WC Notes
WC Notes
WC Notes
OBJECTIVES:
The student should be made to:
• 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
TOTAL: 45 PERIODS
1
2
UNIT I
WIRELESS CHANNELS
The free space propagation model is used to predict received signal strength when the transmitter
and receiver have a clear, unobstructed line of the sight path between them. Satellite
communication systems and microwave line – of- sight radio links typically undergo free
space propagation.
In large – scale radio wave propagation models, the free space model predicts that received
power decays as a function of T-R separation distance raised to some power. The free space
power received by receiver antenna which is separated from a radiating transmitter antenna by a
distance d, is given by Friis free space equation.
The effective aperture Ae is related to the physical size of the antenna, λ is related to
the carrier frequency b
3
Where, f – carrier frequency in Hertz
The values for Pt and Pr must be expressed in the same units, and Gt and Gr is dimensionless
quantities. The various losses L(L≥1) are usually due to transmission line attenuation system.
A value of L=1 indicates no loss in the system hardware.
The free space equation 1 shows that the received power falls off as the square of the T-R
separation distance. This implies that the received power decays with distance at a rate of 20
dB/decade.
The isotropic radiation is an ideal antenna which radiates power with unit gain uniformly in all
directions, and is often used to reference antenna gains in wireless systems.
and represents the maximum radiated power available from a transmitter in the direction
of maximum antenna gain as compared to an isotropic radiator.
The path loss, represents the signal attenuation as a positive quantity measured in dB is clear as
the difference (in dB) between the effective transmitted power and the received power, may or
may not be included the effect of the antenna gains.
The path loss for the free space model when antenna gains are included is given by,
When antenna gains are excluded, the antennas are assumed to have unity gain, and path loss is
given by,
4
The Friis free space model is only a valid prediction for Pr for values of which are in the far–
field of the transmitting antenna. The far – field of (or) fraunhofer region, of a transmitting
antenna is defined as the region beyond the largest linear dimensions of transmitter antenna
aperture and the carrier wave length.
Where,
D – Largest physical linear dimension of the antenna.
Equation 1 does not hold for d=0. So, large scale propagation models use a close – in distance
do as a known received power reference point.
The received power in free space at a distance greater than do is given by
In mobile radio systems, it is unusual to find that Pr may change by many orders of magnitude
over a typical coverage area of several square kilometers. Because of the large dynamic range of
received power levels, often dBm or dBw units are used to express received power levels
Equation 8 can be expressed in dBm or dBw by simply taking logarithm of both sides and
multiplying by 10. For E.g., If Pr is in units of dBm, the received power is given by,
In a mobile radio channel, a single direct path between the base station and a mobile is
rarely occurs for propagation and hence the free space propagation model is in most cases
inaccurate when used alone.
The two-ray ground reflection model is a useful propagation model that is based on
geometric options, and considers both the direct path and a ground reflected
propagation path between transmitter and receiver.
The total received E-field; ETOT is then a result of the direct line of sight
component, ELOS and the ground reflected component Eg.
In this above figure.1, ht is the heights of transmitter and hr is the height of the receiver.
If Eo is free space E-field (units V/m)at a reference distance do from the transmitter, then for
d>do, the free space propagating E-field is given by,
6
Where E(d,t) = Eodo/d represents the envelope the E-field at d meters from the
transmitter.
Two propagating waves arrive at the receiver. The direct wave that travels a distance and
the reflected wave that travels distance . The E field due to the Los component at
the receiver can be expressed as,
Where Ѓ is the reflection co-efficient for ground. For small values of θi , the
reflected wave is equal in magnitude and 180° out of phase with the incident wave.
The resultant E-field, assuming perfect horizontal E-field polarization and ground
reflection is the vector sum of ELOS and Eg, and the resultant total E-field envelope is given
The electric field E TOT (d,t) can be expressed as by using the equ 2&3,
7
By using the method of images and using the geometric figure 2, the path difference
Δ,between the LOS and ground reflected paths can be given as
When the T-R separation distance d is very large compared to ht+hr, the above equation
7 can be simplified by using taylor’s series approximation,
Fig2. Geometry for find the path difference between LOS and ground reflected paths.
Once the path difference is known, the phase difference θΔ between the two E-field components
and time delay ζd between the arrivals of the two components can be easily computed
using the following relations
8
If d becomes large, the difference between the distance d’ and d’’ becomes very small and the
amplitudes of ELOS and Eg are virtually identical and differ only in phase i.e.
If received E-field is evaluated at some time say at t=d”/c, then equ.6 becomes,
Where d-distance over flat earth between the bases of transmitter and receiver antennas. The
below phasor diagram 3 represents how the direct and ground reflected rays combine the
electric field at a distance d from the transmitter can be written
as,
The above equ 15 provides the exact received E-field for the two ray ground reflection model.For
increasing distance form transmitter ETOT decays in an oscillatory fashion, with local maxima
being 6dB greater than free space value and local maxima plummeting to -∞ dB.The equ.15 may
be simplified whenever sin(θΔ /2)≈ θΔ /2. It occurs when θΔ /2 is < 0.3. Using equ. 8&9
9
and equation 15 simplified
Thus as long as d satisfies the above equ,17 then received E-field is given by,
K-constant related to Eo, antenna heights and wavelength λ. This asymptotical behavior is
identical for both E-field in the plane of incidence or normal to plane of incidence.
The received power at a distance d from the transmitter for the two-ray ground model can
be can be expressed as
At a large distance, the received power falls off with distance raised to fourth power
or at rate of 40 dB/decade. This is a much more rapid path loss than in free space.At large
values of d, the received power and path loss become independent of frequency.
Thus the path loss of two-ray model can be expressed in dB as,
At small T-R separation distance, equ 6 is used to calculate total E-field. When θ Δ
=Π, then is the ground appears in the first Fresnel zone between the transmitter
and receiver. The first Fresnel zone distance is a useful parameter in microcell path loss
models.
10
Advantages:
The two-ray ground reflection model is a useful propagation model that is based on geometric
options, and considers both the direct path and a ground reflected propagation path
between transmitter and receiver.
Disadvantages:
If the phase of the two paths are identical, then they add together and no fading occurs. When
the phase of the two paths differ by 180 degrees, then they cancel each other and fading
occurs.
Power delay profiles are found by averaging instantaneous power delay profile
measurements over a local area in order to determine an average small-scale power delay profile.
Figure 4.9 shows typical power delay profile plots from outdoor and indoor channels, determined
from a large number of closely sampled instantaneous profiles.
Some important parameters of Mobile Multipath Channels are:
1. Time Dispersion Parameters
2. Coherence Bandwidth
3. Doppler Spread and Coherence Time
11
The mean excess delay is the first moment of the power delay profile and is defined to be,
.
The rms delay spread is the square root of the second central moment of the power delay
profile and is defined to be,
12
These delays are measured relative to the first detectable signal arriving at the receiver at
= 0. Equations 1 - 3 do not rely on the absolute power level of , but only the relative
amplitudes of the multipath components within .
It is important to note that the rms delay spread and mean excess delay are defined from a
single power delay profile which is the temporal or spatial average of consecutive impulse
response measurements collected and averaged over a local area. The maximum excess delay (X
dh) of the power delay profile is defined to be the time delay during which multipath energy falls
to X dB below the maximum and it is given by,
where is the first arriving signal and is the maximum delay at which a multipath
component is within X dB of the strongest arriving multipath signal.
Figure 4.10 illustrates the computation of the maximum excess delay for multipath components
within 10 dB of the maximum. The maximum excess delay (X dB) defines the temporal extent of
the multipath that is above a particular threshold. The value of is sometimes called the excess
delay spread of a power delay profile, but in all cases must be specified with a threshold that
relates the multipath noise floor to the maximum received multi-path component.
13
Coherence Bandwidth
Coherence bandwidth is a statistical measure of the range of frequencies over which the
channel can be considered "flat" (i.e., a channel which passes all spectral components with
approximately equal gain and linear phase).It is a defined relation derived from the rms delay
spread. If the coherence bandwidth is defined as the bandwidth over which the frequency
correlation function is above 0.9, then the coherence bandwidth is approximately.
If the definition is relaxed so that the frequency correlation function is above 0.5. then the
coherence bandwidth is approximately
14
It is important to note that an exact relationship between coherence band-width and rms delay
spread does not exist, and equations 5 and 6 are "ball park estimates". Thus coherence bandwidth
depends on frequency correlation function.
Delay spread and coherence bandwidth are parameters which describe the time
dispersive nature of the channel in a local area. But the time varying nature of the channel in a
small-scale region is described by the parameter Doppler spread and coherence time.
Doppler spread B D is defined as the set of frequencies over which the Doppler
spectrum at the receiver end is essentially non-zero. When a pure sinusoidal tone of frequency
fc is transmitted, the received signal spectrum, called the Doppler spectrum, will have
components in the range fc - fd to fc + fd , where fd is the Doppler shift. The amount of
spectral broadening depends on fd . If the baseband signal bandwidth is much greater than BD,
the effects of Doppler spread are negligible at the receiver. This is a slow fading channel.
The Doppler spread and coherence time are inversely proportional to one another. That is,
Coherence time is actually a statistical measure of the time duration over which the channel
impulse response is essentially invariant, and quantifies the similarity of the channel response at
different times. If the coherence time is denned as the time over which the time correlation
function is above 0.5, then the coherence time is approximately
Where, f m is the maximum Doppler shift given by f m = v/λ. The term v refers to velocity and
λ refers to wavelength .The quality of received signal thus depends on coherence time factor.
15
Coherence Bandwidth
Coherence bandwidth is a statistical measure of the range of frequencies over which the
channel can be considered "flat" (i.e., a channel which passes all spectral components with
approximately equal gain and linear phase).It is a defined relation derived from the rms delay
spread. If the coherence bandwidth is defined as the bandwidth over which the frequency
correlation function is above 0.9, then the coherence bandwidth is approximately.
If the definition is relaxed so that the frequency correlation function is above 0.5. then the
coherence bandwidth is approximately
It is important to note that an exact relationship between coherence band-width and rms delay
spread does not exist, and the above equations are "ball park estimates". Thus coherence
bandwidth depends on frequency correlation function.
16
Small-Scale Fading
(Based on multipath time delay spread)
Flat fading:
If the mobile radio channel has a constant gain and linear phase response over a
bandwidth which is greater than the bandwidth of the transmitted signal, then the received signal
will undergo flat fading. In flat fading, the multipath structure of the channel is such that the
spectral characteristics of the transmitted signal are preserved at the receiver. However the
strength of the received signal changes with time, due to fluctuations in the gain of the channel
caused by multipath. The characteristics of a flat fading channel are illustrated in Figure 4.12
17
It can be seen from Figure 4.12 that 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, but the
spectrum of the transmission is preserved.
Flat fading channels are also known as amplitude varying channels and are sometimes referred to
as narrowband channels, since the bandwidth of the applied signal is narrow as compared to the
channel flat fading bandwidth.
The distribution of the instantaneous gain of flat fading channels is important for designing radio
links, and the most common amplitude distribution is the Rayleigh distribution.
and Bc are the rms delay spread and coherence bandwidth respectively.
When this occurs, the received signal includes multiple versions of the
transmitted waveform which are attenuated (faded) 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 induces intersymbol interference
(ISI).Frequency selective fading channels are much more difficult to model than flat fading
channels due
18
to the following reasons,
• Each Multipath signal must be modeled and
• The channel must be considered to be a linear filter
The statistical impulse response models such as the 2-ray Rayleigh fading model or computer
generated or measured impulse responses, are generally used for analyzing frequency selective
small-scale fading.
Figure 4.13 illustrates the characteristics of a frequency selective fading channel. In frequency
selective fading gain is different for different frequency components but the channel becomes
frequency selective when Viewed in the frequency domain
Frequency selective fading is caused by multipath delays' which approach or exceed the
symbol period of the transmitted symbol. Frequency selective fading channels are also known as
wideband channels since the bandwidth of the signal s { t ) is wider than the bandwidth of the
channel impulse response.
To summarize, a signal undergoes frequency selective fading if
and
19
Fading effects due to Dopper Spread
Fast Fading
When comparing the rate of change of the radio channel the baseband signal transmitted
will change .Depending on how fast it takes place a channel may be classified either as a fast
fading or slow fading channel.
In a fast fading channel, the channel impulse response changes rapidly within the
symbol duration. That is, the coherence time of the channel is smaller than the symbol period of
the transmitted signal. Therefore, a signal undergoes fast fading if
T S >T C and
Bs < BD
It should be noted that when a channel is specified as a fast or slow fading channel, it
does not specify whether the channel is flat fading or frequency selective in nature. Fast fading
only deals with the rate of change of the channel due to motion.
In the case of the flat fading channel, we can approximate the impulse response to be
simply a delta function (no time delay). Hence, a fiat fading, fast fading channel is a channel in
which the amplitude of the delta function varies faster than the rate of change of the transmitted
baseband signal. In the case of a frequency selective, fast fading channel, the amplitudes, phases,
and time delays of any one of the multipath components vary faster than the rate of change of the
transmitted signal.
20
Slow Fading
In a slow fading channel, the channel impulse response changes at a rate much slower
than the transmitted baseband signal s(t). In this case, the channel may be assumed to be static
over one or several reciprocal bandwidth intervals. Therefore, a signal undergoes slow fading if
and
It should be clear that the velocity of the mobile (or velocity of objects in the channel)
and the baseband signaling determines whether a signal undergoes fast fading or slow
fading.The relation between the various multipath parameters and the type of fading
experienced by the signal are summarized in Figure 4.14.
21
UNIT II
CELLULAR ARCHITECTURE
Frequency Division Multiple Access (FDMA)
FDMA describes scheme to sub divide the bandwidth into several non-over lapping
frequecny bands.FDMA assigns individual channls to individual users each user allocates a
unique frequency band or channelIn FDMA the users are assigned a channles as a pair of
frequecies. One is forward channel and other for reverse channel.
FDMA is distinct from frequency division duplexing (FDD). While FDMA allows
multiple users simultaneous access to a transmission system, FDD refers to how the radio
channel is shared between the uplink and downlink (for instance, the traffic going back and forth
between a mobile-phone and a mobile phone base station).
22
(x) FDMA requires high-performing filters in the radio hardware, in contrast to TDMA
and CDMA.
(xi) FDMA is not vulnerable to the timing problems that TDMA has. Since a
predetermined frequency band is available for the entire period of
communication, stream data (a continuous flow of data that may not be
packetized) can easily be used with FDMA.
(xii) Due to the frequency filtering, FDMA is not sensitive to near-far problem
which is pronounced for CDMA.
(xiii) Each user transmits and receives at different frequencies as each user gets a unique
frequency slots.
TDMA system divides the radio spectrum into time slot and in each slot only one user is allowed
to either transmit or to receive. TDMA system transmit data in buffer and burst method thus the
transmission of any user is not continous.
Time division multiple access (TDMA) is a channel access method for shared medium
networks. It allows several users to share the same
23
frequency channel by dividing the signal into different time slots. The users transmit in rapid
succession, one after the other, each using its own time slot. This allows multiple stations to
share the same transmission medium (e.g. radio frequency channel) while using only a part of its
channel capacity.
TDMA is used in the digital 2Gcellular systems such as Global System for Mobile
Communications (GSM), IS-136, Personal Digital Cellular (PDC) and in the Digital Enhanced
Cordless Telecommunications (DECT) standard for portable phones. It is also used extensively in
satellite systems, combat-net radio systems, and PON networks for upstream traffic from
premises to the operator. For usage of Dynamic TDMA packet mode communication, given
below in fig.
TDMA frame structure showing a data stream divided into frames and those frames divided into
time slots.TDMA is a type of Time-division multiplexing, with the special point that instead
of having one transmitter connected to one receiver, there are multiple transmitters. In the case of
the uplink from a mobile phone to a base station this becomes particularly difficult because the
mobile phone can move around and vary the timing advance required to make its transmission
match the gap in transmission from its peers.
24
Features of TDMA are as follows:
(i) TDMA stores a single carrier frequency with several users without overlapping
of time slots
(ii) Data transmission for users of a TDMA system is not continous but occurs in
buffer and brust . this results in low battery consumption.
(iii) TDMA uses different time slot for transmission and reception thus
duplexers are not required
(iv) High synchronization overhead is required in TDMA systems in brust
transmission
(v) TDMA has an advantage in that it is possible to allocate different no of time
slots per frame to different users. Thus bandwidth can be supplied on demand
to different users
25
Code division multiple access(CDMA):
In CDMA system, a narrow band message signal in multipiled by a large bandwidth signal called
spreading signal .The spreading signal is PSEUDO RANDOM sequence that has chip rate is
much higher than the data rate of the message signal. All the users in CDMA use the same
channel frequency and may transmit simultanously.
Each and every users has its own PSEUDO RANDOM code which is orthogonal to all
other user’s code. The near far problem occurs when many mobile users share the same
channel stronger received signal acts as noise for the weaker signal. There by decreasing the
probability that weaker signal will be received to combat near far problem, power control is
used in CDMA system.
Power control is implemented at the base station by rapidly sampling the radio signal strengh
indicator (RSSI)level of each mobile and then sending a power change command over
forward link Features of CDMA are as follows:Many users of CDMA system share the same
frequency Unlike in TDMA and FDMA, CDMA has soft capacity limit there is no absoulte limit
on no of users
26
in CDMA.RAKE RECEIVER can be used to improve reception by collecting time delayed
versions of the desired signal. The near far problem occurs at a CDMA receiver if an undesired
user has a high detected power as compared to desired user.
Frequency reuse:
Cellular radio systems rely on an intelligent allocation and reuse of channels throughout a
coverage region. Each cellular base station is allocated a group of radio channels to be used
within a small geographic area called a cell.
The base station antennas are designed to achieve the desired coverage within the particular cell.
By limiting the coverage area to within the boundaries of a cell, the same group of channels
may be used to cover different cells that are separated from one
27
another by distances. The design process of selecting and allocating channel groups for all of the
cellular base stations within a system is called frequency reuse or frequency planning.
Figure 2.1 illustrates the concept of cellular frequency reuse, where cells labeled with the
same letter use the same group of channels.
Figure 2.1 is conceptual and is a simplistic model of the radio coverage for each base station, but
it has been universally adopted since the hexagon permits easy and manageable analysis of a
cellular system. The actual radio coverage of a cell is known as the footprint and is determined
from field measurements or propagation prediction models.
To understand the frequency reuse concept, consider a cellular system which has a total of S
duplex channels available for use. If each cell is allocated a group of k channels (k < S), and if
the S channels are divided among N cells into unique and
28
disjoint channel groups which each have the same number of channels, the total number
of available radio channels can be expressed as
S = kN
The N cells which collectively use the complete set of available frequencies is called a cluster. If
a cluster is replicated M times within the system, the total number of duplex channels, C, can be
used as a measure of capacity and is given
C = MkN = MS
As seen from equation 2, 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 duster size and is
typically equal to 4, 7, or 12.
The frequency reuse factor of a cellular system is given by l/N, since each cell within a cluster is
only assigned I /N of the total available channels in the system. To connect without gaps between
adjacent cells the geometry of hexagons is such that the number of cells per cluster, N can only
have values which satisfy equation .
where i and j are non-negative integers. To find the nearest co-channel neighbors of a particular
cell, one must do the following (1) move i cells along any chain of hexagons and then
29
• Dynamic channel assignment strategy.
The choice of channel assignment strategy impacts the performance of the system, particularly as
to how calls are managed when a mobile user is handed off from one cell to another.
Several variations of the fixed assignment strategy exist. In one approach, called the
borrowing strategy, a cell is allowed to borrow channels from a neighboring cell if all of its own
channels are already occupied. The mobile switching center (MSC) supervises such
borrowing procedures and ensures that the borrowing of a channel does not disrupt or interfere
with any of the calls in progress in the donor cell.
In a dynamic channel assignment strategy, voice channels are not allocated to different
cells permanently. Instead, each time a call request is made, the serving base station requests a
channel from the MSC. The switch then allocates a channel to the requested cell following an
algorithm that takes into account the likelihood of future blocking within the cell, the
frequency of use of the candidate channel, the reuse distance of the channel, and other cost
functions.
Dynamic channel assignment reduce the likelihood of blocking, which increases the
trunk-ing capacity of the system, since all the available channels in a market are acces-sible to all
of the cells. Dynamic channel assignment strategies require the MSCDynamic channel
assignment reduce the likelihood of blocking, which increases
30
the trunk-ing capacity of the system, since all the available channels in a market are acces-sible
to all of the cells. Dynamic channel assignment strategies require the MSC.
Handoff strategies
When a mobile station moves from one cell site to another cell site while crossing boundary
frequency switch over from one base station to another base station without affecting the call
which is in progress
This hand off not only involves identifying the new base station but also requires that the voice
and control signals be allocated to channel associated with in the new base stationOnce a
particular signal level is specified as the minimum usable signal for acceptable voice quality at
the base station receiver {-90dBm or -100dBm}.The threshold level is given by,
The ∆ cannot be too large or too small. If ∆ is too large, unnecessary handoffs which burden the MSC
may occur, and if ∆ is too small, there may be insufficient time to complete a handoff before a call is lost
due to weak signal conditions. Figure 2.2 illustrates a handoff situation. Figure
2.2(a) demonstrates the case where a handoff is not made and the signal drops below the
minimum acceptable level to keep the channel active.
This dropped call event can happen when there is an excessive delay by the MSC(Mobile
switching Center) in assigning a handoff, or when the threshold A is set too small for the handoff
time in the system. Excessive delays may occur during high traffic conditions due to
computational loading at the MSC or due to the fact that no channels are available on any
of the nearby base stations (thus forcing the MSC to wait until a channel in a nearby cell
becomes free).
The base station monitors the signal level for a certain period of time before a hand-off is initiated.
This running average measurement of signal strength should be optimized so that unnecessary
handoffs are avoided the length of time needed to decide if a handoff is necessary depends on
the speed at which the vehicle is moving.
31
Fig 2.2 Illustration of handoff strategies
The time over which a call may be maintained within a cell, without hand-off, is called the
dwell time. The dwell time of a particular user is governed by a number of factors
o Propagation,
o Interference,
o Distance between the subscriber and the base station, and other time varying
effects.
In first generation analog cellular systems, signal strength measurements are made by the base
stations and supervised by the MSC. Each base station constantly monitors the signal
strengths of all of its reverse voice channels to determine the relative location of each mobile
user with respect to the base station tower In second generation
32
systems that use digital TDMA technology, handoff decisions are mobile assisted. In mobile
assisted handoff(MAHO), every mobile station measures the received power from
surrounding base stations and continually reports the results of these measurements to the serving
base station.
The MAHO method enables the call to be handed over between base stations at a much faster
rate than in first generation analog systems since the handoff measurements are made by each
mobile, and the MSC no longer constantly monitors signal strengths. During the course of a
call, if a mobile moves from one cellular system to a different cellular system controlled by a
different MSC, an intersystem handoff becomes necessary.
An MSC engages in an intersystem handoff when a mobile signal becomes weak in a given cell
and the MSC cannot find another cell within its system to which it can transfer the call in
progress.
Prioritizing Handoffs
One method for giving priority to handoffs is called the guard channel concept. Guard channels,
however, offer efficient spectrum utilization when dynamic channel assignment
strategies, which minimize the number of required guard channels by efficient demand-based
allocation, are used.
Queuing of handoff requests is another method to decrease the probability of forced termination
of a call due to lack of available channels. There is a tradeoff between the decrease in probability
of forced termination and total carried traffic.
Practical Handoff Considerations In practical cellular systems, several problems arise when
attempting to design for a wide range of mobile velocities. Using different antenna heights
(often on the same building or tower) and different power levels, it is possible to provide
"large" and "small" cells which are co-located at a single location. This technique is called the
umbrella cell approach and is used to provide large area coverage to high speed users while
providing small area coverage to users traveling at low speeds.
33
Fig 2.3 Umbrella Cell Pattern
Figure 2.3 illustrates an umbrella cell which is co-located with some smaller microcells. The
umbrella cell approach ensures that the number of handoffs is minimized for high speed users
and provides additional microcell channels for pedestrian users.
Another practical handoff problem in microcell systems is known as cell dragging. Cell dragging
results from pedestrian users that provide a very strong signal to the base station. Such a situation
occurs in an urban environment when there is a line-of-sight (LOS) radio path between the
subscriber and the base station.
As the user travels away from the base station at a very slow speed, the average signal strength
does not decay rapidly.
34
Trunking and Grade of Service
In cellular mobile communication the two important aspects that has to be considered with
more care are,
1) Trunking
These aspects have to be well planned so that it will lead to a better system
performance.
Trunking
The 'trunking' deals with accommodation of larger number of mobile users in minimum radio
spectrum. By using this trunking concept it is possible to allow many users to share smaller
number of mobile channels in a cell. It is done by assigning channels on demand basis and
allocating a channel from a pool of channels available. That is if an user want to access a channel
for establishing a cell then from the pool of channels the required channel will be assigned to
the user.
Once the call progress is terminated at the end of the call then the channel used so far will
return to the pool and will be ready for any next new access to come. The concept finds
application in telephone circuitry, mobile radio communication in a larger way.
For designing trunked radio system that is capable of handling a particulai! capacity at a
'grade of service'. Some fundamental points regarding trunking theory is required. It was
For example, if a mobile radio channel was occupied for 15 minutes of an hourthen it is said
as,
35
Thus trunking is a main concept used to improve system efficiency. Also the term grade of
service is closely linked with representation of grade of service.
Grade of Service (GoS)
The Grade of Service (GoS) is another important measure that express the ability of mobile
user to access a trunked system mainly in the busiest hour of the day. The busiest hour of day is
statistically studied and considered as 4 pm to 6 pm on the Thursday and Friday of a week. The
busiest hour is also decided with respect to customer demand in particular hours.
The linked system's performance is defined by its grade of service. Only if the
trunked system permits its user to access if even in the busiest hour then such a trunked
system is said to be an ideal system.
To meet out an appropriate GoS, the estimation of maximum capacity required for
allocating enough number of radio channels in the design is a must. The GoS is also a measure
of congestion that is specified as the probability of delaying a call beyond a time limit. The call
request rate multiplied with holding time has to be equal to the traffic intensity which is offered
by an user. If the traffic intensity in Erlangs generated by user is Tu then,
36
Then there are two category of trunking system is available namely,
1) Trunking system with no queuing for the call requests.
Trunking system with queue provision for holding calls that are blocked. This is given
in the tree classification as shown in following Fig
The GoS parameter will be different for the various cellular systems. For example,
the AMPS cellular system is mainly designed for GoS of 2 % blocking status, which implies that
2 calls out of 100 calls allotted will be blocked.
The two broad classes of the trunked radio system is also called as Lost Call
Cleared (LCC) and Lost Call Delayed (LCD) systems.
In Lost Call Cleared (LCC) system queuing is allowed for the call requests made. If a
user wants service it will be served in case of availability of channel within a minimum
time. Otherwise the call will be blocked if there is no channel available to assign.
In the second lost call delayed system the queues are made use to hold the callrequests
which were initially blocked. In this type if a user makes a call request it will served if channel is
available or if the channel is not available at that moment the call will be delayed till the
channel becomes available. It is determined by Erlang C formula.
The grade of service in this system will express the probability of till delaying time for a
call. There is large number of users available in the second type.
Thus in trunked system there are lost call cleared (no queues used) and lost c delayed
(queues used) and the second type of trunking system is widely used.
37
Summary of some important terms related to trunking theory are as follows :
More number of channels, higher will be the coverage range (distance) in the cell thus
leading to a higher coverage capacity.
For enhancing the cellular coverage capacity there are many techniques available and
some important cellular techniques are discussed below in detail.
i) Cell splitting
ii) Cell sectoring
iii) Micro zone method
iv) Repeaters for extending range.
Cell Splitting
Cell splitting is a technique of subdividing the congested (high traffic) cell into smaller
38
sized cells. The parent cell which was originally congested is called as "macro cells" and the
smaller cells are called as micro cells". The main objective of "cell splitting process" is to
increases the cellular capacity of the system where frequency reuse technique can be efficiently
implemented.
For example, a congested cell is subdivided into smaller cells shown in Fig. 1.31. Each smaller
(micro) cell has a base station antenna exclusively and the micro cell radius will be half the
radius value of the macro cell
The transmit power (P t ) will be less for the micro cells. Assuming P r _ 0 as the received power
at old cell boundary and P r _ N as the received power at new cell boundary.
In cell splitting process generally the larger are dedicated to high speed traffic.
The reason for this is the number of 'hand offs' will be less in larger cell and call
progress will be smoothly continued in larger cells.
Also the channels in old cells have to be broken into two groups due to following points. i)
If larger transmit power is used for all the available cells then some of the channels
used by smaller cells may not be completely separated by co-channel cells. This may
lead to interference.
ii) In case if smaller transmit power is used for the available cells then there is chance of
'unserved' problem. That is some parts of the larger cells would be left out as
'unserved'. This is also not acceptable.
Hence the channels of the macro/larger cell has to be divided into two groups. The larger
cells for high speed traffic and micro/smaller cells for low speed traffic regions. Antenna down
tilting :
The process of an antenna down tilting is done mainly to focus the energy radiated from the Base
Station (BS) towards the ground and not towards the horizon so that radio coverage of new micro
cells will be properly limited.
Cell Sectoring
39
The cell sectoring is again a technique to increase the capacity but it keeps the radius of cell as
constant. The size of clusters in cellular region may be reduced because the cell sectoring
increases the Signal to Interference Ratio (SIR) value.
The method of decreasing the co-channel interference value and enhancing thel system
performance by using the directional antennas is known as "cell sectoring".
Sectors :
A cell in the cellular region is generally divided into 120° sectors or 60° sectors. If the sectoring
is 120° a cell of hexagon type consists of three (3) sectors and if the sectoring is 60° sectoring the
hexagonal type cell consists of six (6) sectors as shown in Fig. 1.32.
If cell sectoring is employed then the channel used in a cell will be divided into groups i.e.
called as sectored groups and they are used only within a sector.
For example, in a seven cell reuse pattern with 120° sectoring, the possible number of
interferers in the first tier will be only two. It means that only two cells of the six co-channel
cells get interfered and it is better to apply cell sectoring in the design aspects of mobile
communication to increase cellular capacity with less interference.
40
One of the problem associated with cell sectoring is that "requirement of more number of
handoffs" thereby increased load status at the switching and control link components of the
cellular mobile system.
There is another new concept known as 'micro zone concept' to minimize this problem.
In Fig. 1.33 there are three zones 1 to 3 shown with T/R set up. But they are connected to a single
Base Station (BS) so that they are sharing the same equipment. For establishing connection
between these three zones with the common base station microwave link, coaxial cable or fiber
optic cable are used. Such an arrangement of several cellular zones with one base station
constitutes a cell.
41
Within the cell when a mobile user roams from one zone say 1 to another zone 2 then zone 1
will have strongest signal with respect to the base station.0 In this micro cell zone concept
the antennas are placed at the edges of each zone such that when the user moves from one
zone to another zone the signal strength does not reduce like other methods. The number of
hand offs are less when compared to cell sectoring method when a cell is in progress.The
merits of this technique are listed below. Advantages of micro zone cell concept :
i) When the mobile user travels from one zone to another zone within the same, cell the
same channel is still maintained for the call progress.
ii) Since low power transmitters are used in each zone apart from control base station
the effect of interference is highly reduced.
iii) Improved signal quality is possible.
iv) Reduced number of hand offs when a call is in progress
42
Unit-III
Digital Signaling For Fading Channels
Wireless communication link with the neat block diagram for transmitter and
receiver
• The information source provides an analog source signal and feeds it into the source
ADC and then converts the signal into a stream of digital data
• The source coder uses a priori information on the properties of the source data in order to
reduce redundancy in the source signal.Eg:(GSM) speech coder reduces the source
data rate from 64 kbit/s mentionedabove to 13 kbit/s.
• The channel coder adds redundancy in order to protect data against transmission errors.
This increases the data rate that has to be transmitted at interface E – e.g., GSM
channel coding increases the data rate from 13 to 22.8 kbit/s.
• Signalingadds control information for the establishing and ending of connections, for
associating information with the correct users, synchronization, etc.
43
• The multiplexer combines user data and signaling information, and combines the data
from multiple users.
• In GSM, multiaccess multiplexing increases the data rate from 22.8 to 182.4 kbit/s
• The baseband modulator assigns the gross data bits Spectral properties,
intersymbol interference, peak to-average ratio, and other properties of the transmit signal
are determined by this step.
• The TX Digital to Analog Converter (DAC) generates a pair of analog, discrete
amplitude voltages corresponding to the real and imaginary part of the transmit symbols.
• The analog low-pass filter in the TX eliminates the (inevitable) spectral
components outside the desired transmission bandwidth.
• The TX Local Oscillator (LO) provides an unmodulated sinusoidal signal,
corresponding to one of the admissible center frequencies.
• The upconverterconverts the analog, filtered baseband signal to a passband signal
by mixing it with the LO signal.
• The RF TX filter eliminates out-of-band emissions in the RF domain.
• The (analog) propagation channel attenuates the signal, and leads to delay and
frequency dispersion. The environment adds noise (Additive White Gaussian Noise –
AWGN) and co-channel interference
44
Fig:Block diagram of a digital receiver chain for mobile communications
45
• The baseband demodulator obtains soft-decision data from digitized baseband data,
and hands them over to the decoder.
• multiple antennas, then the RX either selects the signal from one of them for further
processing or the signals from all of the antennas have to be processed
• Symbol-timing recovery uses demodulated data to determine an estimate of the duration
of symbols,and uses it to fine-tune sampling intervals.
• The decoder uses soft estimates from the demodulator to find the original (digital)
source data.
• Signaling recovery identifies the parts of the data that represent signaling
information and controls the subsequent demultiplexer.
• The demultiplexer separates the user data and signaling information and reverses
possible time compression of the TX multiplexer.
OQPSK signaling is similar to QPSK signaling is except for the time alignment of the even
and odd bit streams. In QPSK signaling, the bit transitions of the even and odd bit streams
occur at the same time instants, but in OQPSK signaling, the even and odd bit streams, mI
(t) and m Q (t) are offset in their relative alignment by one bit period.
46
Fig.3.1 Offset waveforms
Due to the time alignment of m l ( t) and mq ( t) in standard QPSK, phase transitions occur only
once every T s = 2T b s, and will be a maximum of 180° if there is a change in the value of
both m l ( t) and m q ( t ) . However, in OQPSK signaling, bit transitions phase transitions)
occur every T b s. Since the transitions instants are offset m l ( t) and mq ( t ) , 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 ±90°. Hence, by switching phases more
frequently (i.e., every T b s instead of 2T b s) OQPSK signaling eliminates 180° phase
transitions.Since 180° Phase transitions have been eliminated, bandlimiting of QPSK signal does
not cause the signal envelope to go to zero. Ultimately some amount of ISI caused by the
bandlimitng process expecially at the 90° phase transition points.
But the envelope variations are considerably less, and hence hardlimiting or non linear
amplification of OQPSK signals does not generate the high frequency sidelobes as much as in
QPSK. The spectral efficiency is significantly reduced, while permitting over efficient RF
bandwidth.
47
The spectrum of an OQPSK signal is identical to that of a QPSK signal, hence both signals occupy
the same bandwidth. The staggered alignment of the even and odd bit streams does not change the
nature of the spectrum. OQPSK retains its bandlimited nature even after nonlinear amplification,
and therefore is very attractive for mobile communication systems where bandwidth efficiency
and efficient nonlinear amplifiers are critical for low power drain. Further, OQPSK signals is to
perform better than QPSK in the presence of phase jitter due to noisy reference signals at the
receiver.
Minimum shift keying (MSK) Derive an expression for MSK and its power spectrum.
Minimum shift keying:
MSK is a special type of continuous phase frequency shift keying (CPFSK). Where the
peak frequency deviation is equal to ¼ the bit rate. In other words, MSK is a continuous phase
FSK with a modulation index of 0.5 reference signal.
The modulation index of FSK signal is similar to FM modulation index, and is defined as
K FSK =(2∆F)/R b..
MSK is sometimes referred as fast FSK, as the frequency spacing used is only half as
much as that used in conventional non-coherent FSK.
48
MSK is spectrally efficient modulation and its particularly attractive for using mobile radio
communication. It possesses properties such as constant envelope, spectral efficiency, good BER
performance, and self-synchronizing capability.
MSK signal is special form of OQPSK where the baseband rectangular pulses are replaced
with half-sinusoidal pulses. If half-sinusoidal pulses are used instead of rectangular pulses, the
modified signal can be defined as MSK and for an N-bit stream is given by
where mI (t) and m Q ( t ) are the "odd" and "even" bits of the bipolar data stream are values of
±1 and which feed the in-phase and quadrature arms of the modulator at a rate of R b /2. There are
a number of variations of MSK. For example, while one version of MSK uses only positive half-
sinusoids as the basic pulse shape, another version uses alternating positive and negative half-
sinusoids as the basic pulse shape.
However, all variations of MSK are continuous phase FSK employing different techniques
to achieve spectral efficiency.
The MSK waveform is a special type of a continuous phase FSK if equation 2 is
rewritten by using trigonometric identities as
49
MSK Power Spectrum
For MSK the baseband pulse shaping function is given by,
Figure 3.2 shows the power spectral density of an MSK signal. The spectral density of both
QPSK and OQPSK are also drawn for comparison in Figure 3.2, the MSK spectrum has
lower side lobes than QPSK and OQPSK. Ninety-nine percent of the MSK power is contained
within a bandwidth B = 1.2 / T , while for QPSK and OQPSK, the 99 percent bandwidth B is
equal to 8/T. Figure 3.2 also shows that the main lobe of MSK is wider than QPSK and OQPSK,
and hence when compared in terms of first null bandwidth, MSK is less spectrally efficient than
the phase-shift keying techniques.
50
MSK (Minimum shift keying) modulation and demodulation technique.
MSK Transmitter:
MSK receiver:
The block diagram of an MSK receiver is shown in Figure 3.4. The received signal
s M S K { 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 discarded to a decision circuit at the end of each two bit periods. Based on the
level of the signal at the output of the integrator, the threshold detector decides whether the
signal is a 0 or a 1. The output data streams
correspond to m I ( t ) and m Q ( t ) .
51
Fig.3.4 Block diagram of MSK Receiver.
Premodulation Gaussian filtering converts the full response message signal into a partial
response scheme where each transmitted symbol spans several bit periods. GMSK can be
coherently detected just as an MSK signal, or non-coherently detected as simple FSK. GMSK is
most attractive for its excellent power efficiency (due to the constant envelope) and its excellent
spectral efficiency.
The pre modulation Gaussian filtering introduces ISI (Inter Symbol Interference) in the
transmitted signal, but it can be shown that the degradation is not severe if the 3dB bandwidth bit
duration (BT) product of the filter is greater than 0.5.
52
In GMSK pre modulation filter bas an impulse response given by
GMSK filter may be completely defined from B and the baseband symbol duration T . Therefore
GMSK its B T product
Figure 3.5 shows the simulated RF power spectrum of the GMSK signal for various values of B
T . The power spectrum of MSK, which is equivalent to GMSK with a B T product of infinity, is
also shown for comparison purposes. In this graph that as the B T product decreases, the side
lobe levels fall off very rapidly.
For example, for a B T =0.5, the peak of the second lobe is more than 30dB below the main lobe,
whereas for simple MSK, the second lobe is only 20 dB below main lobe. However, reducing B
T increases the irreducible error rate produced by the low pass filter due to
ISI. GMSK Bit Error Rate
The bit error rate for GMSK was determined for AWGN channels, and performance
within 1 dB of optimum MSK when BT=0.25. The bit error probability is a function of BT, and
the probability for GMSK is given by
53
GMSK TRANSMITTER:
The simplest way to generate a GMSK signal is to pass a NRZ message bit stream through a
Gaussian baseband filter having an impulse response given by equ 1 followed by an FM
modulator. This modulation technique is shown in Figure 3.6 and is currently used in a variety of
analog and digital implementations for the U.S. Cellular Digital Packet Data (CDPD) system as
well as for the Global System for Mobile (GSM) system.
Figure 3.6 may also be implemented digitally using a standard I/Q modulator.
GMSK RECEIVER
GMSK signals can be detected using orthogonal coherent detectors as shown in Figure
3.7, or with simple non coherent detectors such as standard FM discriminators. The Carrier
recovery is performed by using the sum of two discrete frequency components contained at the
output of a frequency doubler is divided by four.
54
Fig.3.7 Block diagram of GMSK receiver
This type of receiver can be easily implemented using digital logic as shown in Figure
3.8. The two D flip-flops act as a quadrature product demodulator and the XOR gates act as
baseband multipliers. The mutually orthogonal reference carriers are generated using two D
flip-flops, and the VCO center frequency is set equal to four times the carrier center
frequency.
55
Offset Quadrature Phase Shift Keying signal and its advantages.
• Offset QPSK is a modified form of QPSK where the bit waveforms on the I and Q
channels are offset or shifted in phase from each other by one-half of a bit time
• The advantage of OQPSK is the limited phase shift that must be imparted during
modulation and it supports more efficient amplification.
• OQPSK signaling eliminates the 180 degree phase transitions. With 90 degree phase
transitions it has some ISI.
• The spectrum of the OQPSK is identical to QPSK and both signals occupies same
bandwidth.
• The disadvantages is that changes in the output phase occur at twice the data rate in
either the I and Q channels.
• OQPSK signaling is similar to QPSK signaling is except for the time alignment of the
even and odd bit streams.
• To improving the peak-to-average ratio in QPSK.
56
Fig: Difference b/w QPSK and OQPSK
• The principle of π/4-DQPSK can be understood from the signal space diagram of
DQPSK .
• There exist two sets of signal constellations: (0, 90, 180, 270◦) and (45, 135, 225,
315◦). All symbols with an even temporal index i are chosen from the first set, while
all symbols with odd index are chosen from the second set.
• The duration of the dips is longer when non-rectangular basis pulses are used.
Such variations of the signal envelope are undesirable. One possibility for
reducing these problems lies in the use of π/4-DQPSK
57
Fig: QAM Pulse Sequence
58
Principles of OFDM and note on cyclic prefix.
Main idea: split data stream into N parallel streams of reduced data rate and transmit each on
a separate subcarrier.OFDM modulation is equivalent to the IDFT
• OFDM Mechanism:
• Parallel Data Streams
• The available frequency spectrum is divided into several sub-channels
• low-rate bit stream is transmitted over one sub-channel by modulating a sub- carrier
using a standard modulation scheme, for example: 4-QAM
59
OFDM Power Spectrum:
0.8
0.6
OFDM Transmitter :
An OFDM carrier signal is the sum of a number of orthogonal sub-carriers, with base band data
on each sub-carrier being independently modulated commonly using some type of quadrature
amplitude modulation (QAM) or phase-shift keying (PSK).
60
OFDM Receiver:
The receiver picks up the signal r(t), which is then quadrature-mixed down to baseband using
cosine and sine waves at the carrier frequency. This returns N parallel streams, each of which is
converted to a binary stream using an appropriate symbol detector. These streams are then re-
combined into a serial stream, which is an estimate of the original binary stream at the
transmitter.
CYCLIC PREFIX:
• Zeros used in the guard time can alleviate interference (ISI) between OFDM
symbols
• Orthogonality of carriers is lost when multipath channels are involved.
OFDM APPLICATIONS:
61
• ADSL
• Wireless LANs
• IFFT/FFT operation ensures that sub-carriers do not interfere with each other.
• Information from the affected subchannels can be erased and recovered by the forward
error correction (FEC) codes.
• Equalization is very simple compared to Single-Carrier systems
• Cyclic prefix allows the receiver to capture multi- path energy more efficiently.
OFDM DRAWBACKS:
Peak to Average Power Ratio (PAPR) and the methods used in PAPR reduction.
• OFDM has been chosen for high data rate communications and has been widely
deployed in many wireless communication standards such as Digital Video Broadcasting
(DVB) and based mobile worldwide.
• One of the major problems of OFDM is that the peak amplitude of the emitted signal
can be considerably higher than the average amplitude.
62
• Due to the large number of sub carriers, OFDM systems have a large dynamic signal
range with a very high PAPR.
• As a result, the OFDM signal will be clipped when passed through a non linear power
amplifier at the transmitter end.
There are three main methods to deal with the Peak-to-Average Power Ratio (PAPR):
1. Put a power amplifier into the transmitter that can amplify linearly up to the possible
peak value of the transmit signal. This is usually not practical, as it requires expensive and
power-consuming.
2. Use a nonlinear amplifier, and accept the fact that amplifier characteristics will lead to
distortions in the output signal.
• Those nonlinear distortions destroy orthogonality between subcarriers, and also lead to
increased out-of-band emissions.
63
(ii) select the associated codeword of length N;
• Furthermore, the correction function acts as additional pseudo noise, and thus
increases the BER of the syste
64
Unit-IV
Equalization:
Fundamentals of Equalisation
Since the mobile fading channels are random and time varying the equalizer must be
adaptive in nature.General operating modes of adaptive equalizer is training and tracking.A
known fixed length training sequence is sent by the transmitter so that the receiver equalizer
adapt to proper BER detection.Immediately following training sequence the message data is
sent.At the receiver tracking(recursive algorithm) is used to evaluate the channel and filter
coefficients
65
66
Reconstructed message data dt that you saw last time is an estimate and should match x(t)f we use
dt and use it in a feedback loop so adjust the weights of the equalizer we get into the domain of
non-linear equalizer.Feedback is not there we just have a feed forward loop then it is a linear
equalizer.So we realize that equalizer is in fact an inverse filter of the channel but in the absence
of noise.
Types of equalization
Linear equalizers
Non-linear equalizers
Equalization compensates for Intersymbol interference (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 into adjacent symbols. equalizers
must be adaptive since the channel is generally unknown and time varying.
67
Linear equalizers:
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.
This implementation is usually carried out in the digital domain where the samples of the
received signals are stored in a shift register
Where ,
The values N1 and N2 denote the number of taps are used in the forward and
reverse portions of the equalizer.
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Fig.1 Structure of linear transversal equalizer
The minimum mean squared error E[|e(n)|²] that a linear transversal equalizer can achieve
------>2
The linear equalizer can also be implemented as a lattice filter, whose structure is shown
in fig 2. In lattice filter, the input signal y k is transformed into a set of N intermediate forward
and backward error signals, fn (k) and b n (k) respectively, which are used as inputs to the tap
multipliers and are used to calculate the updated coefficients. Each stage of the lattice is
determined by following recursive equation.
69
The backward error signals b n are used as inputs to the tap weights and the output of the
equalizer is given by
Numerical stability
faster convergence.
The unique structure of the lattice filter allows the dynamic assignment of the most effective
length of the lattice equalizer. Hence, if the channel is not very time dispersive, only a fraction
of the stages are used. When the channel becomes more time dispersive, the length of
the equalizer can be increased by the algorithm without stopping the operation of the
equalizer.
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(ii) Non-linear equalizers:
Nonlinear equalizers are used in applications where the channel distortion is too severe for a
linear equalizer to handle. So, Linear equalizers do not perform well on channels which have
deep spectral nulls in the passband. In an attempt to compensate for the distortion, the linear
equalizer places too much gain in the vicinity of the spectral null, thereby enhancing the noise
present in those frequencies.Three very effective nonlinear methods have been developed which
offer improvements over linear equalization techniques and used mostly in 2G and 3G wireless
applications.
The basic idea behind decision feedback equalization is that once an information symbol
has been detected and decided upon, the ISI 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 form is shown in fig3. It consists of
a feed forwarded 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 ISI on the current
symbol from past detected symbols. The equalizers has N1+ N2+1 taps in the feed forward
filter and N3 taps in the feedback
filter and its output can be expressed as,
------>1
Where,
C n * and Y n are tap gains and the inputs respectively to the forward filter, F i *are tap gains
for the feedback filter and d i (i<k) is the previous decision made on the detected signal.
The minimum mean squared error a DFE can achieve is
----->
71
The minimum MSE for a DFE in above equation is always smaller than that of an LTE in
equation 2 of linear equalizer unless F(ejωT) is constant. If there are nulls in F(ejωT) a
DFE has significantly smaller minimum MSE than an LTE.
Therefore, an LTE is well behaved when the channel spectrum is comparatively flat, but if the
channel is severely distorted or exhibits nulls in the spectrum, the performance of an LTE
deteriorates and the mean squared error of a DFE is much better than a LTE. Also, an LTE has
difficulty to equalizing a non minimum phase channel, where the strongest energy arrives after
the first arriving signal component. 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.The other name is shown in
Figure 4. It also consists of a feed forward filter of (FFF) as in the conventional DFE. However,
the feedback filter DFE is called a predictive DFE. (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 is called a noise predictor because
72
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 MSE-based linear equalizers are optimum with respect to the measure of minimum
probability of symbol error when the channel does not introduce any amplitude distortion.
However this is precisely the condition in which an equalizer is needed for a mobile
communications link. 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 sequences (rather than decoding each received symbol by itself), and chooses the data
sequence with the maximum probability as the output. An MLSE usually has a large
computational requirement, especially when the delay spread of the channel is large. Using the
MLSE as an equalizer to predict the MLSE estimator by using the Viterbi algorithm. It has
recently been implemented successfully for equalizers in mobile radio channels.
The MLSE can be viewed as a problem to estimating the state of a discrete-time finite state
machine, which in this case happens to be the radio channel 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. Thus the channel has ML states, where M is the
73
size of the symbol alphabet of the modulation. That is, an ML trellis is used by the receiver to
model the channel over time. The Viterbi algorithm tracks the state of the channel by the paths
through the trellis and gives at stage k a rank ordering of the most probable sequences terminating
in the most recent L symbols.
The block diagram of a MLSE receiver based on the DFE is shown in Figure 5. The MLSE is
optimal to 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
corrupting the signal.
Thus, the probability distribution of the noise determines the form of the metric for optimum
demodulation of the received signal.
In a zero forcing equalizer, the equalizer coefficientsC n 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 the number of coefficients increase without
bound, an infinite length equalizer with zero ISI at the output can be obtained.
When each of thedelay 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
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symbol rate 1/T. The combined response of the channel with the equalizer must satisfy
Nyquist's first criterion.
------->1
whereH ch (f) is the folded frequency response of the channel. Thus, an infinite length, zero,
ISI equalizer is simply an inverse filter which inverts the folded frequency response of the
channel. This infinite length equalizer is usually implemented by a truncated length version.The
zero forcing algorithm was developed for wire line communication.
The zero forcing equalizer has the disadvantage that the inverse filter may excessively amplify
noise at frequencies where the folded channel spectrum has high attenuation. The ZF equalizer
neglects the effect of noise altogether, and is not often used for wireless links.
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.
---->1.
To compute the mean square error \ek\2at time instant k , equation becomes
For a specific channel condition, the prediction error e k is dependent on the tap gain vector
w N , so the MSE of an equalizer is a function of w N . Let the function J (wN) denote he
mean squared error as a function of tap gain vector w N
.
75
The above equ. is called the normal equation, since the error is minimized and is made
orthogonal. When above equation is satisfied, the MMSE of the equalizer is
To obtain the optimal tap gain vector , equation in (5a) must be solved iteratively as the
equalizer converges to an acceptably small value of J opt . The minimization of the MSE is
carried out recursively by use of the stochastic gradient algorithm. This is more commonly called
the Least Mean Square (LMS) algorithm. LMS is computed iteratively,
whereN 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 maximizes the signal to distortion ratio at 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.
Principle of diversity
76
Diversity exploits the random nature of radio propagation by finding independent (or at least
highly uncorrected) signal paths for communication. In virtually all applications, diversity
decisions are made by the receiver, and are unknown to the transmitter.
The diversity concept can be explained simply, 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
20 dB to 30 dB.
Small-scale fading
Large-scale fading
Small-scale fades are characterized by deep fade and rapid amplitude fluctuations which
occurs as mobile moves over distances of just few wavelengths. These fades are caused by
multiple reflections from the surroundings in the vicinity of the mobile. For narrow band signals
small scale fading results in Rayleigh fading distribution of signal strength over small
distances. In order to prevent deep fades from occurring, microscopic diversity techniques
can exploit the rapid changing of the signal. By selecting the best signal at all times, a receiver
can mitigate small-scale fading effects (antenna diversity or space diversity).
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Micro diversity
Methods that can overcome small scale fading are therefore called as “micro diversity”. The
(iv) Angular diversity – Multiple antenna with different antenna patterns are used. (v)
La rge c orre la tio ns of s igna ls be twe e n the a nte nna s a re unde s ira b le .
The firs t im po rta nt s te p in de s ig n ing d ive rs ity a nte nna is to e s ta b lis h
the
relationship
between antenna spacing and correlation coefficient.
Ex: mobile station in cellular and cordless systems, it is standard assumption that waves are
incident from all direction at mobile station.Thus points of the –ve interference of mpc are
spaced approximately λ/4 apart.This is the distance required for decorrelation. Minimum
distance for antenna elements in 900MHz GSM is 8cm and 1800MHz cordless is 4cm.
(ii)Temporal diversity:
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(c) combination of interleaving & coding.
Repetition coding:
The signal is repeated several times, where the repetition intervals are long enough
to achieve declaration.
ARQ:
This method receives sense the message, the transmitted to indicate whether it receive the data
sufficient quality. If not repeat request send by the receiver. ARQ is better than repetition
coding because of higher spectral efficiency.
Combination of interleaving & coding: This is the more advanced reason repetition coding.
The different symbols of a code word are transmitted at different times which increase the
probability that at least some of them arrived with a good SNR.
This method some signal is transmitted at different frequency. If these frequencies are
apart by more than coherent bandwidth then their fading is approximately independent.
Coherent bandwidth: It is the statistical measurement of a range of frequencies area which the
channel can be considered flat or in other words the approximate maximum bandwidth area
which two frequency of signal lightly to experience same fading.
It is not common to actually repeat the same information at two different frequencies rather
information spread area large bandwidth.
Spreading can be done by CDMA, OFDM, multi carrier CDMA etc., (iv)
Angular diversity:
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It is used in conjuction with spatial diversity. This method enhances decorrelation.
Different antenna patterns can be achieved ready easily. This diagram two antenna are used these
two antennas are identical but can have a different radiation pattern when mounted closed to each
other. This effect is due to mutual coupling. Antenna B acts as a reflector for antenna A where
pattern is therefore skewed to the left. Similarly, pattern of antenna – B is skewed to the right due
to the reflections form the antenna – A. Thus two patterns are different, this will lead to
decorrelation.
Thus receiving both polarizations using a dual polarized antenna and processing the signal
separately offers diversity. This diversity can be obtained without any requirement for a
minimum distance between antenna elements.
Macro diversity
The correlation distance is for large scales fading are on the order of tens or hundreds of
meters. So the spatial or temporal diversity cannot be used. For example, there is a hill between
the transmitter and receiver, adding antenna on either the BS or MS doesn’t help to
eliminate the shadowing caused by this hill.
The simplest method for macro diversity is the use of “on frequency repeaters”, that receivers the
signal and retransmit an amplified position of it.
Simulcast:
Simulcast is very similar to ON frequency repeaters in which the same signal is transmitter
simultaneously from different BS. The cellular application to base station should be
synchronized. It is also widely used for broadcast application especially digital TV. The
disadvantages of Simulcast is the large amount of signaling information that
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has to be carried on landlines or synchronization information as well as transmit data have to
be transported on microwave hills of landlines to the BS.
There are two ways of exploiting signals from the multiple diversity branches.
i. Selection diversity where the best signal copy is selected and processed
(demodulated & decoded).
While all other copies are discarded. There are different criteria for what contributes the best
signal.
ii. Combining diversity, where all copies of the signal are combined (before or after the
demodulate), and the combined signal is decoded. Again there are different algorithms for
combination of the signals. Combining diversity leads to better performance, as all available
information is exploited, on the downside, it requires a more complex receiver than selection
diversity. In most receivers all processing is done in the baseband.
Thus an receiver with combining diversity needs to down convert all available signals, and
combine then approximately in the baseband. Thus it requires Nr antenna element as well as Nr
complete radio frequency (RF) chains. An receiver with selection diversity requires only one RF
chain, as it process only a single received signal at a time.
1. Selection Diversity
Selection diversity is the simplest diversity technique. A block diagram of this method is
shown in Figure 7, where m demodulators are used to provide m diversity branches whose gains
are adjusted to provide the same average SNR for each branch. The receiver branch having the
highest instantaneous SNR is connected to the 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.
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Fig 7.Block diagram of selection 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. The resulting fading statistics are somewhat inferior to
those obtained by the other methods but the advantage of this method is very simple to
implement — only one receiver is required. A block diagram of this method is shown in Figure 8.
In this method, the signals from all of the M branches are weighted according to their
individual signal voltage to noise power ratios and then summed. Figure 9 shows a block
diagram of the technique. Here, the individual signals must be co-phased before
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being summed(unlike selection diversity) 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 SNRs.
Thus, it has the advantage of producing an output with an acceptable SNR even when none of the
individual signals are themselves acceptable. This technique gives the best statistical reduction
of fading of any known linear diversity combiner.
Modern DSP techniques and digital receivers are now using this optimal form of diversity
practical applications.
RAKE Receiver.
Propagation delay spread in the radio channel merely provides multiple versions of the
transmitted signal at the receiver. If these multipath components are delayed in time by more than
a chip duration, they appear like uncorrected noise at a CDMA receiver, and equalization is not
required.
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.
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Fig:Rake Receiver
The RAKE receiver, shown in Figure , 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.
A RAKE receiver utilizes multiple correlators to separately detect the M strongest multipath
components. The outputs of each correlator are weighted to provide a better estimate of the
transmitted signal than is provided by a single component. Demodulation and bit
decisions are then based on the weighted outputs of the Mcorrelators.
The basic idea of a RAKE receiver was first proposed by Price and Green [Pri58]. In outdoor
environments, the delay between multipath components is usually large and, if the chip rate is
properly selected, the low autocorrelation properties of a CDMA spreading sequence can assure
that multipath components will appear nearly uncorrelated with each other.
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,. Multipath
component m2 arrives t, later than component m ] .
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decisions based on only a 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, and the
corrupted signal may be discounted through the weighting process.
Decisions based on the combination of the M separate decision statistics 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 form an overall decision statistic as shown in Figure
The outputs of the M correlators are denoted as Z 19 Z Z1 ... and Z M . They are weighted by
a,, a 2 ,... and a M , respectively.
The weighting coefficients are based on the power or the SNR from each correlator 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
The weighting coefficients, a m ,are normalized to the output signal power of the
correlator in such a way that the coefficients sum to unity, as shown in following
equation
Choosing Weighting Coefficients based on the actual outputs of the correlators yield better
RAKE performance.
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Unit-V
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The MEAs of a MIMO system can be used for four different purposes: (i)
beamforming,
(ii) diversity,
first three concepts are the same as for smart antennas. Having multiple
antennas at both link ends leads to some interesting new technical possibilities, but does not
change the fundamental effects of this approach. Spatial multiplexing, on the other hand, is a new
concept, and has thus drawn the greatest attention. It allows direct improvement of capacity by
simultaneous transmission of multiple data streams
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Spatial Multiplexing
Spatial multiplexing uses MEAs at the TX for transmission of parallel data streams.An original
high-rate data stream is multiplexed into several parallel streams, each of which is sent from one
transmit antenna element. The channel “mixes up” these data streams, so that each of the
receive antenna elements sees a combination of them. If the channel is well behaved, the
received signals represent linearly independent combinations. In this case, appropriate signal
processing at the RX can separate the data streams. A basic condition is that the number of
receive antenna elements is at least as large as the number of transmit data streams. It is clear that
this approach
With Nt transmit antennas, we can form Nt different beams. We point all these beams at different
Interacting Objects (IOs), and transmit different data streams over them.
At the RX, we can use Nr antenna elements to form Nr beams, and also point them at different
IOs. If all the beams can be kept orthogonal to each other, there is no interference between the
data streams;in other words, we have established parallel channels.
The IOs (in combination with the beam spointing in their direction) play the same role as wires in
the transmission of multiple data streamson multiple wires.
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Fig:Transmission of different data streams via different interacting objects.
From this description, we can also immediately derive some important principles: the number of
possible data streams is limited by min(Nt,Nr,Ns), where Ns is the number of (significant) IOs.
We have already seen above that the number of data streams cannot be larger than the number
oftransmit antenna elements, and that we need a sufficient number of receive antenna elements
(atleast as many as data streams) to form the receive beams and, thus, be able to separate the data
streams. But it is also very important to notice that the number of IOs poses an upper limit: if two
data streams are transmitted to the same IO, then the RX has no possibility of sorting them out by
forming different beams. Operation
✓ High-rate signal is split into multiple lower-rate streams and each stream is transmitted
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The layered space time schemes, known as the Bell Laboratories layered space time (BLAST)
schemes, were developed to achieve transmission rates above one symbol per channel.
✓ The data stream is first converted into N, parallel streams, each being encoded
Joint ML decoding can be applied on each column. This leads to complexity, this complexity can
be further decreased by using the DFE decoder. is DFE — Decision Feedback Equalizer.
✓ At the receiver side, these data streams are separated based on optimum combining.
Diagonal BLAST
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✓ Each sub stream is subdivided into many sub-blocks and these sub-blocks are
transmitted by different antennas according to a round robin schedule.
At receiver side,
-zero forcing
Vertical BLAST
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✓ In V-BLAST the bit stream is first temporally encoded, interleaved and symbol
mapped.
✓ The resulting Ns symbols are then demultiplexed into N, substreams, and transmitted over the
antennas.
Advantages
Drawback
Precoding:
Precoding is motivated by the concept known as writing on dirty paper. The principle of
dirty paper coding (DPC) states that the effect of the interference can be cancelled by proper
coding.
i --> interference
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n --AWGN —Additive White Gaussiain Noise
SVD Precoding:
Linear transformation on the channel input x and output y is known as transmit precoding.
Transmit precoding and receiver shaping transform the MIMO channel into r H parallel scalar
channels with input
Features
Linear Precoding
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✓ Linear precoding is based on ZF beamforming
✓ As the number of user goes to infinity ZF beamforming capacity = dirty paper coding
capacity .
The linear precoder decouples the input signal into orthogonal signal modes in the form of eigen-
beams.
(i) In the case of perfect CSI [channel state information] the precoded orthogonal spatial
modes match the channel eigen-directions. There is no interference between these
signal streams.
(ii) With spatial CSI, precoder design must reduce the interference among signals
(iii) For perfect CSI at the transmitter, a diversity gain can also be delivered
MIMO-Beam forming
Multiple antennas are used to obtain beamforming or diversity gain. Same information is
transmitted by more than one antenna, in the receiver side, symbol is weighted by a complex
scale factor.
✓ Full diversity gain can be achieved by transmit beamforming and receive combining. The
One solution is quantized beamforming. The receiver quantizes the beamforming vector
using a fixed codebook. Codebook is availble at both the transmitter and the receiver
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System Model and channel state information
Let us first establish the generic system that will be considered for capacity
computations. Figure exhibits a block diagram. At the TX, the data stream enters an encoder,
whose outputs are forwarded to Nt transmit antennas. From the antennas, the signal is sent
through the wireless propagation channel, which is assumed to be quasi-static and frequency-flat
if not stated otherwise.
By quasi-static we mean that the coherence time of the channel is so longthat “a large number” of
bits can be transmitted within this time.We denote the Nr × Nt matrix of the channel as
whose entries hij are complex channel gains (transfer functions) from the j th transmit to the
ithreceive antenna. The received signal vector
contains the signals received by Nr antenna elements, where s is the transmit signal vector and
nis the noise vector. Channel State information Algorithms for MIMO transmission can be
categorized by the amount of CSI that they require. We distinguish the following cases:
1. Full CSI at the TX (CSIT) and full CSI at the RX (CSIR): in this ideal case, both the TX
and the RX have full and perfect knowledge of the channel. This case obviously results in the
highest possible capacity.
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2. Average CSIT and full CSIR: in this case, the RX has full information of the
instantaneouschannel state, but the TX knows only the average CSI – e.g., the correlation matrix
of H or theangular power spectrum. To achieve anddoes not require reciprocity or fast feedback;
however, it does require calibration (to eliminatethe nonreciprocity of transmit and receive
chains) or slow feedback.
3. No CSIT and full CSIR: this is the case that can be achieved most easily, without any feedback
or calibration. The TX simply does not use any CSI, while the RX learns the
instantaneouschannel state from a training sequence or using blind estimation.
4. Noisy CSI : when we assume “full CSI” at the RX, this implies that the RX has
learned thechannel state perfectly. However, any received training sequence will be affected by
additivenoise as well as quantization noise. It is thus more realistic to assume a
“mismatched RX,”where the RX processes the signal based on the observed channel Hobs, while
in reality the signals pass through channel Htrue
5. No CSIT and no CSIR: it is remarkable that channel capacity is also high when neither
the TX nor the RX have CSI. We can, e.g., use a generalization of differential modulation.
The first key step in understanding MIMO systems is the derivation of the capacity equation for
MIMO systems in nonfading channels, often known as “Foschini’s equation” [Foschini and Gan
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1998]. Let us start with the capacity equation for “normal” (single-antenna)
AdditiveWhite GaussianNoise (AWGN) channels.
Where γ is the SNR at the RX, and H is the normalized transfer function from the TX to the RX.
The key statement of this equation is that capacity increases only logarithmically with the SNR,
so that boosting the transmit power is a highly ineffective way of increasing capacity. Consider a
singular value decomposition11 of the channel:
where∑is a diagonal matrix containing singular values, and W and U† are unitary matrices
composed of the left and right singular vectors, respectively. The received signal is then
Then, multiplication of the transmit data vector by matrix U and the received signal vector
by W†diagonalizes the channel:
The capacity of channel H is thus given by the sum of the capacities of the eigenmodes of the
channel:
Where σ2 is noise variance, and Pkis the power allocated to the eigenmode; we assume that
Pk= P is independent of the number of antennas. This capacity expression can be shown to
be equivalent to
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Where INr is the Nr ×Nr identity matrix, γ is the mean SNR per RX branch, and Rssis the
correlation matrix of the transmit data (if data at the different antenna elements are uncorrelated,
it is a diagonal matrix with entries that describe the power distribution among antennas).
No Channel State Information at the Transmitter and Full CSI at the Receiver
When the RX knows the channel perfectly, but no CSI is available at the TX, it is optimum to
assign equal transmit power to all TX antennas, Pk= P/Nt, and use uncorrelated data streams.
Capacity thus takes on the form:
It is worth noting that (for sufficiently large Ns) the capacity of a MIMO system increases
linearly with min(Nt,Nr), irrespective of whether the channel is known at the TX or not.
Let us now look at a few special cases. To make the discussion easier, we assume that
Nr = N:
1. All transfer functions are identical – i.e., h1,1 = h1,2 = . . . = hN,N. This case occurs when all
antenna elements are spaced very closely together, and all waves are coming from similar
directions. In such a case, the rank of the channel matrix is unity. Then, capacity is
We see that in this case the SNR is increased by a factor of N compared with the single antenna
case, due to beam forming gain at the RX. However, this only leads to a logarithmic increase
incapacity with the number of antennas.
2. All transfer functions are different such that the channel matrix is full rank, and has N eigen
values of equal magnitude. This case can occur when the antenna elements are spaced far apart
and are arranged in a special way. In this case, capacity is
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and, thus, increases linearly with the number of antenna elements.
3. Parallel transmission channels – e.g., parallel cables. In this case, capacity also increases
linearly with the number of antenna elements. However, the SNR per channel decreases with N,
so that total capacity is
Full Channel State Information at the Transmitter and Full CSI at the Receiver
Let us next consider the case where both the RX and TX know the channel perfectly. In such a
case, it can be more advantageous to distribute power not uniformly between the different
transmit antennas (or eigen modes) but rather assign it based on the channel state. In other words,
we are faced with the problem of optimally allocating power to several parallel channels, each of
which has a different SNR, and therefore the answer is the same: water filling.
Transmitter diversity
Transmit Diversity
Multiple antennas can be installed at just one link end (usually the BS). For the uplink
transmission from the MS to BS, multiple antennas can act as receive diversity branches.
For the downlink, any possible diversity originates at the transmitter. we will thus discuss ways
of transmitting signals from several TX antennas and achieve a diversity effect with it.
Time diversity and frequency diversity inherently involve the TX, and thus need not be
discussed again here.
The first situation we analyze is the case where the TX knows the channel perfectly. the optimum
transmission scheme linearly weights signals transmitted from different antenna elements
with the complex conjugates of the channel transfer functions from
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the transmit antenna elements to the single receive antenna. This approach is known as maximum
ratio transmission.
In many cases, Channel State Information (CSI) is not available at the TX. We then cannot
simply transmit weighted copies of the same signal from different transmit antennas, because we
cannot know how they would add up at the RX. It is equally likely for the addition of different
components to be constructive or destructive; in other words, we would just be adding up MPCs
with random phases, which results in Rayleigh fading.
We thus cannot gain any diversity (or beam forming).In order to give benefits, transmission of
the signals from different antenna elements has to be done is such a way that it allows the RX
to distinguish different transmitted signal components. One way is delay diversity. In this
scheme, signals transmitted from different antenna elements are delayed copies of the same
signal. This makes sure that the effective impulse response is delay dispersive, even if the
channel itself is flat fading.
So, in a flat-fading channel, we transmit data streams with a delay of 1 symbol duration
(relative to preceding antennas) from each of the transmit antennas. The effective impulse
response of the channel then becomes
where the hn are gains from the nth transmit antenna to the receive antenna, and the impulse
response has been normalized so that total transmit power is independent of the number of
antenna elements. The signals from different transmit antennas to the RX act effectively as
delayed MPCs. If antenna elements are spaced sufficiently far apart, these coefficients fade
independently. If the channel from a single transmit antenna to the RX is already delay
dispersive, then the scheme still works, but care has to be taken in the choice of delays for
different antenna element
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