connections in different locations. This method of reusing the frequency is called as frequency
reuse.

2. What are the trends in cellular radio systems?

The trends in personal cellular radio systems are:
i. PCS – Personal Communication Services
ii. PCN – Personal Communication Networks

3. What do you mean by forward and reverse channel?

Forward channel is a radio channel used for transmission of information from base
station to mobile.Reverse channel is a radio channel used for transmission from mobile to base
station.

4. What is the function of control channel? What are the types?

The function of control channel is to transmit call setup, call request, call initiation and
Control. There are two types of control channels,

i. Forward control channel

ii. Reverse control channel

5. What is channel assignment? What are the types?

For efficient utilization of radio spectrum a frequency reuse scheme with increasing
capacity and minimizing interference is required. For this channel assignment is used. The
types of channel assignment are:
 i. Fixed channel assignment
ii. Dynamic channel assignment.

6. What is fixed channel assignment?

If the channels in each cell are allocated to the users within the cell, it will be called as
fixed channel assignment. If all channels are occupied, the call will be blocked.

7. What is dynamic channel assignment?

If the voice channels are not allocated permanently in a cell, it will be called as
dynamic channel assignment. In this assignment, channels are dynamically allocated to users
by the MSC.
8. Define MS, BS and MSC.
MS – Mobile station. A station in the cellular radio service intended for use.
BS – Base Station. A fixed station in a mobile radio system used for radio
communication with MS.
MSC – Mobile Switching Centre. Mobile switching centre coordinates the routing of
calls in large service area. It connects the base station and mobiles to PSTN. It is also
called as MTSO(Mobile telephone switching office.

9. Define hand off and mode of hand off.

A handoff refers to the process of transferring an active call or data session from one
cell in a cellular network to another or from one channel in a cell to another. A well-
implemented handoff is important for delivering uninterrupted service to a caller or data
session user. Modes of hand off are:
i. MCHO – Mobile Controlled Hand off
ii. NCHO – Network Controlled Hand off
iii. MAHO – Mobile Assisted Hand off

10. Write the types of hand off.

Types of handoff are:
i. Hard hand off – Mobile monitors BS and new cell is allocated to a call with strong
signal.
ii. Soft hand off – MS with 2 or more calls at the same time and find which is the
strongest signal BS, the MSC automatically transfers the call to that BS.

11. Define Cell, Cluster.

For a large geographic coverage area, a high powered transmitter therefore has to be
used. But a high power radio transmitter causes harm to environment. Mobile communication
thus calls for replacing the high power transmitters by low power transmitters by dividing the
coverage area into small segments, called cells.
Each cell uses a certain number of the available channels and a group of adjacent cells together
use all the available channels. Such a group is called a cluster.
12. What do you mean by foot print and dwell time?
The region over which the signal strength lies above this threshold value x dB is known
as the coverage area of a BS and it must be a circular region, considering the BS to be isotropic
radiator. Such a circle, which gives this actual radio coverage, is called the foot print of a
cell.The time over which a call may be maintained within a cell without hand off is called the
dwell time.

13. What are the major types of cellular interference?

The major types of cellular interferences are as follows

i. CCI – Co-channel interference is the interference between signals from co-channel

cells.
ii. ACI – Adjacent channel interference resulting from signals which are adjacent in
frequency to the desired signal.

14. What are the techniques used to expand the capacity of cellular system?
Cell splitting, Sectoring, Coverage Zone approaches are the techniques used to expand
the capacity of cellular system.
Cell splitting – Cell-splitting is a technique which has the capability to add new smaller
cells in specific areas of the system. i.e. divide large cell size into small size.
Sectoring – use of directional antennas to reduce Co-channel interference.
Coverage Zone approaches – large central BS is replaced by several low power
transmitters on the edge of the cell.

15. What is frequency reuse ratio?

If the cell size and the power transmitted at the base stations are same then co-channel
interference will become independent of the transmitted power and will depend on radius of
the cell (R) and the distance between the interfering co-channel cells (D). If D/R ratio is
increased, then the effective distance between the co-channel cells will increase and
interference will decrease. The parameter Q is called the frequency reuse ratio and is related to
the cluster size. For hexagonal geometry

Q= =

From the above equation, small of `Q' means small value of cluster size `N' and increase in
cellular capacity.

16. Define FDMA, TDMA and CDMA.

FDMA - the total bandwidth is divided into non-overlapping frequency subbands.
TDMA – divides the radio spectrum into time slots and in each slot only one user is
allowed to either transmit or receive.
CDMA – many users share the same frequency same tome with different coding.

17. Define Grade of service.

Grade of service is defined as the measure of the ability of a user to access a trunked
system during the busiest hour.
18. What is blocked call clear system (BCC)?
In a system, a user is blocked without access by a system when no channels are
available in the system. The call blocked by the system is cleared and the user should try again
.This is called BCC system.

19. What is blocked call delay system?

If a channel is not available immediately, the call request may be delayed until a channel
becomes available. This is called as blocked call delay system.

20. Define cell splitting.

Cell splitting is the process of subdividing congested cells into smaller cells each with
its own base stations and a corresponding reduction in antenna height and transmitter power. It
increases the capacity of cellular system.

21. What is sectoring?

Sectoring is a technique for decreasing co-channel interference and thus increasing the
system performance by using directional antennas.

22. What are the features of TDMA?

Features of TDMA are:
i. TDMA shares a single carrier frequency with several users, where each user makes use
of non overlapping time slots.
ii. Data transmission occurs in bursts.
iii. Handoff process is much simpler
iv. Duplexers are not required, since transmission and reception occurs at different time
slots.

23. What are the features of FDMA?

Features of FDMA are:

i. FDMA channel carries only one phone circuit at a time

ii. The bandwidth of FDMA channels are relatively narrow as each channel supports only
one circuit per carrier.

24. Write the two types of spread

spectrum? Types of spread
spectrum are:
Direct Sequence Spread Spectrum (DS-SS)
Frequency hop spread spectrum (FH-SS)

25. What do you mean by spread spectrum?

Spread spectrum multiple access uses signals which have a transmission bandwidth
whose magnitude is greater than the minimum required RF bandwidth. A pseudo noise
(PN) sequence converts a narrowband signal to a wideband noise like signal before
transmission

26. What is PN sequence?

Pseudo noise sequence is a coded sequence of 1‟s and 0‟s with autocorrelation
 properties.

27. When is the PN sequence called as maximal length sequence?

When the pseudo-noise sequence generated by linear feedback shift register has the
length (N) of 2m-1 where m is number of stages in shift register is called maximal
length sequence.

28. Write the properties which a PN sequence should have.

Properties of PN sequence are:
i. Balance property
ii. Run property
iii. Correlation property

29. Define chip duration and chip rate.

The duration of every bit in PN sequence is known as chip duration. The number of
bits (chips) per second is called chip rate.

30. State the principles of CDMA.

Principles of CDMA:
i. Many users share the same frequency.
ii. Each user is assigned a different spreading code.

31. How the capacity can be increased in CDMA?

Capacity in CDMA can be increased by
iii. Quiet periods during speech transmission is shared by many users.
iv. Flexible data rate.
v. Soft capacity.
vi. Error Correction coding used.
PART-B

1. Explain in detail the types of services.

ANS: Refer section 1.2 in Wireless communication, Andreas f molisch Pg.No: 8

2. With the help of a neat diagram explain about frequency reuse and the advantages of it.
ANS: Refer section 3.2 in Wireless communication, Rappaport Pg.No: 58

Diagram

Derivation for N=3, 7 and 12

The advantages of frequency reuse

3. Write short notes on fading and intersymbol interference.

ANS: Refer section 2.1 in Wireless communication, Andreas f molisch Pg.No: 27

i. Explain in detail the spectrum limitations that carried out in wireless communication.

ANS : Refer section 2.2 in Wireless communication, Andreas f molisch pg.no:29

ii. Explain in detail the principle of cellular network.

ANS : Refer section 17.6 in Wireless communication, Andreas f molisch pg.no:379

6. Give the difference between FDMA and TDMA.

ANS : Refer section 17.2 in Wireless communication, Andreas f molisch pg.no:366

PART-B

1. Explain the different techniques of improving coverage and capacity in Cellular System.

Explanation about cell splitting

Explanation about sectoring

Explanation about Microcell zone approach

ANS : Refer section 3.7 in Wireless communication, Rappaport Pg.No:86 - 93 4.

2. Explain in detail about reverse CDMA channel.

ANS: Refer section 11.4.3 in Wireless communication, Rappaport Pg.No: 575

3.Compare and contrast the features of FDMA, TDMA and CDMA

 ANS: Comparison based on Bandwidth, Security, Efficiency

ANS: Refer section 9.1.1 in Wireless communication, Rappaport Pg.No: 448

4. With neat diagram explain the forward CDMA channel Structure

ANS: Frequency Hopping, Explanation, Direct Sequence

ANS: Refer section 11.4.2 in Wireless communication, Rappaport Pg.No: 569

5. Explain in detail about DECT?

UNIT III DIGITAL SIGNALING FOR FADING CHANNELS

Part-A

1. List the advantages of digital modulation techniques.

The advantages of digital modulation techniques are:
i. Immunity to channel noise and external interference.
ii. Flexibility operation of the system.
iii. Security of information.
 iv. Reliable since digital circuits are used.
v. Multiplexing of various sources of information into a common format is possible.
vi. Error detection and correction is easy.

2. What are the factors that influence the choice of digital

modulation? The factors that influence the choice of digital
modulation are:
i. Low BER at low received SNR.
ii. Better performance in multipath and fading conditions.
iii. Minimum bandwidth requirement.
iv. Better power efficiency.
v. Ease of implementation and low cost.

3. Define power efficiency and bandwidth efficiency.

Power efficiency describes the ability of a modulation technique to preserve the
fidelity of the digital message at low power levels.
ɳp = Eb/N0 = Bit energy / Noise power spectral density
Ability of a modulation scheme to accommodate data within a limited bandwidth is called
bandwidth efficiency.
ɳB = R/B = Datarate / Bandwidth in bps/Hz

4. What is QPSK?
The Quadrature Phase Shift Keying (QPSK) is a 4-ary PSK signal. The phase of the
carrier in the QPSK takes 1 of 4 equally spaced shifts.
Two successive bits in the data sequence are grouped together. 1
symbol = 2 bits
This reduces bit rate and bandwidth of the channel. Coherent
QPSK = 2 x coherent BPSK system
The phase of the carrier takes on one of four equally spaced values such as π/4, 3π/4, 5π/4
and 7π/4.

5. Define offset QPSK and π/4 differential QPSK.

In offset QPSK the amplitude of data pulses are kept constant. The time alignment
of the even and odd bit streams are offset by one bit period in offset QPSK.
In π/4 QPSK, signaling points of the modulated signal are selected from two QPSK
constellations which are shifted by π/4 with respect to each other. It is differentially encoded
and detected so called π/4 differential QPSK.

6. What is meant by MSK?

A continuous phase FSK signal with a deviation ratio of one half is referred to as
MSK. It is a spectrally efficient modulation scheme.

7. List the salient features of MSK scheme.

Salient features of MSK are:
i. It has constant envelope, smoother waveforms than QPSK.
ii. Relatively narrow bandwidth.
iii. Coherent detection suitable for satellite communications.
iv. Side lobes are zero outside the frequency band, so it has resistance to co-
channel
 interference.

8. Why GMSK is preferred for multiuser, cellular communication?

It is a simple binary modulation scheme.
Premodulation is done by Gaussian pulse shaping filter, so side lobe levels are much
reduced. GMSK has excellent power efficiency and spectral efficiency than FSK.
For the above reasons GMSK is preferred for multiuser, cellular communication.

9. How can we improve the performance of digital modulation under fading channels?
By the using of diversity technique, error control coding and equalization techniques
performance of the digital modulation under fading channels are improved.

10. Write the advantages of MSK over QPSK.

Advantages of MSK over QPSK:
iii. In QPSK the phase changes by 90degree or 180 degree .This creates abrupt
amplitude variations in the waveform, Therefore bandwidth requirement of QPSK is
more filters of other methods overcome these problems , but they have other side
effects.
iv. MSK overcomes those problems. In MSK the output waveform is continuous in
phase hence there are no abrupt changes in amplitude.

11. Define M-ary transmission system?

In digital modulations instead of transmitting one bit at a time, two or more bits are
transmitted simultaneously. This is called M-ary transmission.

12. What is quadrature modulation?

Sometimes two or more quadrature carriers are used for modulation. It is called
quadrature modulation.

13. What is QAM?

At high bit rates a combination of ASK and PSK is employed in order to minimize
the errors in the received data. This method is known as “Quadrature Amplitude
Modulation”.

14. Define QPSK

QPSK is defined as the multilevel modulation scheme in which four phase shifts are
used for representing four different symbols.

15. What is linear modulation?

In linear modulation technique the amplitude of the transmitted signal varies linearly with
the modulating digital signal. In general, linear modulation does not have a constant
envelope.

16. Define non linear modulation.

In the non linear modulation the amplitude of the carrier is constant, regardless of the
variation in the modulating signals.
 Non-linear modulations may have either linear or constant envelopes depending on whether
or not the baseband waveform is pulse shaped.

17. What is the need of Gaussian filter?

Need for Gaussian Filter:
i. Gaussian filter is used before the modulator to reduce the transmitted bandwidth
of the signal.
ii. It uses less bandwidth than conventional FSK.

18. Mention some merits of MSK.

Merits of MSK:
i. Constant envelope
ii. Spectral efficiency
iii. Good BER performance
iv. Self-synchronizing capability
v. MSK is a spectrally efficient modulation scheme and is particularly attractive for use
in mobile radio communication systems.

19. Give some examples of linear modulation.

Examples of linear modulation:
i. Pulse shaped QPSK
ii. OQPSK
20..Write short notes on OFDM.
OFDM splits the information into N distinct carriers and then transmitted. receiver, they
have to be orthogonal.
parallel streams which are modulated by N In order to separate the subcarriers by the
21.. Why cyclic prefix?
In delay dispersive channel, inter carrier interference occur. To overcome the effect
of inter carrier interference and ISI, cyclic prefix is introduced. It is a cyclically extended
guard interval whereby each symbol sequence is preceded by a periodic extension of the
sequence itself.

PART-B

1. Write about the GMSK transmitter and receiver with neat diagram?

ANS: Refer section 6.93 in Wireless communication, Rappaport Pg.No: 320

2. Write about the BPSK with neat diagram?

ANS: Refer section 11.3.1 in Wireless communication, Andreas f molisch Pg.No: 196

3. Write about the QPSK transmitter and receiver with neat diagram?

ANS: Refer section 11.3.2 in Wireless communication, Andreas f molisch Pg.No: 199
4. Explain the Free space propagation model?
ANS: Refer section 4.2 in Wireless communication, Rappaport Pg.No: 107

5. Write about the BFSK transmitter and receiver with neat diagram?
ANS: Refer section 11.3.6 in Wireless communication, Andreas f molisch Pg.No: 208

6. Write about the MSK and DMSK with neat diagram?

ANS: Refer section 11.3.7 and 11.3.8 in Wireless communication, Andreas f molisch Pg.No:
212 - 214

7. Explain in detail the power spectrum and error performance of Gausian MSK.
ANS: Refer section 11.3.9 in Wireless communication, Andreas f molisch Pg.No: 215

8. Explain in detail the error probability in flat fading channels.

ANS: Refer section 12.2 in Wireless communication, Andreas f molisch Pg.No: 332-
338

9. Explain in detail the error probability in delay and frequency dispersive fading
channels
ANS: Refer section 12.3 in Wireless communication, Andreas f molisch Pg.No: 339-
345

UNIT IV MULTIPATH MITIGATION TECHNIQUES

Part-A
1. What are the techniques used to improve the received signal quality?
Techniques such as,
Equalization
Diversity
Channel coding
are used to improve the received signal quality.

2. What is the need of equalization?

Equalization can be used to compensate the Inter Symbol Interference created by
multipath within time dispersion channel.

3. What is diversity?
Diversity is used to compensate the fading channel impairments and is usually
implemented by using two or more receiving antennas. Diversity improves transmission
performance by making use of more than one independently faded version of the transmitted
signal.

4. Define spatial diversity.

 The most common diversity technique is spatial diversity, whereby multiple
antennas are strategically spaced and connected to a common receiving system. While one
antenna sees a signal null, one of the other antenna may sees a signal peak, and the receiver
is able to select the antenna with the best signals at any time.
5. Define STCM.
Channel coding can also be combined with diversity a technique called Space-Time
Coded Modulation. The space-time coding is a bandwidth and power efficient method for
wireless communication.
6. Define adaptive equalization?
To combine Inter Symbol Interference, the equalizer coefficients should change
according to the channel status so as to break channel variations. Such an equalizer is called
an adaptive equalizer since it adapts to the channel variations.
7. Define training mode in an adaptive equalizer?
First, a known fixed length training sequence is sent by the transmitter then the
receivers equalizers may adapt to a proper setting of minimum bit error detection where the
training sequence is a pseudo random binary signal or a fixed and prescribed bit pattern.

8. What is tracking mode in an adaptive equalizer?

Immediately following this training sequence the user data is sent and the adaptive
equalizer at the receiver utilizes a recursive algorithm to evaluate the channel and estimate
filter coefficients to compensate for the distortion created by multipath in the channel.

9. Write a short note on linear equalizers and non linear equalizers?

Linear equalizers: If the output d(t) is not used in the feedback path to adapt the
equalizer. This type of equalizers is called linear equalizer.
Nonlinear equalizers: If the output d(t) is fed back to change the subsequent outputs of the
equalizers is called non linear equalizers.

10. Why non linear equalizers are preferred?

The linear equalizers are very effective in equalizing channels where ISI is not
severe.The severity of the ISI is directly related to the spectral characteristics. In this case
that there are spectral noise in the transfer function of the effective channel, the additive
noise at the receiver input will be dramatically enhanced by the linear equalizer. To
overcome this problem non linear equalizers are used.

11. What are the nonlinear equalization methods used?

Commonly used non linear equalization methods are:
i. Decision feedback equalization
ii. Maximum likelihood symbol detection
iii. Maximum likelihood sequence estimation

12. What are the factors used in adaptive algorithms?

Rate of convergence Mis
adjustments
Computational complexity

13. Define diversity concept.

 If one radio path undergoes a deep fade, another independent path may have a strong signal.
By having more than one path to select from, both the instantaneous and average SNRs at the
receiver may be improved often by as much as 20dB to 30dB. The principle of diversity is to
ensure that the same information reaches the receiver on statistically independent channels.

14. How the link performance can be improved?

Link performance can be improved by various techniques such as
i. Equalization
ii. Diversity
iii. Channel coding

15. Why diversity and equalization techniques are used?

To reduce ISI, Equalization technique is used. Diversity is used to reduce fading effects.
16. What is diversity?
Signal is transmitted by more than one antenna via channel. It ensures that the same
information reaches the receiver on statistically independent channels.

17. Differentiate selection diversity and combining diversity.

Selection Diversity Combining Diversity
The best signal is selected and processed All signals are combined before
while all other signals are discarded. processing and the combined signal is
decoded.
Simple circuits are used. At individual receiver, phasing circuits
are needed.
None of the signal is not in acceptable It works well.
SNR.

18. Define Switched Diversity

If the signal level falls below the threshold, then the receiver switches to a new
antenna which is called as switched diversity.

19. Define feedback or scanning diversity.

All the signals are scanned in a fixed sequence until one signal is found to be above
a predetermined threshold.
20. Define temporal diversity.
Wireless propagation channel is time variant, so for sufficient decorrelation, the temporal
distance between antennas must be atleast the half of maximum Doppler frequency.

21. What is meant by frequency diversity?

Correlation is increased by transmitting information on more than one carrier frequency.
Frequencies are separated by more than one coherence bandwidth of the channel. So the
signals will not experience same fades.
22. Differentiate micro and macro diversity.
Micro diversity
Used to reduce small scale fading effects.
Multiple reflection causes deep fading.
This effect is reduced.
BS-MS are separated by small distance.
Macro diversity
Used to reduce large scale fading effects.
Deep shadow causes fading. This effect is
reduced.
BS-MS are separated by large distance.

23. What is transmit diversity?

Diversity effect is achieved by transmitting signals from several transmit antenna.

24. What is an equalizer?

Equalizer is a linear pulse shaping circuit which is used to reduce ISI.

25. What is linear and non-linear equalizer?

Linear equalizer: the current and past values of the received signal are linearly
weighted by the filter coefficients and summed to produce the output. No feedback
path is used. Simple and easy to implement. Not suitable for severely distorted channel.
Noise power signal is enhanced.
Nonlinear equalizer: If the past decisions are correct, then the ISI contributed by present
symbol can be cancelled exactly, feedback path is used. Suitable for severely distorted
channel. Noise power signal is not enhanced. Complex in structure. channels with low
SNR. Suffers from error propagation.

PART-B

1. Briefly explain the vitterbi-decoding algorithm.

ANS: Refer section 14.3.2 in Wireless communication, Andreas F Molisch Pg.No: 290.

2. With a diagram explain the performance of RAKE receiver?

ANS: Refer section 7.11 in Wireless communication, Rappaport Pg.No: 391 5.

3. Enumerate the fundamental of equalization and reduction in intersymbol interference in

communication channels.
 ANS: Refer section 7.2 in Wireless communication, Rappaport Pg.No: 356

4. What is the non linear equalization? Explain the three non linear methods of Equalization
with Suitable diagrams?
ANS: Refer section 7.7 in Wireless communication, Rappaport Pg.No: 368

5. Draw the block diagram of LPC coding system and explain the different types of LPC used for
Wireless systems?
ANS: Refer section 8.7 in Wireless communication, Rappaport Pg.No: 431

6. Draw the diagram of a rate -1/2-convolution encoder with constraint length 3. What
is the
generator polynomial of the encoder? Find the encoded sequence you have
drawn, corresponding to the message sequence.
ANS: Refer section 14.3 in Wireless communication, Andreas F Molisch Pg.No: 285.

7. Draw the block diagram of adaptive equalization and explain.

ANS: Refer Wireless communication, Andreas F Molisch Pg.No: 295.

UNIT V MULTIPLE ANTENNA TECHNIQUES

Part-A

1. Why need MIMO system?

The wireless system before MIMO is been constrained by network capacity which is
related with channel quality and coverage. To see how problem occurred, we need to
talk about the transmission on a multipath channel. In wireless communication the
propagation channel is characterized by multipath propagation due to scattering on
different obstacle. The multipath problem is a typical issue in communication system
with time variations and time spread. For time variations the channel is fading and
caused SNR variations. For time spread, it becomes important for suitable frequency
selectivity.

2. What are the Types of MIMO System

There are two majorclassifications to determine types of MIMO:

(1) Single User MIMO (SU-MIMO) vs. Multi User MIMO (MU-MIMO)
(2) Open loop MIMO vs. Close loop MIMO

3. What is Single User MIMO (SU-MIMO):

When the data rate is to be increased for a single UE, this is called Single User
MIMO(SU-MIMO).

4. What is Multi User MIMO (MU-MIMO):

When the individual streams are assigned to various users, this is called Multi
UserMIMO (MU-MIMO). This mode is particularly useful in the uplink because the
complexity on the UE side can be kept at a minimum by using only one transmitantenna. This
is also called 'collaborative MIMO'.
5. What is Space Time Transmit Diversity (STTD) MIMO
 Space-time block coding based transmit diversity (STTD) is a method of transmit
diversity used in UMTSS third-generation cellular systems. STTD is optional in
the UTRANN air interface but mandatory for user equipment. STTD utilizes space-time
block code (STBC) in order to exploit redundancy in multiply transmitted versions of a
signal.The same data is coded and transmitted through different antennas, which effectively
doubles the power in the channel. This improves Signal Noise Ratio (SNR) for cell edge
performance.

6. What is Spatial Multiplexing (SM) MIMO

Spatial multiplexing is transmission techniques in MIMO wireless communication to
transmit independent and separately encoded data signals, so-called streams, from each of the
multiple transmit antennas. Therefore, the space dimension is reused, or multiplexed, more
than one time. SMdelivers parallel streams of data to CPE by exploiting multi-path. It can
double (2x2 MIMO) or quadruple (4x4) capacity and throughput.SM gives higher capacity
when RF conditions are favorable andusers are closer to the BTS.

7. what is beamforming?
Beamforming or spatial filtering is a signal processing technique used in sensor arrays for
directional signal transmission or reception.[1] This is achieved by combining elements in
a phased array in such a way that signals at particular angles experience
constructive interference while others experience destructive interference. Beamforming can
be used at both the transmitting and receiving ends in order to achieve spatial selectivity. The
improvement compared with omnidirectional reception/transmission is known as the
receive/transmit gain (or loss).
8. Define transmit precoding

Precoding is a generalization of beamforming to support multi-stream (or multi-layer)

transmission in multi-antenna wireless communications. In conventional single-stream
beamforming, the same signal is emitted from each of the transmit antennas with appropriate
weighting (phase and gain) such that the signal power is maximized at the receiver output.
When the receiver has multiple antennas, single-stream beamforming cannot simultaneously
maximize the signal level at all of the receive antennas. In order to maximize the throughput
in multiple receive antenna systems, multi-stream transmission is generally required.

9. What are Beamforming techniques

Beamforming techniques can be broadly divided into two categories:

 conventional (fixed or switched beam) beamformers

 adaptive beamformers or phased array

 Desired signal maximization mode

 Interference signal minimization or cancellation mode

10. Define transmit diversity

Transmit diversity is radio communication using signals that originate from two or more
independent sources that have been modulated with identical information-bearing signals and
that may vary in their transmission characteristics at any given instant.
11. Define Spatial diversity

Spatial diversity employs multiple antennas, usually with the same characteristics, that
are physically separated from one another. Depending upon the expected incidence of the
incoming signal, sometimes a space on the order of a wavelength is sufficient. Other
times much larger distances are needed. Cellularization or sectorization, for example, is a
spatial diversity scheme that can have antennas or base stations miles apart. This is
especially beneficial for the mobile communication industry since it allows multiple users
to share a limited communication spectrum and avoid co-channel interference.

12. Define Pattern diversity

Pattern diversity consists of two or more co-located antennas with different radiation
patterns. This type of diversity makes use of directive antennas that are usually physically
separated by some (often short) distance. Collectively they are capable of discriminating
a large portion of angle space and can provide a higher gain versus a single
omnidirectional radiator.

13. . Define Polarization diversity

Polarization diversity combines pairs of antennas with orthogonal polarizations (i.e.

horizontal/vertical, ± slant 45°, Left-hand/Right-hand CP etc.). Reflected signals can
undergo polarization changes depending on the medium through which they are
travelling. A polarisation difference of 90° will result in an attenuation factor of up to
34dB in signal strength. By pairing two complementary polarizations, this scheme can
immunize a system from polarization mismatches that would otherwise cause signal fade.
Additionally, such diversity has proven valuable at radio and mobile communication base
stations since it is less susceptible to the near random orientations of transmitting
antennas.

14. Define Transmit/Receive diversity

Transmit/Receive diversity uses two separate, collocated antennas for transmit and
receive functions. Such a configuration eliminates the need for a duplexer and can protect
sensitive receiver components from the high power used in transmit.

15. Define ergodic capacity

This is the expected value of capacity (or) instantaneous capacity taken over all realizations
of the channel.

16. Define outage capacity

This is the minimum transmission rate that is achieved over a certain fraction of time of 90-
95%.

17. Define capacity of a fading channel

 2
the capacity of a channel is given as C=log2(1+γӀHӀ ); where γ is the signal to noise ratio at
the receiver; H is normalized transfer function from the transmitter to the receiver.

18. why are smart antennas required?

1. increasing coverage 2. increasing capacity 3. improving link quality

4. decrease delay dispersion 5. improvement of user position estimation.

part-B
1. With diagram explain the system model for MIMO systems.

2. Discuss about the operation of spatial multiplexing systems.

3. Explain the operation of transmit precoding and receiver precoding schemes

4. Why is beamforming important for wireless systems. With illustration

explain transmit beamforming, receive beamforming and opertunistic
beamforming.

5. Using diagrams explain transmit diversity and receive diversity.

6. derive the channel capacity of a fading channel for information transmitted

from a wireless system.

7. derive the capacity of non-fading channel for information transmitted

from a wireless system.