Wireless Communication Notes PDF
Wireless Communication Notes PDF
Wireless Communication Notes PDF
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WIRELESS COMMUNICATION
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NETWORKS
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SYLLABUS
UNIT-I
The Cellular Concept-System Design Fundamentals: Introduction, Frequency Reuse, Channel
Assignment Strategies, Handoff Strategies- Prioritizing Handoffs, Practical Handoff Considerations,
Interference and system capacity — Co channel Interference and system capacity, Channel planning for
Wireless Systems, Adjacent Channel interference , Power Control for Reducing interference, Trunking
and Grade of Service, Improving Coverage & Capacity in Cellular Systems- Cell Splitting, Sectoring.
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UNIT—II
Mobile Radio Propagation: Large-Scale Path Loss: Introduction to Radio Wave Propagation, Free Space
Propagation Model, Relating Power to Electric Field, The Three Basic Propagation Mechanisms,
Reflection- Reflection from Dielectrics, Brewster Angle, Reflection from prefect conductors, Ground
Reflection (Two-Ray) Model, Diffraction-Fresnel Zone Geometry, Knife-edge Diffraction Model,
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Multiple knife-edge Diffraction, Scattering, Outdoor Propagation Models- Longley-Ryce Model,
Okumura Model, Hata Model, PCS Extension to Hata Model, Walfisch and Bertoni Model, Wideband
PCS Microcell Model, lndoor Propagation Models-Partition losses (Same Floor), Partition losses between
Floors, Log-distance path loss model, Ericsson Multiple Breakpoint Model, Attenuation Factor Model,
Signal penetration into buildings, Ray Tracing and Site Specific Modeling.
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UNIT —III
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Mobile Radio Propagation: Small —Scale Fading and Multipath: Small Scale Multipath propagation-
Factors influencing small scale fading, Doppler shift, Impulse Response Model of a multipath channel-
Relationship between Bandwidth and Received power, Small-Scale Multipath Measurements-Direct RF
Pulse System, Spread Spectrum Sliding Correlator Channel Sounding, Frequency Domain Channels
Sounding, Parameters of Mobile Multipath Channels-Time Dispersion Parameters, Coherence Bandwidth,
Doppler Spread and Coherence Time, Types of Small-Scale Fading-Fading effects Due to Multipath Time
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Delay Spread, Flat fading, Frequency selective fading, Fading effects Due to Doppler Spread-Fast fading,
slow fading, Statistical Models for multipath Fading Channels-Clarke’s model for flat fading, spectral
shape due to Doppler spread in Clarke’s model, Simulation of Clarke and Gans Fading Model, Level
crossing and fading statistics, Two-ray Rayleigh Fading Model.
UNIT -IV
Equalization and Diversity: Introduction, Fundamentals of Equalization, Training A Generic Adaptive
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Wireless Networks: Introduction to wireless Networks, Advantages and disadvantages of Wireless Local
Area Networks, WLAN Topologies, WLAN Standard IEEE 802.11 ,IEEE 802.11 Medium Access
Control, Comparision
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of IEEE 802.11 a,b,g and n standards, IEEE 802.16 and its enhancements, Wireless PANs, Hiper Lan,
WLL.
TEXT BOOKS
1. Wireless Communications, Principles, Practice — Theodore, S.Rappaport, 2nd Ed., 2002, PHI.
2. Wireless Communications-Andrea Goldsmith, 2005 Cambridge University Press.
3. Mobile Cellular Communication — Gottapu Sasibhushana Rao, Pearson Education, 2012.
REFERENCE BOOKS
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1. Principles of Wireless Networks — Kaveh Pah Laven and P. Krishna Murthy, 2002, PE
2. Wireless Digital Communications — Kamilo Feher, 1999, PHI.
3. Wireless Communication and Networking — William Stallings, 2003,PHI.
4. Wireless Communication — Upen Dalal, Oxford Univ. Press
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5. Wireless Communications and Networking — Vijay K. Gary, Elsevier.
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The Cellular Engineering
Fundamentals
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Introduction
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In Chapter 1, we have seen that the technique of substituting a single high power
transmitter by several low power transmitters to support many users is the backbone
of the cellular concept. In practice, the following four parameters are most important
while considering the cellular issues: system capacity, quality of service, spectrum
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efficiency and power management. Starting from the basic notion of a cell, we would
deal with these parameters in the context of cellular engineering in this chapter.
What is a Cell?
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The power of the radio signals transmitted by the BS decay as the signals travel
away from it. A minimum amount of signal strength (let us say, x dB) is needed in
order to be detected by the MS or mobile sets which may the hand-held personal
units or those installed in the vehicles. The region over which the signal strength
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lies above this threshold value x dB is known as the coverage area of a BS and
it must be a circular region, considering the BS to be isotropic radiator. Such a
circle, which gives this actual radio coverage, is called the foot print of a cell (in
reality, it is amorphous). It might so happen that either there may be an overlap
between any two such side by side circles or there might be a gap between the
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Figure 3.1: Footprint of cells showing the overlaps and gaps.
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coverage areas of two adjacent circles. This is shown in Figure 3.1. Such a circular
geometry, therefore, cannot serve as a regular shape to describe cells. We need a
regular shape for cellular design over a territory which can be served by 3 regular
polygons, namely, equilateral triangle, square and regular hexagon, which can cover
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the entire area without any overlap and gaps. Along with its regularity, a cell must
be designed such that it is most reliable too, i.e., it supports even the weakest mobile
with occurs at the edges of the cell. For any distance between the center and the
farthest point in the cell from it, a regular hexagon covers the maximum area. Hence
regular hexagonal geometry is used as the cells in mobile communication.
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Frequency Reuse
systems are based that involve the partitioning of an RF radiating area into cells.
The increased capacity in a commercial wireless network, compared with a network
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with a single transmitter, comes from the fact that the same radio frequency can be
reused in a different area for a completely different transmission.
Frequency reuse in mobile cellular systems means that frequencies allocated to
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Figure 3.2: Frequency reuse technique of a cellular system.
the service are reused in a regular pattern of cells, each covered by one base station.
The repeating regular pattern of cells is called cluster. Since each cell is designed
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to use radio frequencies only within its boundaries, the same frequencies can be
reused in other cells not far away without interference, in another cluster. Such cells
are called ‘co-channel’ cells. The reuse of frequencies enables a cellular system to
handle a huge number of calls with a limited number of channels. Figure 3.2 shows
a frequency planning with cluster size of 7, showing the co-channels cells in different
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clusters by the same letter. The closest distance between the co-channel cells (in
different clusters) is determined by the choice of the cluster size and the layout of
the cell cluster. Consider a cellular system with S duplex channels available for
use and let N be the number of cells in a cluster. If each cell is allotted K duplex
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channels with all being allotted unique and disjoint channel groups we have S = KN
under normal circumstances. Now, if the cluster are repeated M times within the
total area, the total number of duplex channels, or, the total number of users in the
system would be T = MS = KMN . Clearly, if K and N remain constant, then
T ∝M (3.1)
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N∝ . (3.2)
M
Hence the capacity gain achieved is directly proportional to the number of times
a cluster is repeated, as shown in (3.1), as well as, for a fixed cell size, small N
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decreases the size of the cluster with in turn results in the increase of the number
of clusters (3.2) and hence the capacity. However for small N, co-channel cells are
located much closer and hence more interference. The value of N is determined by
calculating the amount of interference that can be tolerated for a sufficient quality
communication. Hence the smallest N having interference below the tolerated limit
is used. However, the cluster size N cannot take on any value and is given only by
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the following equation
N = i2 + ij + j 2 , i ≥ 0, j ≥ 0, (3.3)
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where i and j are integer numbers.
Ex. 1: Find the relationship between any two nearest co-channel cell distance D
and the cluster size N.
Solution: For hexagonal
√
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cells, it can be shown that the distance between two adjacent
cell centers = 3R, where R is the radius of any cell. The normalized co-channel
cell distance Dn can be calculated by traveling ’i’ cells in one direction and then
traveling ’j’ cells in anticlockwise 120 o of the primary direction. Using law of vector
addition,
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D2 2 2 o o 2
(3.4)
n = j cos (30 ) + (i + j sin(30 ))
√ √
D = Dn 3R = 3NR. (3.6)
Ex. 2: Find out the surface area of a regular hexagon with radius R, the surface
area of a large hexagon with radius D, and hence compute the total number of cells
in this large hexagon.
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Hint: In general, this large hexagon with radius D encompasses the center cluster of
N cells and one-third of the cells associated with six other peripheral large hexagons.
Thus, the answer must be N + 6( 3N ) = 3N .
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Channel Assignment Strategies
With the rapid increase in number of mobile users, the mobile service providers
had to follow strategies which ensure the effective utilization of the limited radio
spectrum. With increased capacity and low interference being the prime objectives,
a frequency reuse scheme was helpful in achieving this objectives. A variety of
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channel assignment strategies have been followed to aid these objectives. Channel
assignment strategies are classified into two types: fixed and dynamic, as discussed
below.
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Fixed Channel Assignment (FCA)
In fixed channel assignment strategy each cell is allocated a fixed number of voice
channels. Any communication within the cell can only be made with the designated
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unused channels of that particular cell. Suppose if all the channels are occupied,
then the call is blocked and subscriber has to wait. This is simplest of the channel
assignment strategies as it requires very simple circuitry but provides worst channel
utilization. Later there was another approach in which the channels were borrowed
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from adjacent cell if all of its own designated channels were occupied. This was
named as borrowing strategy. In such cases the MSC supervises the borrowing pro-
cess and ensures that none of the calls in progress are interrupted.
In dynamic channel assignment strategy channels are temporarily assigned for use
in cells for the duration of the call. Each time a call attempt is made from a cell the
corresponding BS requests a channel from MSC. The MSC then allocates a channel
to the requesting the BS. After the call is over the channel is returned and kept in
a central pool. To avoid co-channel interference any channel that in use in one cell
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can only be reassigned simultaneously to another cell in the system if the distance
between the two cells is larger than minimum reuse distance. When compared to the
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FCA, DCA has reduced the likelihood of blocking and even increased the trunking
capacity of the network as all of the channels are available to all cells, i.e., good
quality of service. But this type of assignment strategy results in heavy load on
switching center at heavy traffic condition.
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Ex. 3: A total of 33 MHz bandwidth is allocated to a FDD cellular system with
two 25 KHz simplex channels to provide full duplex voice and control channels.
Compute the number of channels available per cell if the system uses (i) 4 cell, (ii)
7 cell, and (iii) 8 cell reuse technique. Assume 1 MHz of spectrum is allocated to
control channels. Give a distribution of voice and control channels.
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Solution: One duplex channel = 2 x 25 = 50 kHz of spectrum. Hence the total
available duplex channels are = 33 MHz / 50 kHz = 660 in number. Among these
channels, 1 MHz / 50 kHz = 20 channels are kept as control channels.
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(a) For N = 4, total channels per cell = 660/4 = 165.
Among these, voice channels are 160 and control channels are 5 in number.
(b) For N = 7, total channels per cell are 660/7 ≈ 94. Therefore, we have to go for
a more exact solution. We know that for this system, a total of 20 control channels
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and a total of 640 voice channels are kept. Here, 6 cells can use 3 control channels
and the rest two can use 2 control channels each. On the other hand, 5 cells can use
92 voice channels and the rest two can use 90 voice channels each. Thus the total
solution for this case is:
6 x 3 + 1 x 2 = 20 control channels, and,
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Handoff Process
When a user moves from one cell to the other, to keep the communication between
the user pair, the user channel has to be shifted from one BS to the other without
interrupting the call, i.e., when a MS moves into another cell, while the conversation
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is still in progress, the MSC automatically transfers the call to a new FDD channel
without disturbing the conversation. This process is called as handoff. A schematic
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Figure 3.3: Handoff scenario at two adjacent cell boundary.
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level is set as the minimum acceptable for good voice quality (Prmin ), then a slightly
stronger level is chosen as the threshold (PrH )at which handoff has to be made, as
shown in Figure 3.4. A parameter, called power margin, defined as
∆ = Pr H − Pr min (3.7)
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is quite an important parameter during the handoff process since this margin ∆ can
neither be too large nor too small. If ∆ is too small, then there may not be enough
time to complete the handoff and the call might be lost even if the user crosses the
cell boundary.
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If ∆ is too high o the other hand, then MSC has to be burdened with unnecessary
handoffs. This is because MS may not intend to enter the other cell. Therefore ∆
should be judiciously chosen to ensure imperceptible handoffs and to meet other
objectives.
(a) Transmitted power: as we know that the transmission power is different for dif-
ferent cells, the handoff threshold or the power margin varies from cell to cell.
(b) Received power: the received power mostly depends on the Line of Sight (LoS)
path between the user and the BS. Especially when the user is on the boundary of
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from i-th cell to j-th cell.
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Figure 3.4: Handoff process associated with power levels, when the user is going
the two cells, the LoS path plays a critical role in handoffs and therefore the power
margin ∆ depends on the minimum received power value from cell to cell.
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(c) Area and shape of the cell: Apart from the power levels, the cell structure also
a plays an important role in the handoff process.
(d) Mobility of users: The number of mobile users entering or going out of a partic-
ular cell, also fixes the handoff strategy of a cell.
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To illustrate the reasons (c) and (d), let us consider a rectangular cell with sides R1
and R2 inclined at an angle θ with horizon, as shown in the Figure 3.5. Assume N1
users are having handoff in horizontal direction and N2 in vertical direction per unit
length.
The number of crossings along R1 side is : (N1cosθ + N2sinθ)R1 and the number of
crossings along R2 side is : (N1sinθ + N2cosθ)R2.
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Figure 3.5: Handoff process with a rectangular cell inclined at an angle θ.
R1 by A and equating
R2
dλ H
to zero, we get
dR 1 W
Now, given the fixed area A = R1R2, we need to find λmin
H for a given θ. Replacing
N1sinθ + N2cosθ
R21 = A( N cosθ + N sinθ ).
1 2
(3.9)
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Similarly, for R2, we get
N1cosθ + N2sinθ
R2 = A( ). (3.10)
2
N 1sinθ + N 2cosθ
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From the above equations, we have λH = 2 A(N1 N2 + (N 2 + N 2)cosθsinθ) which
1 2
√
means it it minimized at θ = 0o. Hence λmin
H = 2 AN 1 N 2 . Putting the value of θ
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R1 N1
in (3.9) or (3.10), we have = . This has two implications: (i) that handoff is
R2 N2
minimized if rectangular cell is aligned with X-Y axis, i.e., θ = 0o, and, (ii) that the
number of users crossing the cell boundary is inversely proportional to the dimension
of the other side of the cell. The above analysis has been carried out for a simple
square cell and it changes in more complicated way when we consider a hexagonal
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cell.
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strengths of voice channels to determine the relative positions of the subscriber.
The special receivers located on the BS are controlled by the MSC to monitor the
signal strengths of the users in the neighboring cells which appear to be in need
of handoff. Based on the information received from the special receivers the MSC
decides whether a handoff is required or not. The approximate time needed to make
a handoff successful was about 5-10 s. This requires the value of ∆ to be in the
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order of 6dB to 12dB.
In the 2G systems, the MSC was relieved from the entire operation. In this
generation, which started using the digital technology, handoff decisions were mobile
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assisted and therefore it is called Mobile Assisted Hand-Off (MAHO). In MAHO,
the mobile center measures the power changes received from nearby base stations
and notifies the two BS. Accordingly the two BS communicate and channel transfer
occurs. As compared to 1G, the circuit complexity was increased here whereas the
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delay in handoff was reduced to 1-5 s. The value of ∆ was in the order of 0-5 dB.
However, even this amount of delay could create a communication pause.
In the current 3G systems, the MS measures the power from adjacent BS and
automatically upgrades the channels to its nearer BS. Hence this can be termed as
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Mobile Controlled Hand-Off (MCHO). When compared to the other generations,
delay during handoff is only 100 ms and the value of ∆ is around 20 dBm. The
Quality Of Service (QoS) has improved a lot although the complexity of the circuitry
has further increased which is inevitable.
All these types of handoffs are usually termed as hard handoff as there is a shift
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in the channels involved. There is also another kind of handoff, called soft handoff,
as discussed below.
Handoff in CDMA: In spread spectrum cellular systems, the mobiles share the same
channels in every cell. The MSC evaluates the signal strengths received from different
BS for a single user and then shifts the user from one BS to the other without actually
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changing the channel. These types of handoffs are called as soft handoff as there is
no change in the channel.
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Handoff Priority
While assigning channels using either FCA or DCA strategy, a guard channel concept
must be followed to facilitate the handoffs. This means, a fraction of total available
channels must be kept for handoff requests. But this would reduce the carried
traffic and only fewer channels can be assigned for the residual users of a cell. A
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good solution to avoid such a dead-lock is to use DCA with handoff priority (demand
based allocation).
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(a) Different speed of mobile users: with the increase of mobile users in urban areas,
microcells are introduced in the cells to increase the capacity (this will be discussed
later in this chapter). The users with high speed frequently crossing the micro-cells
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become burdened to MSC as it has to take care of handoffs. Several schemes thus
have been designed to handle the simultaneous traffic of high speed and low speed
users while minimizing the handoff intervention from the MSC, one of them being
the ‘Umbrella Cell’ approach. This technique provides large area coverage to high
speed users while providing small area coverage to users traveling at low speed. By
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using different antenna heights and different power levels, it is possible to provide
larger and smaller cells at a same location. As illustrated in the Figure 3.6, umbrella
cell is co-located with few other microcells. The BS can measure the speed of the
user by its short term average signal strength over the RVC and decides which cell
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to handle that call. If the speed is less, then the corresponding microcell handles
the call so that there is good corner coverage. This approach assures that handoffs
are minimized for high speed users and provides additional microcell channels for
pedestrian users.
(b) Cell dragging problem: this is another practical problem in the urban area with
additional microcells. For example, consider there is a LOS path between the MS
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and BS1 while the user is in the cell covered by BS2. Since there is a LOS with the
BS1, the signal strength received from BS1 would be greater than that received from
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BS2. However, since the user is in cell covered by BS2, handoff cannot take place
and as a result, it experiences a lot of interferences. This problem can be solved by
judiciously choosing the handoff threshold along with adjusting the coverage area.
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(c) Inter-system handoff: if one user is leaving the coverage area of one MSC and is
entering the area of another MSC, then the call might be lost if there is no handoff in
this case too. Such a handoff is called inter-system handoff and in order to facilitate
this, mobiles usually have roaming facility.
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Susceptibility and interference problems associated with mobile communications
equipment are because of the problem of time congestion within the electromag-
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netic spectrum. It is the limiting factor in the performance of cellular systems. This
interference can occur from clash with another mobile in the same cell or because
of a call in the adjacent cell. There can be interference between the base stations
operating at same frequency band or any other non-cellular system’s energy leaking
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inadvertently into the frequency band of the cellular system. If there is an interfer-
ence in the voice channels, cross talk is heard will appear as noise between the users.
The interference in the control channels leads to missed and error calls because of
digital signaling. Interference is more severe in urban areas because of the greater
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RF noise and greater density of mobiles and base stations. The interference can be
divided into 2 parts: co-channel interference and adjacent channel interference.
For the efficient use of available spectrum, it is necessary to reuse frequency band-
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width over relatively small geographical areas. However, increasing frequency reuse
also increases interference, which decreases system capacity and service quality. The
cells where the same set of frequencies is used are call co-channel cells. Co-channel
interference is the cross talk between two different radio transmitters using the same
radio frequency as is the case with the co-channel cells. The reasons of CCI can be
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If the cell size and the power transmitted at the base stations are same then CCI
will become independent of the transmitted power and will depend on radius of the
cell (R) and the distance between the interfering co-channel cells (D). If D/R ratio
is increased, then the effective distance between the co-channel cells will increase
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and interference will decrease. The parameter Q is called the frequency reuse ratio
and is related to the cluster size. For hexagonal geometry
√
Q = D/R = 3N. (3.11)
From the above equation, small of ‘Q’ means small value of cluster size ‘N’ and
increase in cellular capacity. But large ‘Q’ leads to decrease in system capacity
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but increase in transmission quality. Choosing the options is very careful for the
selection of ‘N’, the proof of which is given in the first section.
The Signal to Interference Ratio (SIR) for a mobile receiver which monitors the
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forward channel can be calculated as
S S (3.12)
= . i0
I i=1 Ii
where i0 is the number of co-channel interfering cells, S is the desired signal power
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from the baseband station and Ii is the interference power caused by the i-th interfer-
ing co-channel base station. In order to solve this equation from power calculations,
we need to look into the signal power characteristics. The average power in the
mobile radio channel decays as a power law of the distance of separation between
transmitter and receiver. The expression for the received power Pr at a distance d
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where P0 is the power received at a close-in reference point in the far field region at
a small distance do from the transmitting antenna, and ‘n’ is the path loss exponent.
Let us calculate the SIR for this system. If Di is the distance of the i-th interferer
from the mobile, the received power at a given mobile due to i-th interfering cell
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is proportional to (Di)−n (the value of ’n’ varies between 2 and 4 in urban cellular
systems).
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Let us take that the path loss exponent is same throughout the coverage area
and the transmitted power be same, then SIR can be approximated as
. i0 n
IS R− i (3.15)
= −n
i=1 D
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where the mobile is assumed to be located at R distance from the cell center. If
we consider only the first layer of interfering cells and we assume that the interfer-
ing base stations are equidistant from the reference base station and the distance
between the cell centers is ’D’ then the above equation can be converted as
n √ n
S (D/R) ( 3N ) (3.16)
= =
I i0 i0
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which is an approximate measure of the SIR. Subjective tests performed on AMPS
cellular system which uses FM and 30 kHz channels show that sufficient voice quality
can be obtained by SIR being greater than or equal to 18 dB. If we take n=4
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, the value of ’N’ can be calculated as 6.49. Therefore minimum N is 7. The
above equations are based on hexagonal geometry and the distances from the closest
interfering cells can vary if different frequency reuse plans are used.
We can go for a more approximate calculation for co-channel SIR. This is the
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example of a 7 cell reuse case. The mobile is at a distance of D-R from 2 closest
interfering cells and approximately D+R/2, D, D-R/2 and D+R distance from other
interfering cells in the first tier. Taking n = 4 in the above equation, SIR can be
approximately calculated as
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S R−4 (3.17)
=
I 2(D − R)−4 + (D + R)−4 + (D)−4 + (D + R/2)−4 + (D − R/2)−4
which can be rewritten in terms frequency reuse ratio Q as
S 1 . (3.18)
=
I 2(Q − 1) + (Q + 1) + (Q) + (Q + 1/2)−4 + (Q − 1/2)−4
−4 −4 −4
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Using the value of N equal to 7 (this means Q = 4.6), the above expression yields
that worst case SIR is 53.70 (17.3 dB). This shows that for a 7 cell reuse case the
worst case SIR is slightly less than 18 dB. The worst case is when the mobile is at
the corner of the cell i.e., on a vertex as shown in the Figure 3.6. Therefore N = 12
cluster size should be used. But this reduces the capacity by 7/12 times. Therefore,
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Figure 3.6: First tier of co-channel interfering cells
channel. This is called near-far effect. The more adjacent channels are packed into
the channel block, the higher the spectral efficiency, provided that the performance
degradation can be tolerated in the system link budget. This effect can also occur
if a mobile close to a base station transmits on a channel close to one being used
by a weak mobile. This problem might occur if the base station has problem in
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discriminating the mobile user from the ”bleed over” caused by the close adjacent
channel mobile.
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Adjacent channel interference occurs more frequently in small cell clusters and heav-
ily used cells. If the frequency separation between the channels is kept large this
interference can be reduced to some extent. Thus assignment of channels is given
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such that they do not form a contiguous band of frequencies within a particular
cell and frequency separation is maximized. Efficient assignment strategies are very
much important in making the interference as less as possible. If the frequency fac-
tor is small then distance between the adjacent channels cannot put the interference
level within tolerance limits. If a mobile is 10 times close to the base station than
other mobile and has energy spill out of its passband, then SIR for weak mobile is
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approximately
S
= 10−n (3.19)
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which can be easily found from the earlier SIR expressions. If n = 4, then SIR is
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−52 dB. Perfect base station filters are needed when close-in and distant users share
the same cell. Practically, each base station receiver is preceded by a high Q cavity
filter in order to remove adjacent channel interference. Power control is also very
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much important for the prolonging of the battery life for the subscriber unit but also
reduces reverse channel SIR in the system. Power control is done such that each
mobile transmits the lowest power required to maintain a good quality link on the
reverse channel.
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Previously, we have seen that the frequency reuse technique in cellular systems
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allows for almost boundless expansion of geographical area and the number of mobile
system users who could be accommodated. In designing a cellular layout, the two
parameters which are of great significance are the cell radius R and the cluster size
√
N, and we have also seen that co-channel cell distance D = 3NR. In the following,
a brief description of the design trade-off is given, in which the above two parameters
play a crucial role.
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The cell radius governs both the geographical area covered by a cell and also
the number of subscribers who can be serviced, given the subscriber density. It is
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easy to see that the cell radius must be as large as possible. This is because, every
cell requires an investment in a tower, land on which the tower is placed, and radio
transmission equipment and so a large cell size minimizes the cost per subscriber.
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Eventually, the cell radius is determined by the requirement that adequate signal
to noise ratio be maintained over the coverage area. The SNR is determined by
several factors such as the antenna height, transmitter power, receiver noise figure
etc. Given a cell radius R and a cluster size N , the geographic area covered by a
cluster is
√
Acluster = N Acell = N 3 3R2 /2. (3.20)
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If the total serviced area is Atotal, then the number of clusters M that could be
accommodated is given by
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M = Atotal /Acluster = Atotal /(N 3 3R2 /2). (3.21)
Note that all of the available channels N, are reused in every cluster. Hence, to make
the maximum number of channels available to subscribers, the number of clusters
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M should be large, which, by Equation (3.21), shows that the cell radius should
be small. However, cell radius is determined by a trade-off: R should be as large
as possible to minimize the cost of the installation per subscriber, but R should
be as small as possible to maximize the number of customers that the system can
accommodate. Now, if the cell radius R is fixed, then the number of clusters could be
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maximized by minimizing the size of a cluster N . We have seen earlier that the size
of a cluster depends on the frequency reuse ratio Q. Hence, in determining the value
of N , another trade-off is encountered in that N must be small to accommodate
large number of subscribers, but should be sufficiently large so as to minimize the
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interference effects.
Now, we focus on the issues regarding system expansion. The history of cellular
phones has been characterized by a rapid growth and expansion in cell subscribers.
Though a cellular system can be expanded by simply adding cells to the geographical
area, the way in which user density can be increased is also important to look at.
This is because it is not always possible to counter the increasing demand for cellular
ll
systems just by increasing the geographical coverage area due to the limitations in
obtaining new land with suitable requirements. We discuss here two methods for
A
dealing with an increasing subscriber density: Cell Splitting and Sectoring. The
other method, microcell zone concept can treated as enhancing the QoS in a cellular
system.
20
The basic idea of adopting the cellular approach is to allow space for the growth
of mobile users. When a new system is deployed, the demand for it is fairly low and
users are assumed to be uniformly distributed over the service area. However, as new
users subscribe to the cellular service, the demand for channels may begin to exceed
the capacity of some base stations. As discussed previously,the number of channels
available to customers (equivalently, the channel density per square kilometer) could
ld
be increased by decreasing the cluster size. However, once a system has been initially
deployed, a system-wide reduction in cluster size may not be necessary since user
density does not grow uniformly in all parts of the geographical area. It might be
or
that an increase in channel density is required only in specific parts of the system
to support an increased demand in those areas. Cell-splitting is a technique which
has the capability to add new smaller cells in specific areas of the system.
Cell-Splitting
W
Cell Splitting is based on the cell radius reduction and minimizes the need to modify
the existing cell parameters. Cell splitting involves the process of sub-dividing a
congested cell into smaller cells, each with its own base station and a corresponding
TU
reduction in antenna size and transmitting power. This increases the capacity of
a cellular system since it increases the number of times that channels are reused.
Since the new cells have smaller radii than the existing cells, inserting these smaller
cells, known as microcells, between the already existing cells results in an increase
of capacity due to the additional number of channels per unit area. There are few
JN
challenges in increasing the capacity by reducing the cell radius. Clearly, if cells
are small, there would have to be more of them and so additional base stations
will be needed in the system. The challenge in this case is to introduce the new
base stations without the need to move the already existing base station towers.
The other challenge is to meet the generally increasing demand that may vary quite
ll
rapidly between geographical areas of the system. For instance, a city may have
highly populated areas and so the demand must be supported by cells with the
A
smallest radius. The radius of cells will generally increase as we move from urban to
sub urban areas, because the user density decreases on moving towards sub-urban
areas. The key factor is to add as minimum number of smaller cells as possible
21
ld
or
W
Figure 3.7: Splitting of congested seven-cell clusters.
wherever an increase in demand occurs. The gradual addition of the smaller cells
implies that, at least for a time, the cellular system operates with cells of more than
TU
one size.
Figure 3.7 shows a cellular layout with seven-cell clusters. Consider that the cells
in the center of the diagram are becoming congested, and cell A in the center has
reached its maximum capacity. Figure also shows how the smaller cells are being
superimposed on the original layout. The new smaller cells have half the cell radius
JN
of the original cells. At half the radius, the new cells will have one-fourth of the area
and will consequently need to support one-fourth the number of subscribers. Notice
that one of the new smaller cells lies in the center of each of the larger cells. If
we assume that base stations are located in the cell centers, this allows the original
base stations to be maintained even in the new system layout. However, new base
ll
stations will have to be added for new cells that do not lie in the center of the larger
cells. The organization of cells into clusters is independent of the cell radius, so that
A
the cluster size can be the same in the small-cell layout as it was in the large-cell
layout. Also the signal-to-interference ratio is determined by cluster size and not by
cell radius. Consequently, if the cluster size is maintained, the signal-to-interference
ratio will be the same after cell splitting as it was before. If the entire system is
22
replaced with new half-radius cells, and the cluster size is maintained, the number
of channels per cell will be exactly as it was before, and the number of subscribers
per cell will have been reduced.
When the cell radius is reduced by a factor, it is also desirable to reduce the
transmitted power. The transmit power of the new cells with radius half that of the
old cells can be found by examining the received power PR at the new and old cell
ld
boundaries and setting them equal. This is necessary to maintain the same frequency
re-use plan in the new cell layout as well. Assume that PT1 and PT2 are the transmit
powers of the larger and smaller base stations respectively. Then, assuming a path
or
loss index n=4, we have power received at old cell boundary = PT 1/R4 and the
power received at new cell boundary = PT 2 /(R/2)4 . On equating the two received
powers, we get PT 2 = PT 1 / 16. In other words, the transmit power must be reduced
by 12 dB in order to maintain the same S/I with the new system lay-out.
W
At the beginning of this channel splitting process, there would be fewer channels
in the smaller power groups. As the demand increases, more and more channels need
to be accommodated and hence the splitting process continues until all the larger
cells have been replaced by the smaller cells, at which point splitting is complete
TU
within the region and the entire system is rescaled to have a smaller radius per cell.
If a cellular layout is replaced entirety by a new layout with a smaller cell radius, the
signal-to-interference ratio will not change, provided the cluster size does not
change. Some special care must be taken, however, to avoid co-channel interference
when both large and small cell radii coexist. It turns out that the only way to
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avoid interference between the large-cell and small-cell systems is to assign entirely
different sets of channels to the two systems. So, when two sizes of cells co-exist in
a system, channels in the old cell must be broken down into two groups, one that
corresponds to larger cell reuse requirements and the other which corresponds to the
smaller cell reuse requirements. The larger cell is usually dedicated to high speed
ll
Ex. 4: When the AMPS cellular system was first deployed, the aim of the
A
system designers was to guarantee coverage. Initially the number of users was not
significant. Consequently cells were configured with an eight-mile radius, and a 12-
cell cluster size was chosen. The cell radius was chosen to guarantee a 17 dB
23
ld
Figure 3.8: A cell divided into three 120 o sectors.
or
signal-to-noise ratio over 90% of the coverage area. Although a 12-cell cluster size
provided more than adequate co-channel separation to meet a requirement for a
17 dB signal-to-interference ratio in an interference-limited environment, it did not
W
provide adequate frequency reuse to service an explosively growing customer base.
The system planners reasoned that a subsequent shift to a 7-cell cluster size would
provide an adequate number of channels. It was estimated that a 7-cell cluster
size should provide an adequate 18.7 dB signal-to-interference ratio. The margin,
TU
however, is slim, and the 17 dB signal-to-interference ratio requirement could not
be met over 90 % of the coverage area.
Sectoring
Sectoring is basically a technique which can increase the SIR without necessitating
JN
an increase in the cluster size. Till now, it has been assumed that the base station is
located in the center of a cell and radiates uniformly in all the directions behaving as
an omni-directional antenna. However it has been found that the co-channel inter-
ference in a cellular system may be decreased by replacing a single omni-directional
antenna at the base station by several directional antennas, each radiating within a
ll
specified sector. In the Figure 3.8, a cell is shown which has been split into three
120o sectors. The base station feeds three 120o directional antennas, each of which
A
radiates into one of the three sectors. The channel set serving this cell has also been
divided, so that each sector is assigned one-third of the available number cell of
channels. This technique for reducing co-channel interference wherein by using suit-
24
ld
Figure 3.9: A seven-cell cluster with 60o sectors.
or
able directional antennas, a given cell would receive interference and transmit with
a fraction of available co-channel cells is called ’sectoring’. In a seven-cell-cluster
layout with 120o sectored cells, it can be easily understood that the mobile units in
W
a particular sector of the center cell will receive co-channel interference from only
two of the first-tier co-channel base stations, rather than from all six. Likewise, the
base station in the center cell will receive co-channel interference from mobile units
in only two of the co-channel cells. Hence the signal to interference ratio is now
TU
modified to √ n
S ( 3N ) (3.22)
I = 2
where the denominator has been reduced from 6 to 2 to account for the reduced
number of interfering sources. Now, the signal to interference ratio for a seven-cell
cluster layout using 120o sectored antennas can be found from equation (3.24) to be
JN
23.4 dB which is a significant improvement over the Omni-directional case where the
worst-case S/I is found to be 17 dB (assuming a path-loss exponent, n=4). Some
cellular systems divide the cells into 60o sectors. Similar analysis can be performed
on them as well.
Ex. 5: A cellular system having a seven-cell cluster layout with omni-directional
ll
antennas has been performing satisfactorily for a required signal to interference ratio
of 15 dB. However due to the need for increasing the number of available channels, a
A
60o sectoring of the cells has been introduced. By what percentage can the number
of channels Ntotal be increased assuming a path-loss component n=4?
Solution: The seven-cell cluster layout with 60o sectoring is shown in the Figure 3.9.
25
It is easy to see that the shaded region in the center receives interference from just
one first-tier cell and hence the signal to interference ratio can be obtained suitably
as
√ n
, 4
S ( 3N ) ( (3)(7)) = 26.4dB. (3.23)
I = 1 = 1
Since the SIR exceeds 15 dB, one can try reducing the cluster size from seven to
ld
four. Now, the SIR for this reduced cluster size layout can be found to be
√ n
, 4
S ( 3N ) ( (3)(4)) = 21.6dB. (3.24)
I = 1 = 1
The S/I ratio is still above the requirement and so a further reduction in the cell
or
cluster size is possible. For a 3-cell cluster layout, there are two interfering sources
and hence the S/I ratio is found to be
√ √
3N )
n
( 33)4
S ( = = 16.07dB. (3.25)
I = 1
W 2
This is just above the adequate S/I ratio and further reduction in cluster size is
not possible. So, a 3-cluster cell layout could be used for meeting the growth re-
quirements. Thus, when the cluster size is reduced from 7 to 3, the total number of
TU
smaller sets, thus reducing the trunking efficiency. Moreover, dividing a cell into
sectors requires that a call in progress will have to be handed off (that is, assigned
A
a new channel) when a mobile unit travels into a new sector. This increases the
complexity of the system and also the load on the mobile switching center/base
station.
26
3.7.4 Microcell Zone Concept
ld
of the three zone sites connected to the base station and sharing the same radio
equipment. It is necessary to note that all the microcell zones, within a cell, use the
same frequency used by that cell; that is no handovers occur between microcells.
Thus when a mobile user moves between two microcell zones of the cell, the BS
or
simply switches the channel to a different zone site and no physical re-allotment of
channel takes place.
Locating the mobile unit within the cell: An active mobile unit sends a signal to all
strongest signal. W
zone sites, which in turn send a signal to the BS. A zone selector at the BS uses that
signal to select a suitable zone to serve the mobile unit - choosing the zone with the
Base Station Signals: When a call is made to a cellular phone, the system already
knows the cell location of that phone. The base station of that cell knows in which
TU
zone, within that cell, the cellular phone is located. Therefore when it receives the
signal, the base station transmits it to the suitable zone site. The zone site receives
the cellular signal from the base station and transmits that signal to the mobile
phone after amplification. By confining the power transmitted to the mobile phone,
JN
co-channel interference is reduced between the zones and the capacity of system is
increased.
Benefits of the micro-cell zone concept: 1) Interference is reduced in this case as
compared to the scheme in which the cell size is reduced.
2) Handoffs are reduced (also compared to decreasing the cell size) since the micro-
cells within the cell operate at the same frequency; no handover occurs when the
ll
27
ld
or
W
Figure 3.10: The micro-cell zone concept.
the signal power is reduced, the microcells can be closer and result in an increased
TU
surrounding (a new building, say, within a microcell) will require a change of the
transmission power.
In the previous sections, we have discussed the frequency reuse plan, the design
ll
trade-offs and also explored certain capacity expansion techniques like cell-splitting
and sectoring. Now, we look at the relation between the number of radio channels
A
a cell contains and the number of users a cell can support. Cellular systems use
the concept of trunking to accommodate a large number of users in a limited radio
spectrum. It was found that a central office associated with say, 10,000 telephones
28
requires about 50 million connections to connect every possible pair of users. How-
ever, a worst case maximum of 5000 connections need to be made among these
telephones at any given instant of time, as against the possible 50 million connec-
tions. In fact, only a few hundreds of lines are needed owing to the relatively short
duration of a call. This indicates that the resources are shared so that the number of
lines is much smaller than the number of possible connections. A line that connects
ld
switching offices and that is shared among users on an as-needed basis is called a
trunk.
The fact that the number of trunks needed to make connections between offices
or
is much smaller than the maximum number that could be used suggests that at
times there might not be sufficient facilities to allow a call to be completed. A call
that cannot be completed owing to a lack of resources is said to be blocked. So one
important to be answered in mobile cellular systems is: How many channels per cell
In a trunked radio system, a channel is allotted on per call basis. The perfor-
mance of a radio system can be estimated in a way by looking at how efficiently the
TU
calls are getting connected and also how they are being maintained at handoffs.
Some of the important factors to take into consideration are (i) Arrival statistics,
(ii)Service statistics, (iii)Number of servers/channels.
Let us now consider the following assumptions for a bufferless system handling ’L’
users as shown in Figure 3.11:
JN
µ 2.
ll
(v) The channel holding rate therefore is exponentially distributed with a parameter
µ = µ1 + µ2.
A
29
ld
or
W
Figure 3.11: The bufferless J-channel trunked radio system.
TU
JN
ll
Figure 3.12: Discrete-time Markov chain for the M/M/J/J trunked radio system.
A
30
standard queuing theory nomenclature whereby:
(i) the first letter indicates the nature of arrival process(e.g. M stands for memory-
less which here means a Poisson process).
(ii) the second letter indicates the nature of probability distribution of service
times.(e.g M stands for exponential distribution). In all cases,successive inter ar-
rival times and service times are assumed to be statistically independent of each
ld
other.
(iii) the third letter indicates the number of servers.
(iv) the last letter indicates that if an arrival finds all ’m’ users to be busy, then it
or
will not enter the system and is lost.
In view of the above, the bufferless system as shown in Figure 3.11 can be modeled
as M/M/J/J system and the discrete-time Markov chain of this system is shown in
Figure 3.12.
W
Trunking mainly exploits the statistical behavior of users so that a fixed number
of channels can be used to accommodate a large, random user community. As the
number of telephone lines decrease, it becomes more likely that all channels are
busy for a particular user. As a result, the call gets rejected and in some systems,
TU
a queue may be used to hold the caller’s request until a channel becomes available.
In the telephone system context the term Grade of Service (GoS) is used to mean
the probability that a user’s request for service will be blocked because a required
facility, such as a trunk or a cellular channel, is not available. For example, a GoS of
2 % implies that on the average a user might not be successful in placing a call on 2
JN
out of every 100 attempts. In practice the blocking frequency varies with time. One
would expect far more call attempts during business hours than during the middle of
the night. Telephone operating companies maintain usage records and can identify a
”busy hour”, that is, the hour of the day during which there is the greatest demand
for service. Typically, telephone systems are engineered to provide a specified grade
ll
of call requests per unit time λuser and the average holding time H. The parameter
λuser is also called the average arrival rate, referring to the rate at which calls from
a single user arrive. The average holding time is the average duration of a call. The
31
product:
Auser = λuser H (3.26)
that is, the product of the average arrival rate and the average holding time–is called
the offered traffic intensity or offered load. This quantity represents the average
traffic that a user provides to the system. Offered traffic intensity is a quantity that
is traditionally measured in Erlangs. One Erlang represents the amount of traffic
ld
intensity carried by a channel that is completely occupied. For example, a channel
that is occupied for thirty minutes during an hour carries 0.5 Erlang of traffic.
Call arrivals or requests for service are modeled as a Poisson random process. It
or
is based on the assumption that there is a large pool of users who do not cooperate
in deciding when to place calls. Holding times are very well predicted using an
exponential probability distribution. This implies that calls of long duration are
W
much less frequent than short calls. If the traffic intensity offered by a single user is
Auser , then the traffic intensity offered by N users is A = NAuser . The purpose of
the statistical model is to relate the offered traffic intensity A, the grade of service
Pb, and the number of channels or trunks C needed to maintain the desired grade
of service.
TU
Two models are widely used in traffic engineering to represent what happens
when a call is blocked. The blocked calls cleared model assumes that when a channel
or trunk is not available to service an arriving call, the call is cleared from the
system. The second model is known as blocked calls delayed. In this model a call
that cannot be serviced is placed on a queue and will be serviced when a channel or
JN
K n /n!
n=0 A
When the blocked-calls-delayed model is used, the ”grade of service” refers to the
A
probability that a call will be delayed. In this case the statistical model leads to the
Erlang C formula,
AK /[(K A)(K 1)]!
P [delay] = − − . (3.28)
.
A /[(K − A)(K − 1)]! + K
K
n=0 A n /n!
32
Ex. 6: In a certain cellular system, an average subscriber places two calls per
hour during a busy hour and the average holding time is 3 min. Each cell has 100
channels. If the blocked calls are cleared, how many subscribers can be serviced by
each cell at 2 % GoS?
Solution: Using Erlang B table, it can be seen that for C = 100 and ttoS = Pb = 2%,
ld
the total offered load A=87.972 Erlangs. Since an individual subscriber offers a load
of Auser = (2 calls / 60 min)3 min = 0.1 Erlang, the maximum number of subscribers
served is
or
N = A/Auser = 87.972/0.1 ≈ 880. (3.29)
Ex. 4: In the previous example, suppose that the channels have been divided into
two groups of 50 channels each. Each subscriber is assigned to a group and can be
served only by that group. How many subscribers can be served by the two group
cell?
A = 40.255Erlangs (3.30)
TU
Thus the maximum number of users per group is
Thus, counting both the groups, maximum number of users in the two group cell is
JN
806.
The above example indicates that the number of subscribers that can be sup-
ported by a given number of channels decreases as the pool of channels is sub-divided.
We can express this in terms of the trunking efficiency, defined as the carrier load
per channel, that is,
ξ = (1 − Pb )A/C. (3.32)
ll
This explains why the sectoring of a cell into either 120o or 60o sectors reduces
the trunking efficiency of the system. Thus the system growth due to sectoring is
A
33
References
ld
3. S. Haykin and M. Moher, Modern Wireless Communications. Singapore: Pear-
son Education, Inc., 2002.
or
4. J. W. Mark and W. Zhuang, Wireless Communications and Networking. New
Delhi: PHI, 2005.
W
TU
JN
ll
A
34
Chapter 4
ld
Free Space Radio Wave
or
Propagation
Introduction
W
There are two basic ways of transmitting an electro-magnetic (EM) signal, through a
guided medium or through an unguided medium. Guided mediums such as coaxial
cables and fiber optic cables, are far less hostile toward the information carrying
TU
EM signal than the wireless or the unguided medium. It presents challenges and
conditions which are unique for this kind of transmissions. A signal, as it travels
through the wireless channel, undergoes many kinds of propagation effects such as
reflection, diffraction and scattering, due to the presence of buildings, mountains and
JN
other such obstructions. Reflection occurs when the EM waves impinge on objects
which are much greater than the wavelength of the traveling wave. Diffraction
is a phenomena occurring when the wave interacts with a surface having sharp
irregularities. Scattering occurs when the medium through the wave is traveling
contains objects which are much smaller than the wavelength of the EM wave.
These varied phenomena’s lead to large scale and small scale propagation losses. Due
ll
to the inherent randomness associated with such channels they are best described
with the help of statistical models. Models which predict the mean signal strength
A
for arbitrary transmitter receiver distances are termed as large scale propagation
models. These are termed so because they predict the average signal strength for
large Tx-Rx separations, typically for hundreds of kilometers.
35
ld
or
Figure 4.1: Free space propagation model, showing the near and far fields.
W
Although EM signals when traveling through wireless channels experience fading
effects due to various effects, but in some cases the transmission is with a direct
line of sight such as in satellite communication. Free space model predicts that
the received power decays as negative square root of the distance. Friis free space
TU
equation is given by
Pttttttrλ2
Pr (d) = (4.1)
(4π)2d2L
where Pt is the transmitted power, Pr(d) is the received power, ttt is the transmitter
antenna gain, ttr is the receiver antenna gain, d is the Tx-Rx separation and L is the
system loss factor depended upon line attenuation, filter losses and antenna losses
JN
and not related to propagation. The gain of the antenna is related to the effective
aperture of the antenna which in turn is dependent upon the physical size of the
antenna as given below
tt = 4πAe /λ2. (4.2)
The path loss, representing the attenuation suffered by the signal as it travels
ll
through the wireless channel is given by the difference of the transmitted and re-
ceived power in dB and is expressed as:
A
36
The fields of an antenna can broadly be classified in two regions, the far field and
the near field. It is in the far field that the propagating waves act as plane waves
and the power decays inversely with distance. The far field region is also termed
as Fraunhofer region and the Friis equation holds in this region. Hence, the Friis
equation is used only beyond the far field distance, df , which is dependent upon the
largest dimension of the antenna as
ld
df = 2D2/λ. (4.4)
Also we can see that the Friis equation is not defined for d=0. For this reason, we
or
use a close in distance, do, as a reference point. The power received, Pr(d), is then
given by:
Pr (d) = Pr (do)(do/d)2. (4.5)
Ex. 1: Find the far field distance for a circular antenna with maximum dimension
λ=
W
of 1 m and operating frequency of 900 MHz.
Solution: Since the operating frequency f = 900 Mhz, the wavelength
3 × 108m/s m
900 10 6 Hz
×
TU
. Thus, with the largest dimension of the antenna, D=1m, the far field distance is
2D2 2(1)2
df = = = 6m
λ 0.33
.
Ex. 2: A unit gain antenna with a maximum dimension of 1 m produces 50 W
JN
power at 900 MHz. Find (i) the transmit power in dBm and dB, (ii) the received
power at a free space distance of 5 m and 100 m.
Solution:
(i) Tx power = 10log(50) = 17 dB = (17+30) dBm = 47 dBm
2
(ii) df = 2×D2 = 2 ×1 = 6m
λ 1/3
ll
Thus the received power at 5 m can not be calculated using free space distance
formula.
A
At 100 m ,
PT ttT ttRλ2
PR =
4πd2
50 × 1 × (1/3)2
=
4π1002
37
= 3.5 × 10−3 mW
PR (dBm) = 10logPr(mW ) = −24.5dBm
Reflection, diffraction and scattering are the three fundamental phenomena that
ld
cause signal propagation in a mobile communication system, apart from LoS com-
munication. The most important parameter, predicted by propagation models based
on above three phenomena, is the received power. The physics of the above phe-
nomena may also be used to describe small scale fading and multipath propagation.
or
The following subsections give an outline of these phenomena.
Reflection
W
Reflection occurs when an electromagnetic wave falls on an object, which has very
large dimensions as compared to the wavelength of the propagating wave. For ex-
ample, such objects can be the earth, buildings and walls. When a radio wave falls
on another medium having different electrical properties, a part of it is transmitted
into it, while some energy is reflected back. Let us see some special cases. If the
TU
medium on which the e.m. wave is incident is a dielectric, some energy is reflected
back and some energy is transmitted. If the medium is a perfect conductor, all
energy is reflected back to the first medium. The amount of energy that is reflected
back depends on the polarization of the e.m. wave.
JN
Further, considering perfect conductors, the electric field inside the conductor is
always zero. Hence all energy is reflected back. Boundary conditions require that
A
θi = θr (4.7)
and
Ei = Er (4.8)
57
for vertical polarization, and
Ei = −Er (4.9)
Diffraction
ld
Diffraction is the phenomenon due to which an EM wave can propagate beyond the
horizon, around the curved earth’s surface and obstructions like tall buildings. As
the user moves deeper into the shadowed region, the received field strength decreases.
But the diffraction field still exists an it has enough strength to yield a good signal.
or
This phenomenon can be explained by the Huygen’s principle, according to which,
every point on a wavefront acts as point sources for the production of sec- ondary
wavelets, and they combine to produce a new wavefront in the direction of
W
propagation. The propagation of secondary wavelets in the shadowed region results
in diffraction. The field in the shadowed region is the vector sum of the electric field
components of all the secondary wavelets that are received by the receiver.
Scattering
TU
The actual received power at the receiver is somewhat stronger than claimed by the
models of reflection and diffraction. The cause is that the trees, buildings and lamp-
posts scatter energy in all directions. This provides extra energy at the receiver.
Roughness is tested by a Rayleigh criterion, which defines a critical height hc of
JN
πσhsinθ i 2
ρS = exp(−8( ) ) (4.11)
λ
A
where σh is the standard deviation of the Gaussian random variable h. The following
result is a better approximation to the observed value
πσ hsinθ i 2 πσhsinθ i 2
ρS = exp(−8( ) )I0[−8( ) ] (4.12)
λ λ
58
ld
or
W
Figure 4.2: Two-ray reflection model.
which agrees very well for large walls made of limestone. The equivalent reflection
TU
coefficient is given by,
Γrough = ρS Γ. (4.13)
of incidence and frequency of the wave. For example, as the EM waves can not pass
through conductors, all the energy is reflected back with angle of incidence equal to
the angle of reflection and reflection coefficient Γ = −1. In general, for parallel and
A
59
Γ⊥ = Er/Ei = η2 sin θi − η1 sin θt/η2 sin θi + η1 sin θt. (4.15)
Seldom in communication systems we encounter channels with only LOS paths and
hence the Friis formula is not a very accurate description of the communication link.
A two-ray model, which consists of two overlapping waves at the receiver, one direct
path and one reflected wave from the ground gives a more accurate description as
ld
shown in Figure 4.2. A simple addition of a single reflected wave shows that power
varies inversely with the forth power of the distance between the Tx and the Rx.
This is deduced via the following treatment. From Figure 4.2, the total transmitted
and received electric fields are
or
ETT OT = Ei + ELOS , (4.16)
where
W
Let E0 is the free space electric field (in V/m) at a reference distance d0. Then
E d
E(d, t) = 0 0 cos(ω t − φ)
d
d
c
(4.18)
φ = ωc (4.19)
TU
c
and d > d0. The envelop of the electric field at d meters from the transmitter at
any time t is therefore
E0d0
|E(d, t)| = . (4.20)
d
This means the envelop is constant with respect to time.
JN
Two propagating waves arrive at the receiver, one LOS wave which travels a
t tt
distance of d and another ground reflected wave, that travels d . Mathematically,
it can be expressed as:
t E0 d0 t
E(d , t) = cos(ω t − φ ) (4.21)
c
dt
where
ll
t t
φ = ωc d (4.22)
c
and
E0 d0
A
tt tt
E(d , t) = cos(ω t − φ ) (4.23)
tt c
d
where
tt
tt d
φ = ωc . (4.24)
c
60
ld
Figure 4.3: Phasor diagram of electric Figure 4.4: Equivalent phasor diagram of
Figure 4.3.
or
fields.
i.e.,
W
For small values of θ i, reflected wave is equal in magnitude and 180o out of phase
with respect to incident wave. Assuming perfect horizontal electric field polarization,
Γ⊥ = −1 =⇒ Et = (1 − 1)Ei = 0, (4.26)
TU
the resultant electric field is the vector sum of ELOS and Eg . This implies that,
ET OT E0d0 t E0d0 tt
However, when T-R separation distance is very large compared to (ht + hr), then
ll
2hthr
∆ ≈ (4.30)
d
A
Ex 3: Prove the above two equations, i.e., equation (4.29) and (4.30).
Once the path difference is known, the phase difference is
∆ωc
θ∆ = 2π∆ = (4.31)
λ λ
61
and the time difference,
∆ θ∆
τd = = . (4.32)
c 2πfc
When d is very large, then ∆ becomes very small and therefore ELOS and Eg are
virtually identical with only phase difference,i.e.,
E0d0 E0d0 E0d0
| |≈| | ≈| |. (4.33)
ld
t
d d dtt
tt
Say, we want to evaluate the received E-field at any t = dc . Then,
t
tt E0d0 tt d E 0 d0 tt tt
ET OT d d d d
(d, t = ) = dt cos(ω c − ωc c ) − tt cos(ωc − ωc )
c (4.34)
or
R
c E d ∆ωc d c c
= 0 0 cos( ) E0d0
t − cos(0o) (4.35)
d c dtt
E0d0
E 0 d0 ƒ
= θ∆ − (4.36)
dt dtt
we get
|ET OT
R (d)| =
.
( +
E0d0
≈ d (ƒ θ∆ − 1).
E0d0 E0d0
cos(θ ∆
W
Using phasor diagram concept for vector addition as shown in Figures 4.3 and 4.4,
2
)) + (
E0d0
sin(θ ∆))
2
(4.38)
(4.37)
d d d
TU
E0 d 0 . 2 2
= (cos(θ ) − 1) + sin (θ ∆ ) (4.39)
∆
d
E0 d 0 ,
=
2 − 2cosθ (4.40)
∆
d
E0 d0 θ
=2 sin( ∆ ). (4.41)
d 2
JN
For θ∆ < 0.5rad, sin( θ∆ ) ≈ θ∆ . Using equation (4.31) and further equation (4.30),
2 2 2
we can then approximate that
θ∆ π
sin( ) ≈ ∆ = 2πht hr < 0.5rad. (4.42)
2 λ λd
4πhthr
d > dc = 20πhthr = . (4.43)
5λ λ
A
62
Therefore, using equation (4.43) in (4.1), we get the received power as
Pttttttr h2th2r
Pr =
. (4.45)
Ld4
The cross-over distance shows an approximation of the distance after which the
received power decays with its fourth order. The basic difference between equation
(4.1) and (4.45) is that when d < dc , equation (4.1) is sufficient to calculate the
ld
path loss since the two-ray model does not give a good result for a short distance
due to the oscillation caused by the constructive and destructive combination of the
two rays, but whenever we distance crosses the ‘cross-over distance’, the power falls
or
off rapidly as well as two-ray model approximation gives better result than Friis
equation.
Observations on Equation (4.45): The important observations from this
equation are:
cross-over distance. W
1. This equation gives fair results when the T-R separation distance crosses the
Diffraction
Diffraction is the phenomena that explains the digression of a wave from a straight
line path, under the influence of an obstacle, so as to propagate behind the obstacle.
ll
without having any line of sight. The similar phenomena occurs for light also but
the diffracted light intensity is not noticeable. This is because the obstacle or slit
need to be of the order of the wavelength of the wave to have a significant effect.
Thus radiation from a point source radiating in all directions can be received at any
63
ld
or
Figure 4.5: Huygen’s secondary wavelets.
W
point, even behind an obstacle (unless it is not completely enveloped by it), as shown
in Figure 4.5. Though the intensity received gets smaller as receiver is moved into the
shadowed region. Diffraction is explained by Huygens-Fresnel principle which states
that all points on a wavefront can be considered as the point source for secondary
wavelets which form the secondary wavefront in the direction of the prorogation.
TU
Normally, in absence of an obstacle, the sum of all wave sources is zero at a point
not in the direct path of the wave and thus the wave travels in the straight line. But
in the case of an obstacle, the effect of wave source behind the obstacle cannot be
felt and the sources around the obstacle contribute to the secondary wavelets in the
JN
64
ld
or
Figure 4.6: Diffraction through a sharp edge.
1
W
δ = d1 (1 + h2 /2d2 ) + d2 (1 + h2 /2d2 ) − (d1 + d2 )
2
α=β +γ (4.50)
and
α ≈ tanα (4.51)
we can write,
ll
v, expressed as
. .
v = h 2(d1 + d2 )/(λd1 d2 ) = α (2d1 d2 )/(λ(d1 + d2 )) (4.53)
65
ld
or
W
TU
φ = πv 2 /2. (4.54)
JN
From this, we can observe that: (i) phase difference is a function of the height of
the obstruction, and also, (ii) phase difference is a function of the position of the
obstruction from transmitter and receiver.
As mentioned before, the more is the object in the shadowed region greater is the
diffraction loss of the signal. The effect of diffraction loss is explained by Fresnel
A
zones as a function of the path difference. The successive Fresnel zones are limited
by the circular periphery through which the path difference of the secondary waves
is nλ/2 greater than total length of the LOS path, as shown in Figure 4.7. Thus
successive Fresnel zones have phase difference of π which means they alternatively
66
provide constructive and destructive interference to the received the signal. The
radius of the each Fresnel zone is maximum at middle of transmitter and receiver
(i.e. when d1 = d2 ) and decreases as moved to either side. It is seen that the loci
of a Fresnel zone varied over d1 and d2 forms an ellipsoid with the transmitter and
receiver at its focii. Now, if there’s no obstruction, then all Fresnel zones result in
only the direct LOS prorogation and no diffraction effects are observed. But if an
ld
obstruction is present, depending on its geometry, it obstructs contribution from
some of the secondary wavelets, resulting in diffraction and also the loss of energy,
which is the vector sum of energy from unobstructed sources. please note that height
or
of the obstruction can be positive zero and negative also. The diffraction losses are
minimum as long as obstruction doesn’t block volume of the 1st Fresnel zone. As a
rule of thumb, diffraction effects are negligible beyond 55% of 1st Fresnel zone.
Ex 4: Calculate the first Fresnel zone obstruction height maximum for f = 800
MHz.
Solution:
λ=
c
f
=
W 3
×
108 3
8 × 102 × 106 = 8 m
.
TU
λ(d1+d2 )
H= . d1+d2
3
250×250
H1 = 8
= 6.89m
500
Thus H1 = 10 + 6.89 = 16.89m
JN
(b)
. 100 × 400 .
3 ×
H2 = 8 = 10 (0.3) = 5.48m
500
Thus
H2 = 10 + 5.6 = 15.48m
. To have good power strength, obstacle should be within the 60% of the first fresnel
ll
zone.
A
67
ld
or
Figure 4.8: Knife-edge Diffraction Model
Given,
W
ttd(dB) = 20 log(0.5 − 0.62v)
v > 2.24
TU
Solution: . .
2(d1 + d 2) 2 × 2000
v=h = 2.74
λd1d2 = 25 1
3 10
Gd(dB) = 20 log( 225 ) = −21.7dB
v
JN
(2.74)2
n= = 3.5
2
ll
Thus n=4.
Knife-edge diffraction model is one of the simplest diffraction model to estimate the
diffraction loss. It considers the object like hill or mountain as a knife edge sharp
68
object. The electric field strength, Ed of a knife-edge diffracted wave is given by
¸ ∞
Ed/Eo = F (v) = (1 + j)/2 (exp((−jπt2 )/2)dt. (4.55)
v
ld
ttd (db) = 0v <= −1 (4.57)
or
ttd(db) = 20log(0.5exp(−0.95v)) 0 <= v <= 1 (4.59)
W
When there are more than one obstruction, then the equivalent model can be found
by one knife-edge diffraction model as shown in Figure 4.8.
The value of n varies with propagation environments. The value of n is 2 for free
space. The value of n varies from 4 to 6 for obstruction of building, and 3 to 5 for
urban scenarios. The important factor is to select the correct reference distance d0.
For large cell area it is 1 Km, while for micro-cell system it varies from 10m-1m.
Limitations:
ll
69
Log Normal Shadowing
ld
deviation σ also in dB. In practice n and σ values are computed from measured
data.
or
The ‘Q’ function is given by,
z
Q(z) = 0.5(1 − erf ( √ )) (4.64)
2
and
γ is
W
Q(z) = 1 − Q(−z)
γ − Pr
(4.65)
So the probability that the received signal level (in dB) will exceed a certain value
TU
P (Pd > γ) = Q( ). (4.66)
σ
There are many empirical outdoor propagation models such as Longley-Rice model,
JN
Durkin’s model, Okumura model, Hata model etc. Longley-Rice model is the most
commonly used model within a frequency band of 40 MHz to 100 GHz over different
terrains. Certain modifications over the rudimentary model like an extra urban
factor (UF) due to urban clutter near the reciever is also included in this model.
Below, we discuss some of the outdoor models, followed by a few indoor models too.
ll
Okumura Model
The Okumura model is used for Urban Areas is a Radio propagation model that is
A
used for signal prediction.The frequency coverage of this model is in the range of
200 MHz to 1900 MHz and distances of 1 Km to 100 Km.It can be applicable for
base station effective antenna heights (ht) ranging from 30 m to 1000 m.
70
Okumura used extensive measurements of base station-to-mobile signal attenua-
tion throughout Tokyo to develop a set of curves giving median attenuation relative
to free space (A mu) of signal propogation in irregular terrain. The empirical path-
loss formula of Okumura at distance d parameterized by the carrier frequency fc is
given by
ld
PL(d)dB = L(fc , d) + Amu (fc , d) − tt(ht) − tt(hr) − ttAREA (4.67)
where L(fc, d) is free space path loss at distance d and carrier frequency fc , A mu(fc , d)
is the median attenuation in addition to free-space path loss across all environments,tt(ht)
or
is the base station antenna height gain factor,tt(hr) is the mobile antenna height gain
factor,ttAREA is the gain due to type of environment. The values of Amu(fc, d) and
ttAREA are obtained from Okumura’s empirical plots. Okumura derived empirical
formulas for tt(ht) and tt(hr) as follows:
W
tt(ht ) = 20 log10 (ht /200), 30m < ht < 1000m
(4.69)
(4.70)
TU
Correlation factors related to terrain are also developed in order to improve the
models accuracy. Okumura’s model has a 10-14 dB empirical standard deviation
between the path loss predicted by the model and the path loss associated with one
of the measurements used to develop the model.
JN
Hata Model
The Hata model is an empirical formulation of the graphical path-loss data provided
by the Okumura and is valid over roughly the same range of frequencies, 150-1500
MHz. This empirical formula simplifies the calculation of path loss because it is
closed form formula and it is not based on empirical curves for the different param-
ll
eters. The standard formula for empirical path loss in urban areas under the Hata
model is
A
PL,urban (d)dB = 69.55+26.16 log10 (fc )−13.82 log10 (ht )−a(hr )+(44.9−6.55 log10 (ht )) log10 (d)
(4.71)
71
The parameters in this model are same as in the Okumura model,and a(hr) is a
correction factor for the mobile antenna height based on the size of coverage area.For
small to medium sized cities this factor is given by
ld
a(hr) = 3.2(log10(11.75hr))2 − 4.97dB
else it is
a(hr) = 8.29(log10(1.54hr))2 − 1.1dB
or
Corrections to the urban model are made for the suburban, and is given by
Unlike the Okumura model,the Hata model does not provide for any specific path-
W
correlation factors. The Hata model well approximates the Okumura model for
distances d > 1 Km. Hence it is a good model for first generation cellular systems,
but it does not model propogation well in current cellular systems with smaller cell
sizes and higher frequencies. Indoor environments are also not captured by the Hata
TU
model.
The indoor radio channel differs from the traditional mobile radio channel in ways
JN
- the distances covered are much smaller ,and the variability of the environment
is much greater for smaller range of Tx-Rx separation distances.Features such as lay-
out of the building,the construction materials,and the building type strongly in-
fluence the propagation within the building.Indoor radio propagation is dominated
by the same mechanisms as outdoor: reflection, diffraction and scattering with vari-
able conditions. In general,indoor channels may be classified as either line-of-sight
ll
or obstructed.
A
The internal and external structure of a building formed by partitions and obstacles
vary widely.Partitions that are formed as a part of building structure are called
72
hard partitions , and partitions that may be moved and which do not span to
the ceiling are called soft partitions. Partitions vary widely in their physical and
electrical characteristics,making it difficult to apply general models to specific indoor
installations.
ld
The losses between floors of a building are determined by the external dimensions
and materials of the building,as well as the type of construction used to create the
floors and the external surroundings. Even the number of windows in a building
or
and the presence of tinting can impact the loss between floors.
W
It has been observed that indoor path loss obeys the distance power law given by
where n depends on the building and surrounding type, and Xσ represents a normal
random variable in dB having standard deviation of σ dB.
(4.73)
TU
Summary
In this chapter, three principal propagation models have been identified: free-space
propagation, reflection and diffraction, which are common terrestrial models and
JN
these mainly explains the large scale path loss. Regarding path-loss, one important
factor introduced in this chapter is log-distance path loss model. These, however,
may be insignificant when we consider the small-scale rapid path losses. This is
discussed in the next chapter.
ll
References
73
3. J. W. Mark and W. Zhuang, Wireless Communications and Networking. New
Delhi: PHI, 2005.
ld
or
W
TU
JN
ll
A
74
Chapter 5
ld
Multipath Wave Propagation
or
and Fading
Multipath Propagation
W
In wireless telecommunications, multipath is the propagation phenomenon that re-
sults in radio signals reaching the receiving antenna by two or more paths. Causes
of multipath include atmospheric ducting, ionospheric reflection and refraction, and
TU
reflection from water bodies and terrestrial objects such as mountains and buildings.
The effects of multipath include constructive and destructive interference, and phase
shifting of the signal. In digital radio communications (such as GSM) multipath can
cause errors and affect the quality of communications. We discuss all the related
JN
Multipath signals are received in a terrestrial environment, i.e., where different forms
of propagation are present and the signals arrive at the receiver from transmitter via
ll
clutter to it, the received overall signal amplitude or phase changes over a small
amount of time. Mainly this causes the fading.
75
Fading
The term fading, or, small-scale fading, means rapid fluctuations of the amplitudes,
phases, or multipath delays of a radio signal over a short period or short travel
distance. This might be so severe that large scale radio propagation loss effects
might be ignored.
ld
Multipath Fading Effects
or
1. Rapid changes in signal strength over a small travel distance or time interval.
The following physical factors influence small-scale fading in the radio propagation
channel:
TU
(2) Speed of the mobile – The relative motion between the base station and the
mobile results in random frequency modulation due to different doppler shifts
on each of the multipath components.
(3) Speed of surrounding objects – If objects in the radio channel are in mo-
tion, they induce a time varying Doppler shift on multipath components. If
ll
the surrounding objects move at a greater rate than the mobile, then this effect
dominates fading.
A
76
Types of Small-Scale Fading
The type of fading experienced by the signal through a mobile channel depends
on the relation between the signal parameters (bandwidth, symbol period) and the
channel parameters (rms delay spread and Doppler spread). Hence we have four
different types of fading. There are two types of fading due to the time dispersive
ld
nature of the channel.
or
Flat Fading
Such types of fading occurs when the bandwidth of the transmitted signal is less than
the coherence bandwidth of the channel. Equivalently if the symbol period of the
signal is more than the rms delay spread of the channel, then the fading is flat fading.
W
So we can say that flat fading occurs when
BS BC (5.1)
TU
where BS is the signal bandwidth and BC is the coherence bandwidth. Also
TS στ (5.2)
where TS is the symbol period and στ is the rms delay spread. And in such a case,
mobile channel has a constant gain and linear phase response over its bandwidth.
JN
Frequency selective fading occurs when the signal bandwidth is more than the co-
herence bandwidth of the mobile radio channel or equivalently the symbols duration
of the signal is less than the rms delay spread.
ll
BS BC (5.3)
A
and
TS στ (5.4)
77
At the receiver, we obtain multiple copies of the transmitted signal, all attenuated
and delayed in time. The channel introduces inter symbol interference. A rule of
thumb for a channel to have flat fading is if
στ
0.1 (5.5)
≤
TS
ld
Fading Effects due to Doppler Spread
Fast Fading
In a fast fading channel, the channel impulse response changes rapidly within the
or
symbol duration of the signal. Due to Doppler spreading, signal undergoes frequency
dispersion leading to distortion. Therefore a signal undergoes fast fading if
TS TC (5.6)
W
BS BD (5.7)
where BD is the Doppler spread. Transmission involving very low data rates suffer
TU
from fast fading.
Slow Fading
In such a channel, the rate of the change of the channel impulse response is much
less than the transmitted signal. We can consider a slow faded channel a channel in
JN
which channel is almost constant over atleast one symbol duration. Hence
TS TC (5.8)
and
BS BD (5.9)
ll
We observe that the velocity of the user plays an important role in deciding whether
the signal experiences fast or slow fading.
A
78
ld
or
Figure 5.1: Illustration of Doppler effect.
Doppler Shift
W
The Doppler effect (or Doppler shift) is the change in frequency of a wave for an
observer moving relative to the source of the wave. In classical physics (waves in
a medium), the relationship between the observed frequency f and the emitted
TU
frequency fo is given by:
. .
v ± vr
f = f0 (5.10)
v ± vs
where v is the velocity of waves in the medium, vs is the velocity of the source
relative to the medium and vr is the velocity of the receiver relative to the medium.
In mobile communication, the above equation can be slightly changed according
JN
to our convenience since the source (BS) is fixed and located at a remote elevated
level from ground. The expected Doppler shift of the EM wave then comes out to
vr vr
be ± fo or, ± . As the BS is located at an elevated place, a cos φ factor would
c λ
also be multiplied with this. The exact scenario, as given in Figure 5.1, is illustrated
below.
ll
difference in path lengths traveled by the wave from source S to the mobile at points
A and B is ∆l = d cos θ = v∆t cos θ, where ∆t is the time required for the mobile
to travel from A to B, and θ is assumed to be the same at points A and B since the
79
source is assumed to be very far away. The phase change in the received signal due
to the difference in path lengths is therefore
2π∆l 2πv∆t
∆ϕ = cos θ (5.11)
λ = λ
ld
fd = . = . cos θ. (5.12)
2π ∆t λ
Example 1
An aircraft is heading towards a control tower with 500 kmph, at an elevation of 20◦ .
or
Communication between aircraft and control tower occurs at 900 MHz. Find out
the expected Doppler shift.
Solution As given here,
v = 500 kmph
W
the horizontal component of the velocity is
If the plane banks suddenly and heads for other direction, the Doppler shift change
JN
Mobile radio channel may be modeled as a linear filter with time varying impulse
response in continuous time. To show this, consider time variation due to receiver
ll
motion and time varying impulse response h(d, t) and x(t), the transmitted signal.
The received signal y(d, t) at any position d would be
¸
A
∞
y(d, t) = x(t) ∗ h(d, t) = x(τ ) h(d, t − τ ) dτ (5.13)
−∞
¸ ∞
For a causal system: h(d, t) = 0, for t < 0 and for a stable system |h(d, t)| dt <
−∞
80
Applying causality condition in the above equation, h(d, t − τ ) = 0 for t − τ < 0
⇒ τ > t, i.e., the integral limits are changed to
¸ t
y(d, t) = x(τ ) h(d, t − τ ) dτ.
−∞
ld
Since the receiver moves along the ground at a constant velocity v, the position of
the receiver is d = vt, i.e.,
¸ t
y(vt, t) = x(τ ) h(vt, t − τ ) dτ.
−∞
or
Since v is a constant, y(vt, t) is just a function of t. Therefore the above equation
can be expressed as
¸ t
y(t) = x(τ ) h(vt, t − τ ) dτ = x(t) ∗ h(vt, t) = x(t) ∗ h(d, t) (5.14)
−∞
W
It is useful to discretize the multipath delay axis τ of the impulse response into equal
time delay segments called excess delay bins, each bin having a time delay width
equal to ( τi+1 − τi ) = ∆τ and τi = i∆τ for i ∈ {0, 1, 2, ..N − 1}, where N represents
the total number of possible equally-spaced multipath components, including the
TU
first arriving component. The useful frequency span of the model is 2/∆τ . The
model may be used to analyze transmitted RF signals having bandwidth less than
2/ .
∆τ
If there are N multipaths, maximum excess delay is given by N ∆τ .
JN
1 1
c(t) → hb(t, τ ) → r(t) = c(t) ∗ hb(t, τ ) (5.17)
2 2
A
Average power is 1
x2(t) = c(t) 2 (5.18)
| |
2
81
The baseband impulse response of a multipath channel can be expressed as
N−1
.
hb(t, τ ) = ai(t, τ ) exp[j(2πfcτi(t) + ϕi(t, τ ))]δ(τ − τi(t)) (5.19)
i=0
where ai(t, τ ) and τi(t) are the real amplitudes and excess delays, respectively, of
the ith multipath component at time t. The phase term 2πfc τ i(t) + ϕi(t, τ ) in the
above equation represents the phase shift due to free space propagation of the ith
ld
multipath component, plus any additional phase shifts which are encountered in the
channel.
If the channel impulse response is wide sense stationary over a small-scale time or
or
distance interval, then
N−1
.
hb(τ ) = ai exp[jθi ]δ(τ − τi ) (5.20)
i=0
W
p(t) ≈ δ(t − τ )
sured using channel sounding techniques. Let us consider two extreme channel
sounding cases.
Consider a pulsed, transmitted RF signal
T bb
is
N−1
.
ai exp[jθi ]p(t − τi )
A
r(t) = 2
i=0 .
N−1 τmax Tb
.
= ai exp[jθi ]. − τi).
Tbb rect(t −
i=0 2
82
ld
or
Figure 5.2: A generic transmitted pulsed RF signal.
|r(t0 )| 2 = 1
τmax
1
τmax
¸
0
τmax
¸
W
r(t)r∗ (t)dt
.N −1 .
1
TU
.
= a2k(t0)p2(t − τk) dt
τmax 4
0 k=0
τmax
1 .
N−1 ¸ .. .2
= a2 (t0) τ max rect(t − Tb − τ i) dt
k
τmax k=0 Tbb 2
0
N−1
.
= a2k(t0).
JN
k=0
. .
2
Ea,θ [PW B ] = Ea,θ [ |ai exp(jθi )| ] ≈ a2i (5.24)
i=0 i=0
Now instead of a pulse, consider a CW signal, transmitted into the same channel
A
83
and the instantaneous power is
N−1
.
|r(t)|2 =| ai exp[jθi(t, τ )] |2 (5.26)
i=0
Over local areas, ai varies little but θ i varies greatly resulting in large fluctuations.
N−1
.
Ea,θ [PCW ] = Ea,θ [ |ai exp(jθi )| 2]
ld
i=0
N−1 N−1 N
. . .
≈ a2i + 2 rij cos(θi − θj )
i=0 i=0 i,jƒ=i
or
where rij = Ea [ai aj ].
If, rij = cos(θi − θj ) = 0, then Ea,θ [PCW ] = Ea,θ [PWB ]. This occurs if multipath
components are uncorrelated or if multipath phases are i.i.d over [0, 2π].
Bottomline:
W
1. If the signal bandwidth is greater than multipath channel bandwidth then
fading effects are negligible
TU
2. If the signal bandwidth is less than the multipath channel bandwidth, large
fading occurs due to phase shift of unresolved paths.
¸ ∞
H(f, t) = F T [h(τ, t)] = h(τ, t)e−j 2πf τ dτ (5.27)
−∞
¸ ∞
h(τ, t) = F T −1 [H(f, t)] = H(f, t)ej2πf τ df (5.28)
−∞
−∞
For flat fading channel, h(τ, t) = Z(t)δ(τ − τi) where Z(t) = αn(t)e−j2πfcτn(t). In
84
ld
or
where the channel becomes multiplicative.
Doppler spread functions:
W
Figure 5.3: Relationship among different channel functions.
¸
TU
∞
H(f, ν) = F T [H(f, t)] = H(f, t)e−j 2πνt dt (5.31)
−∞
and ¸ ∞
H(f, t) = F T −1
[H(f, ν)] = H(f, ν)ej 2πνt dν (5.32)
−∞
¸ ∞
H(τ, ν) = F T [h(τ, t)] = h(τ, t)e−j 2πνt dt (5.33)
−∞
A wideband pulsed bistatic radar usually transmits a repetitive pulse of width Tbb s,
2
and uses a receiver with a wide bandpass filter (BW = T bb
Hz). The signal is then
A
amplified, envelope detected, and displayed and stored on a high speed oscilloscope.
Immediate measurements of the square of the channel impulse response convolved
with the probing pulse can be taken. If the oscilloscope is set on averaging mode,
then this system provides a local average power delay profile.
85
ld
or
W
Figure 5.4: Direct RF pulsed channel IR measurement.
TU
This system is subject to interference noise. If the first arriving signal is blocked
or fades, severe fading occurs, and it is possible the system may not trigger properly.
In this case we measure the channel in the frequency domain and then convert it
into time domain impulse response by taking its inverse discrete Fourier transform
(IDFT). A vector network analyzer controls a swept frequency synthesizer. An S-
parameter test set is used to monitor the frequency response of the channel. The
sweeper scans a particular frequency band, centered on the carrier, by stepping
through discrete frequencies. The number and spacing of the frequency step impacts
ll
the time resolution of the impulse response measurement. For each frequency step,
the S-parameter test set transmits a known signal level at port 1 and monitors
A
the received signal at port 2. These signals allow the analyzer to measure the
complex response, S21(ω), of the channel over the measured frequency range. The
S21(ω) measure is the measure of the signal flow from transmitter antenna to receiver
86
ld
or
W
TU
Figure 5.5: Frequency domain channel IR measurement.
To compare the different multipath channels and to quantify them, we define some
parameters. They all can be determined from the power delay profile. These pa-
ll
These parameters include the mean excess delay,rms delay spread and excess delay
spread. The mean excess delay is the first moment of the power delay profile and is
87
defined as
.
. τ P(τk)τk
τ¯ = .a2k k = . (5.34)
a2 k P(τk)
where ak is the amplitude, τk is the excess delay and P (τk) is the power of the
individual multipath signals.
The mean square excess delay spread is defined as
ld
.
P (τ )τ 2
τ¯2 = . k k (5.35)
P(τk )
Since the rms delay spread is the square root of the second central moment of the
or
power delay profile, it can be written as
.
στ = τ¯2 − (τ̄ )2 (5.36)
As a rule of thumb, for a channel to be flat fading the following condition must be
satisfied
TS W στ
≤
0.1
where TS is the symbol duration. For this case, no equalizer is required at the
receiver.
(5.37)
TU
Example 2
1. Sketch the power delay profile and compute RMS delay spread for the follow-
ing:
1
.
P (τ ) = δ(τ − n × 10−6) (in watts)
n=0
JN
2. If BPSK modulation is used, what is the maximum bit rate that can be sent
through the channel without needing an equalizer?
Solution
(1)(0) + (1)(1)
τ = = 0.5µs
1+1
A
τ 2 = 0.5µs2 στ = 0.5µs
88
ld
1
2. For flat fading channel, we need στ
0.1 ⇒ Rs = = 0.2 × 104 = 200 kbps
Ts Ts
or
Consider a simple worst-case delay spread scenario as shown in figure below.
W
TU
Here dmin = d0 and dmax = di + dr
Transmitted power = PT , Minimum received power = PRmin = PT hreshold
PRmin λ
= ttT ttR ( )2
PT 4πdmax
JN
λ PT 1
dmax = ( )( )2
4π P Rmin
dmax λ PT 1
τmax = =( )( )2
c 4πc PRmin
1 PT 1
τmax =( )( )2
ll
4πf PRmin
To characterize the channel in the frequency domain, we have the following param-
eters.
89
(1) Coherence bandwidth: it is a statistical measure of the range of frequencies
over which the channel can be considered to pass all the frequency components with
almost equal gain and linear phase. When this condition is satisfied then we say the
channel to be flat.
Practically, coherence bandwidth is the minimum separation over which the two
frequency components are affected differently. If the coherence bandwidth is con-
ld
sidered to be the bandwidth over which the frequency correlation function is above
0.9, then it is approximated as
1
BC ≈ . (5.38)
50στ
or
However, if the coherence bandwidth is considered to be the bandwidth over which
the frequency correlation function is above 0.5, then it is defined as
1
. (5.39)
BC ≈
W 5στ
The coherence bandwidth describes the time dispersive nature of the channel in the
local area. A more convenient parameter to study the time variation of the channel
is the coherence time. This variation may be due to the relative motion between the
mobile and the base station or the motion of the objects in the channel.
TU
(2) Coherence time: this is a statistical measure of the time duration over which
the channel impulse response is almost invariant. When channel behaves like this,
it is said to be slow faded. Essentially it is the minimum time duration over which
two received signals are affected differently. For an example, if the coherence time
JN
is considered to be the bandwidth over which the time correlation is above 0.5, then
it can be approximated as
9
TC ≈ (5.40)
16πfm
where fm is the maximum doppler spread given be fm = νλ.
Another parameter is the Doppler spread (BD) which is the range of frequencies
ll
Many multipath models have been proposed to explain the observed statistical na-
ture of a practical mobile channel. Both the first order and second order statistics
90
ld
or
Figure 5.6: Two ray NLoS multipath, resulting in Rayleigh fading.
W
have been examined in order to find out the effective way to model and combat the
channel effects. The most popular of these models are Rayleigh model, which de-
scribes the NLoS propagation. The Rayleigh model is used to model the statistical
time varying nature of the received envelope of a flat fading envelope. Below, we
discuss about the main first order and second order statistical models.
TU
Let there be two multipath signals S1 and S2 received at two different time instants
due to the presence of obstacles as shown in Figure 5.6. Now there can either be
JN
Now if N→ ∞(i.e. are sufficiently large number of multipaths) and all the En are
ll
N →∞
N →∞
n=1
91
where Zr and Zi are Gaussian Random variables. For the above case
.
2 (5.44)
R= Z 2r + Z i
and
Zi
φ = tan−1 (5.45)
Zr
For all practical purposes we assume that the relative phase Θn is uniformaly dis-
ld
tributed. 2π
jθn 1 ¸
E[e ]= ejθ dθ = 0 (5.46)
2π
0
or
It can be seen that En and Θn are independent. So,
.
E[Ẽ] = E[ En ejθn ] = 0 (5.47)
. .2 . . .. .
N
E[. Ẽ . ] = E[ En ejθn E ∗ e−jθn ] = E[ En Em ej (θn −θ m ) ] = E 2 = P0
. . n
W m n
where P0 is the total power obtained. To find the Cumulative Distribution Func-
tion(CDF) of R, we proceed as follows.
¸ ¸
n=1
n
(5.48)
where A is determined by the values taken by the dummy variable r. Let Zi and Zr
be zero mean Gaussian RVs. Hence the CDF can be written as
¸ ¸ −(Z2 +Z 2 )
FR(r) = √ 1 e r
2σ2
i
dZidZr (5.50)
2πσ2
JN
= 1 − e 2σ2 (5.52)
ll
− r2
A
r 2
(5.53)
fR(r) = e 2σ
σ2
and is shown in Figure 5.7 with different σ values. This equation too is valid for all
r ≥ 0. Above distribution is known as Rayleigh distribution and it has been derived
92
ld
or
Figure 5.7: Rayleigh probability density function.
W
E[R] =
.
E[R2 ] = 2σ2
var[R] = (2
π
2
π 2
− 2
)σ
σ (5.54)
(5.55)
(5.56)
TU
median[R] = 1.77σ. (5.57)
Rician Fading is the addition to all the normal multipaths a direct LOS path.
JN
ll
A
93
r −(r2 +A2 ) Ar
2σ2
I( ) (5.58)
fR(r) = e
σ2 σ2 0
for all A ≥ 0 and r ≥ 0. Here A is the peak amplitude of the dominant signal and
I0(.) is the modified Bessel function of the first kind and zeroth order.
A factor K is defined as
A2
ld
K dB = 10 log (5.59)
2σ2
As A → 0 then KdB → ∞.
or
A generalization of the Rayleigh and Rician fading is the Nakagami distribution.
W
TU
2rm−1 mm −mr2
fR(r) = ( )e Ω (5.60)
Γ(m) Ωm
where,
Γ(m) is the gamma function
Ω is the average signal power and
m is the fading factor.It is always greater than or equal to 0.5.
ll
When
(M + 1)2
m=
2M + 1
94
ld
or
Figure 5.10: Schematic representation of level crossing with a Rayleigh fading enve-
lope at 10 Hz Doppler spread.
where
To design better error control codes, we have two important second order param-
eters of fading model, namely the level crossing rate (LCR) and average fade
duration (AFD). These parameters can be utilized to assess the speed of the user
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by measuring them through the reverse channel. The LCR is the expected rate at
which the Rayleigh fading envelope normalized to the local rms amplitude crosses a
specific level ’R’ in a positive going direction.
¸∞ √ 2
NR = ṙp(R, ṙ)dṙ = 2πfD ρe−ρ (5.61)
0
ll
where r˙ is the time derivative of r(t), fD is the maximum Doppler shift and ρ is the
value of the specified level R, normalized to the local rms amplitude of the fading
A
envelope.
The other important parameter, AFD, is the average period time for which the
95
receiver power is below a specified level R.
1
τ¯ = Pr(r ≤ R) (5.62)
Nr
As
¸R
Pr (r ≤ R) = p(r)dr = 1 − e−ρ ,
2
(5.63)
0
ld
therefore, 2
1 − e−ρ
(5.64)
τ¯ = √ D ρe−ρ
2
2πf
2 − 1
e− .ρ (5.65)
= √2πf ρ
or
D
Apart from LCR, another parameter is fading rate, which is defined as the number of
times the signal envelope crosses the middle value (rm) in a positive going direction
per unit time. The average rate is expressed as
W
N (rm ) =
2v
λ
.
In it, two independent Gaussian low pass noise sources are used to produce in-phase
and quadrature fading branches. This is the basic model and is useful for slow fading
channel. Also the Doppler effect is not accounted for.
ll
In this model, the output of the Clarke’s model is passed through Doppler filter in
A
the RF or through two initial baseband Doppler filters for baseband processing as
shown in Figure 5.11. Here, the obtained Rayleigh output is flat faded signal but
not frequency selective.
96
ld
or
W
Figure 5.11: Clarke and Gan’s model for Rayleigh fading generation using quadra-
ture amplitude modulation with (a) RF Doppler filter, and, (b) baseband Doppler
TU
filter.
To get a frequency selective output we have the following simulator through which
both the frequency selective and flat faded Rayleigh signal may be obtained. This
JN
is achieved through varying the parameters ai and τi, as given in Figure 5.12.
The above model is, however, very complex and difficult to implement. So, we have
the two ray Rayleigh fading model which can be easily implemented in software as
ll
where α1 and α2 are independent Rayleigh distributed and φ1 and φ2 are indepen-
dent and uniformaly distributed over 0 to 2π. By varying τ it is possible to create
a wide range of frequency selective fading effects.
97
ld
or
W
TU
Figure 5.12: Rayleigh fading model to get both the flat and frequency selective
channel conditions.
This method involved averaging the square law detected pulse response while sweep-
ing the frequency of the transmitted pulse. The model assumes that the multipath
components arrive in clusters. The amplitudes of the received components are in-
dependent Rayleigh random variables with variances that decay exponentially with
cluster delay as well as excess delay within a cluster. The clusters and multipath
ll
components within a cluster form Poisson arrival processes with different rates.
A
98
ld
Figure 5.13: Two-ray Rayleigh fading model.
or
sequent work by Huang produced SMRCIM (Simulation of Mobile Radio Channel
Impulse-response Model), a similar program that generates small-scale urban cellu-
lar and micro-cellular channel impulse responses.
Conclusion
W
In this chapter, the main channel impairment, i.e., fading, has been introduced which
becomes so severe sometimes that even the large scale path loss becomes insignificant
TU
in comparison to it. Some statistical propagation models have been presented based
on the fading characteristics. Mainly the frequency selective fading, fast fading and
deep fading can be considered the major obstruction from the channel severity view
point. Certain efficient signal processing techniques to mitigate these effects will be
JN
discussed in Chapter 7.
References
99
4. K. Feher, Wireless Digital Communications: Modulation and Spread Spectrum
Applications. Upper Saddle River, NJ: Prentice Hall, 1995.
ld
Nelson, India: Thomson Learning, 2007.
or
W
TU
JN
ll
A
100
Chapter 6
ld
Transmitter and Receiver
or
Techniques
Introduction
W
Electrical communication transmitter and receiver techniques strive toward obtain-
ing reliable communication at a low cost, with maximum utilization of the channel
resources. The information transmitted by the source is received by the destina-
TU
tion via a physical medium called a channel. This physical medium, which may be
wired or wireless, introduces distortion, noise and interference in the transmitted
information bearing signal. To counteract these effects is one of the requirements
while designing a transmitter and receiver end technique. The other requirements
JN
Modulation
a passband signal. The baseband signal is called the modulating signal and the
passband signal is called the modulated signal. Modulation can be done by varying
A
101
Choice of Modulation Scheme
ld
scheme is often measured in terms of its power efficiency and bandwidth efficiency.
Power efficiency describes the ability of a modulation technique to preserve the
fidelity of the digital message at low power levels. In a digital communication system,
in order to increase noise immunity, it is necessary to increase the signal power.
or
Bandwidth efficiency describes the ability of a modulation scheme to accommodate
data within a limited bandwidth.
The system capacity of a digital mobile communication system is directly related
W
to the bandwidth efficiency of the modulation scheme, since a modulation with a
greater value of ηb(= RB) will transmit more data in a given spectrum allocation.
There is a fundamental upper bound on achievable bandwidth efficiency. Shan-
non’s channel coding theorem states that for an arbitrarily small probability of error,
the maximum possible bandwidth efficiency is limited by the noise in the channel,
TU
and is given by the channel capacity formula
C S
ηBmax = = log2(1 + ) (6.1)
B N
Advantages of Modulation
JN
3. Reduction in antenna size: The antenna height and aperture is inversely pro-
portional to the radiated signal frequency and hence high frequency signal
radiation result in smaller antenna size.
102
Linear and Non-linear Modulation Techniques
The mathematical relation between the message signal (applied at the modulator
input) and the modulated signal (obtained at the modulator output) decides whether
a modulation technique can be classified as linear or non-linear. If this input-output
relation satisfies the principle of homogeneity and superposition then the modulation
ld
technique is said to be linear. The principle of homogeneity states that if the input
signal to a system (in our case the system is a modulator) is scaled by a factor then
the output must be scaled by the same factor. The principle of superposition states
that the output of a linear system due to many simultaneously applied input signals
or
is equal to the summation of outputs obtained when each input is applied one at a
time.
For example an amplitude modulated wave consists of the addition two terms: the
Then,
W
message signal multiplied with the carrier and the carrier itself. If m(t) is the
message signal and sAM (t) is the modulated signal given by:
Here, sAM 1(t) and sAM 2(t) are the outputs obtained when m1(t) and m2(t)
A
103
Amplitude and Angle Modulation
ld
varies a sinusoidal carrier signal in such a way that the angle of the carrier is varied
according to the amplitude of the modulating baseband signal.
or
The nature of the information generating source classifies a modulation technique as
an analog or digital modulation technique. When analog messages generated from
a source passe through a modulator, the resulting amplitude or angle modulation
W
technique is called analog modulation. When digital messages undergo modulation
the resulting modulation technique is called digital modulation.
Any arbitrary signal can be expressed as the linear combination of a set of orthog-
onal signals or equivalently as a point in an M dimensional signal space, where M
denotes the cardinality of the set of orthogonal signals. These orthogonal signals are
JN
normalized with respect to their energy content to yield an orthonormal signal set
having unit energy. These orthonormal signals are independent of each other and
form a basis set of the signal space.
Generally a digitally modulated signal s(t), having a symbol duration T, is ex-
pressed as a linear combination of two orthonormal signals φ1(t) and φ2(t), consti-
tuting the two orthogonal axis in this two dimensional signal space and is expressed
ll
mathematically as,
A
104
.
φ2(t) = 2 cos(2πfct) (6.7)
T
The coefficients s1 and s2 form the coordinates of the signal s(t) in the two dimen-
sional signal space.
ld
nals and Band Pass Systems
A band-pass signal s(t) can be resolved in terms of two sinusoids in phase quadrature
or
as follows:
Hence sI (t) and sQ(t) are known as the in-phase and quadrature-phase components
W
respectively. When sI (t) and sQ(t) are incorporated in the formation of the following
complex signal,
The complex baseband model for the impulse response therefore becomes,
When s(t) passes through h(t), then in the complex baseband domain, the output
r̃(t) of the bandpass system is given by the following convolution
105
Linear Modulation Techniques
ld
represented as:
sAM (t) = Ac[1 + km(t)] cos(2πfct). (6.15)
The modulation index k of an AM signal is defined as the ratio of the peak message
or
signal amplitude to the peak carrier amplitude. For a sinusoidal modulating signal
Am
m(t) = Ac cos(2πfmt), the modulation index is given by
A
k = m. (6.16)
Ac
W
This is a nonlinear technique and can be made linear by multiplying the carrier with
the message signal.The resulting modulation scheme is known as DSBSC modula-
tion. In DSBSC the amplitude of the transmitted signal, s(t), varies linearly with
the modulating digital signal, m(t). Linear modulation techniques are bandwidth
TU
efficient and hence are very attractive for use in wireless communication systems
where there is an increasing demand to accommodate more and more users within
a limited spectrum. The transmitted signal DSBSC signal s(t) can be expressed as:
If m(t) is scaled by a factor of a, then s(t), the output of the modulator, is also
scaled by the same factor as seen from the above equation. Hence the principle of
homogeneity is satisfied. Moreover,
= s1(t) + s2(t)
where A is the carrier amplitude and fc is the carrier frequency. Hence the principle
A
106
ld
Figure 6.1: BPSK signal constellation.
or
of the transmitted carrier frequency and phase at the receiver, whereas non-coherent
detection requires no phase information.
BPSK
W
In binary phase shift keying (BPSK), the phase of a constant amplitude carrier
signal is switched between two values according to the two possible signals m1 and
m2 corresponding to binary 1 and 0, respectively. Normally, the two phases are
TU
separated by 180o. If the sinusoidal carrier has an amplitude A, and energy per bit
1 2
Eo = A Tb then the transmitted BPSK signal is
2 c
.
2Eb
sBP SK (t) = m(t) cos(2πfct + θc ). (6.19)
Tb
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QPSK
The Quadrature Phase Shift Keying (QPSK) is a 4-ary PSK signal. The phase of
A
the carrier in the QPSK takes 1 of 4 equally spaced shifts. Although QPSK can
be viewed as a quaternary modulation, it is easier to see it as two independently
modulated quadrature carriers. With this interpretation, the even (or odd) bits are
107
ld
or
Figure 6.2: QPSK signal constellation.
W
TU
used to modulate the in-phase component of the carrier, while the odd (or even)
bits are used to modulate the quadrature-phase component of the carrier.
The QPSK transmitted signal is defined by:
Offset-QPSK
As in QPSK, as shown in Figure 6.3, the NRZ data is split into two streams of odd
and even bits. Each bit in these streams has a duration of twice the bit duration,
108
ld
or
Figure 6.4: DQPSK constellation diagram.
T b, of the original data stream. These odd (d1(t)) and even bit streams (d2(t)) are
then used to modulate two sinusoidals in phase quadrature,and hence these data
W
streams are also called the in-phase and and quadrature phase components. After
modulation they are added up and transmitted. The constellation diagram of Offset-
QPSK is the same as QPSK. Offset-QPSK differs from QPSK in that the d1(t) and
d2(t) are aligned such that the timing of the pulse streams are offset with respect
TU
to each other by Tb seconds. From the constellation diagram it is observed that a
signal point in any quadrant can take a value in the diagonally opposite quadrant
only when two pulses change their polarities together leading to an abrupt 180 degree
phase shift between adjacent symbol slots. This is prevented in O-QPSK and the
allowed phase transitions are ± 90 degree.
JN
Abrupt phase changes leading to sudden changes in the signal amplitude in the
time domain corresponds to significant out of band high frequency components in
the frequency domain. Thus to reduce these sidelobes spectral shaping is done at
baseband. When high efficiency power amplifiers, whose non-linearity increases as
the efficiency goes high, are used then due to distortion, harmonics are generated
ll
and this leads to what is known as spectral regrowth. Since sudden 180 degree phase
changes cannot occur in OQPSK, this problem is reduced to a certain extent.
A
109
π/4 DQPSK
The data for π/4 DQPSK like QPSK can be thought to be carried in the phase of a
single modulated carrier or on the amplitudes of a pair of quadrature carriers. The
modulated signal during the time slot of kT < t < (k + 1)T given by:
ld
Here, ψk+1 = ψk + ∆ψk and ∆ψk can take values π/4 for 00, 3π/4 for 01, −3π/4
for 11 and −π/4 for 10. This corresponds to eight points in the signal constellation
but at any instant of time only one of the four points are possible: the four points
or
on axis or the four points off axis. The constellation diagram along with possible
transitions are shown in Figure 6.4.
Line Coding
W
Specific waveforms are required to represent a zero and a one uniquely so that a
sequence of bits is coded into electrical pulses. This is known as line coding. There
are various ways to accomplish this and the different forms are summarized below.
TU
4. Return to zero (RZ): 1 goes high for half a period while 0 remains at zero
state.
110
ld
or
W
TU
Pulse Shaping
Let us think about a rectangular pulse as defined in BPSK. Such a pulse is not
desirable for two fundamental reasons:
111
ld
Figure 6.6: Rectangular Pulse
or
(a) the spectrum of a rectangular pulse is infinite in extent. Correspondingly, its
frequency content is also infinite. But a wireless channel is bandlimited, means it
would introduce signal distortion to such type of pulses,
W
(b) a wireless channel has memory due to multipath and therefore it introduces ISI.
In order to mitigate the above two effects, an efficient pulse shaping funtion or
a premodulation filter is used at the Tx side so that QoS can be maintained to the
mobile users during communication. This type of technique is called pulse shaping
TU
technique. Below, we start with the fundamental works of Nyquist on pulse shaping
and subsequently, we would look into another type of pulse shaping technique.
There are a number of well known pulse shaping techniques which are used to simul-
JN
taneously to reduce the inter-symbol effects and the spectral width of a modulated
digital signal. We discuss here about the fundamental works of Nyquist. As pulse
shaping is difficult to directly manipulate the transmitter spectrum at RF frequen-
cies, spectral shaping is usually done through baseband or IF processing.
Let the overall frequency response of a communication system (the transmitter,
ll
112
ld
or
Figure 6.7: Raised Cosine Pulse.
is given by:
sin( πt )
W
hef f (t) = πt Ts
(6.26)
1
If we take a rectangular filter with bandwidth f0 ≥
TU
2Ts and convolve it with any
arbitrary even function Z(f) with zero magnitude outside the passband of the rect-
angular filter then a zero ISI effect would be achieved. Mathematically,
f
Hef f (f ) = rect( ) ∗ Z(f ), (6.27)
f0
sin( πt )
JN
Ts
hef f (t) = πt
z(t), (6.28)
Ts
cos(πρt/Ts)
z(t) = . (6.29)
1 − (∆ρt/2T s)2
with ρ being the roll off factor ∈ [0, 1]. As ρ increases roll off in frequency domain
increases but that in time domain decreases.
ll
Since hef f (t) is non-causal, pulse shaping filters are usually truncated within ±6Ts
about t = 0 for each symbol. Digital communication systems thus often store several
symbols at a time inside the modulator and then clock out a group of symbols by
113
using a look up table that represents discrete time waveforms of stored symbols.
This is the way to realize the pulse shaping filters using real time processors.
Non-Nyquist pulse shaping are also useful, which would be discussed later in this
chapter while discussing GMSK.
ld
Many practical mobile radio communications use nonlinear modulation methods,where
the amplitude of the carrier is constant,regardless of the variations in the modulating
or
signal.The Constant envelope family of modulations has the following advantages :
W
2. Low out-of-band radiation of the order of -60 dB to -70dB can be achieved.
However, even if constant envelope has many advantages it still uses more BW
than linear modulation schemes.
There are a number of ways in which the phase of a carrier signal may be varied
in accordance with the baseband signal; the two most important classes of angle
modulation being frequency modulation and phase modulation.
Frequency modulation (FM) involves changing of the frequency of the carrier
signal according to message signal. As the information in frequency modulation is
in the frequency of modulated signal, it is a nonlinear modulation technique. In this
ll
method, the amplitude of the carrier wave is kept constant (this is why FM is called
constant envelope). FM is thus part of a more general class of modulation known
A
as angle modulation.
Frequency modulated signals have better noise immunity and give better perfor-
mance in fading scenario as compared to amplitude modulation.Unlike AM, in an
114
FM system, the modulation index, and hence bandwidth occupancy, can be varied
to obtain greater signal to noise performance.This ability of an FM system to trade
bandwidth for SNR is perhaps the most important reason for its superiority over
AM. However, AM signals are able to occupy less bandwidth as compared to FM
signals, since the transmission system is linear.
An FM signal is a constant envelope signal, due to the fact that the envelope of
ld
the carrier does not change with changes in the modulating signal. The constant
envelope of the transmitted signal allows efficient Class C power amplifiers to be
used for RF power amplification of FM. In AM, however, it is critical to maintain
or
linearity between the applied message and the amplitude of the transmitted signal,
thus linear Class A or AB amplifiers, which are not as power efficient, must be used.
FM systems require a wider frequency band in the transmitting media (generally
several times as large as that needed for AM) in order to obtain the advantages of
W
reduced noise and capture effect. FM transmitter and receiver equipment is also
more complex than that used by amplitude modulation systems. Although frequency
modulation systems are tolerant to certain types of signal and circuit nonlinearities,
special attention must be given to phase characteristics. Both AM and FM may be
TU
demodulated using inexpensive noncoherent detectors. AM is easily demodulated
using an envelope detector whereas FM is demodulated using a discriminator or
slope detector. In FM the instantaneous frequency of the carrier signal is varied
linearly with the baseband message signal m(t), as shown in following equation:
¸
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where Ac, is the amplitude of the carrier, fc is the carrier frequency, and kf is the
frequency deviation constant (measured in units of Hz/V).
Phase modulation (PM) is a form of angle modulation in which the angle θ(t) of
the carrier signal is varied linearly with the baseband message signal m(t), as shown
ll
in equation below.
sP M (t) = Ac cos(2πfc t + kθ m(t)) (6.31)
A
The frequency modulation index βf , defines the relationship between the message
amplitude and the bandwidth of the transmitted signal, and is given by
kf Am ∆
βf = = (6.32)
W W
115
where Am is the peak value of the modulating signal, ∆f is the peak frequency
deviation of the transmitter and W is the maximum bandwidth of the modulating
signal.
The phase modulation index βp is given by
βp = kθ Am = ∆θ (6.33)
ld
where, ∆θ is the peak phase deviation of the transmitter.
BFSK
or
In Binary Frequency Shift keying (BFSK),the frequency of constant amplitude car-
rier signal is switched between two values according to the two possible message
states (called high and low tones) corresponding to a binary 1 or 0. Depending on
S(t) =
.
W
how the frequency variations are imparted into the transmitted waveform,the FSK
signal will have either a discontinuous phase or continuous phase between bits. In
general, an FSK signal may be represented as
where θ(0) sums the phase up to t = 0. Let us now consider a continuous phase
ll
FSK as
.
S(t) = E /T
(2 b
) cos(2πf ct + θ(t)). (6.39)
A
116
and therefore
. .
2Eb 2Eb cos(2π(f ± h/2T )t + θ(0)).(6.41)
S(t) = cos(2πfct ± πht/T + θ(0)) = c
T T
f1 = fc + h/2T (6.42)
ld
f2 = fc − h/2T (6.43)
for which the expression of FSK conforms to that of CPFSK. On the other hand, fc
or
and h can be expressed in terms of f1 and f2 as
W
Therefore, the unknown factor h can be treated as the difference between f1 and f2,
normalized with respect to bit rate 1/T . It is called the deviation ratio. We know
that θ(t) − θ(0) = ±πht/T , 0 ≤ t ≤ T . If we substitute t = T , we have
= −πh →0 (6.48)
This type of CPFSK is advantageous since by looking only at the phase, the trans-
mitted bit can be predicted. In Figure 6.8, we show a phase tree of such a CPFSK
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the baseband rectangular pulses are replaced by half sinusoidal pulses. Spectral
characteristics of an MSK signal is shown in Figure 6.9 from which it is clear that
A
117
ld
or
Figure 6.8: Phase tree of 1101000 CPFSK sequence.
W
TU
2. an impulse response with relatively low overshoot (to limit FM instant fre-
quency deviation;
3. a phase trellis with ±π/2 for odd T and 0 or π values for even T.
ll
GMSK Scheme
A
118
ld
or
Figure 6.10: GMSK generation scheme.
W
return to zero (NRZ-L) data waveform through a premodulation Gaussian pulse
shaping filter. Baseband Gaussian pulse shaping smoothes the trajectory of the
MSK signals and hence stabilizes instantaneous frequency variations over time. This
has the effect of considerably reducing the sidelobes in the transmitted spectrum.
A GMSK generation scheme with NRZ-L data is shown in Figure 6.10 and a receiver
TU
of the same scheme with some MSI gates is shown in Figure 6.11.
GMSK Generator
H(f ) = exp(−(αf ) 2)
where,
(α)2 = ln 2/2(1/B)2 . (6.50)
ll
The premodulation Gaussian filtering introduces ISI in the transmitted signal, but
it can be shown that the degradation is not that great if the 3dB bandwidth-bit
A
119
ld
or
W
Figure 6.11: A simple GMSK receiver.
on the other hand irreducible error rate of the LPF for ISI increases. Therefore, a
TU
Thus the 3 - dB bandwidth is 67.567 kHz. We use below table fig 6 to find out that
90 % power bandwidth is 0.57 Rb.
A
120
ld
or
Figure 6.12: Spectrum of GMSK scheme.
W
Two Practical Issues of Concern
In FDMA, subscribers are allotted frequency slots called channels in a given band
TU
of the electromagnetic spectrum. The side lobes generated due to the transmission
of a symbol in a particular channel overlaps with the channels placed adjacently.
This is because of the fact that transmission of a time limited pulse leads to spectral
spreading in the frequency domain. During simultaneous use of adjacent channels,
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when there is significant amount of power present in the side lobes, this kind of
interference becomes so severe that the required symbol in a particular frequency
slot is completely lost.
Moreover if two terminals transmit equal power then due to wave propagation
through different distances to the receiver, the received signal levels in the two
frequency slots will differ greatly. In such a case the side lobes of the stronger signal
ll
will severely degrade the transmitted signal in the next frequency slot having low
power level. This is known as the near far problem.
A
121
Power Amplifier Nonlinearity
Power amplifiers may be designed as class A, class B, class AB, class C and class D.
They form an essential section of mobile radio terminals. Due to power constraints
on a transmitting terminal, an efficient power amplifier is required which can convert
most of the input power to RF power. Class A amplifier is a linear amplifier but
ld
it has a power efficiency of only 25 %. As we go for subsequent amplifiers having
greater power efficiency, the nonlinearity of the amplifier increases.
In general, an amplifier has linear input output characteristics over a range
of input signal level, that is, it has a constant gain. However, beyond an input
or
threshold level, the gain of the amplifier starts decreasing. Thus the amplitude of
a signal applied at the input of an amplifier suffers from amplitude distortion and
the resulting waveform obtained at the output of the amplifier is of the form of
The operating point of a practical amplifier is given in terms of either the input
back-off or the output back-off.
TU
. .
Vin,rms
Input back − off = 10 log1 0 (6.53)
Vout,rms
. .
Vout,rms
Output back − off = 10 log1 0 (6.54)
Vout,rms
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For a flat fading channel, the probability of error for coherent BPSK and coherent
BFSK are respectively given as,
. .
1 γ . (6.55)
Pe,BP SK = 1− 1+γ
ll
2 . .
1 γ . (6.56)
Pe,BF SK = 1 − 2 +γ
2
A
(6.57)
122
α2 represents the instantaneous power values of the Rayleigh fading channel and E
denotes the expectation operator.
Similarly, for differential BPSK and non coherent BFSK probability of error
expressions are
1
Pe,DP SK = (6.59)
2(11+ γ)
ld
Pe,N CF SK = . (6.60)
(2 + γ)
For large values of SNR = Eb the error probability given above have the simplified
N0
expression.
or
1
Pe,BP SK = (6.61)
4γ
1
Pe,BF SK = (6.62)
2γ
1
W
Pe,DP SK =
2γ
Pe,N CFSK = .
1
γ
(6.63)
(6.64)
From the above equations we observe that an inverse algebraic relation exists be-
tween the BER and SNR. This implies that if the required BER range is around
TU
10−3 to 10−6, then the SNR range must be around 30dB to 60dB.
Bit error rate (Peb) is the same as symbol error rate (Pes) when a symbol consists
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of a single bit as in BPSK modulation. For an MPSK scheme employing gray coded
modulation, where N bits are mapped to a one of the M symbols, such that 2 N = M ,
Peb is given by Pes
P (6.65)
eb ≈
log2M
And for M-ary orthogonal signalling Peb is given by
M/2
ll
Pes. (6.66)
Peb = 1
M−
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123
avoid interference during parallel transmission, the signals can be separated in fre-
quency and then the resulting technique is called Frequency Division Multiplexing
(FDM). In FDM, the adjacent bands are non overlapping but if overlap is allowed by
transmitting signals that are mutually orthogonal (that is, there is a precise math-
ematical relationship between the frequencies of the transmitted signals) such that
one signal has zero effect on another, then the resulting transmission technique is
ld
known as Orthogonal Frequency Division Multiplexing (OFDM).
OFDM is a technique of transmitting high bit rate data into several parallel
streams of low bit rate data. At any instant, the data transmitted simultaneously
or
in each of these parallel data streams is frequency modulated by carriers (called
subcarriers) which are orthogonal to each other. For high data rate communication
the bandwidth (which is limited) requirement goes on increasing as the data rate
increases or the symbol duration decreases. Thus in OFDM, instead of sending a
W
particular number of symbols, say P, in T seconds serially, the P symbols can be
sent in parallel with symbol duration now increased to T seconds instead of T/P
seconds as was previously.
This offers many advantages in digital data transmission through a wireless time
TU
varying channel. The primary advantage of increasing the symbol duration is that
the channel experiences flat fading instead of frequency selective fading since it is
ensured that in the time domain the symbol duration is greater than the r.m.s.
delay spread of the channel. Viewed in the frequency domain this implies that the
bandwidth of the OFDM signal is less than coherent bandwidth of the channel.
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Although the use of OFDM was initially limited to military applications due to
cost and complexity considerations, with the recent advances in large-scale high-
speed DSP, this is no longer a major problem. This technique is being used, in
digital audio broadcasting (DAB), high definition digital television broadcasting
(HDTV), digital video broadcasting terrestrial TV (DVB-T), WLAN systems based
ll
UWA (underwater acoustic channel) has also been indicated. Moreover related or
combined technology such as CDMA-OFDM, TDMA-OFDM, MIMO-OFDM, Vec-
tor OFDM (V-OFDM), wide-band OFDM (W-OFDM), flash OFDM (F-OFDM),
124
OFDMA, wavelet-OFDM have presented their great advantages in certain applica-
tion areas.
Orthogonality of Signals
Orthogonal signals can be viewed in the same perspective as we view vectors which
are perpendicular/orthogonal to each other. The inner product of two mutually
ld
orthogonal vectors is equal to zero. Similarly the inner product of two orthogonal
signals is also equal to zero.
Let ψ k (t) = ej 2πfk t and ψn (t) = ej 2πfn t be two complex exponential signals whose
or
inner product, over the time duration of Ts, is given by:
¸ (i+1)T s
N= ψ k (t).ψ ∗n (t)dt (6.67)
iT s
W
When this integral is evaluated, it is found that if fk and fn are integer multiples
of 1/Ts then N equals zero. This implies that for two harmonics of an exponential
function having a fundamental frequency of 1/Ts, the inner product becomes zero
.But if fk = fn then N equals T s which is nothing but the energy of the complex
exponential signal in the time duration of T s.
TU
P −1 . .
. t
p(t) = cngn(t)exp j2πn for 0 ≤ t ≤ Ts (6.68)
A
n=0
T s
125
If p(t) is sampled at t = kTs /P , then the resulting waveform, is:
P −1 . .
. kTs /P
p(k) = cn gn (kTs/P )exp j2πn
Ts
n=0
1 .P −1 . .
k
c n exp j2πn for 0 ≤ k ≤ P − 1 (6.69)
= √ s P
T n=0
ld
This is nothing but the IDFT on the symbol block of P symbols. This can be realized
using IFFT but the constraint is that P has to be a power of 2. So at the receiver,
FFT can be done to get back the required block of symbols. This implementation is
or
better than using multiple oscillators for subcarrier generation which is uneconomical
and since digital technology has greatly advanced over the past few decades, IFFTs
and FFTs can be implemented easily. The frequency spectrum, therefore consists
of a set of P partially overlapping sinc pulses during any time slot of duration T s .
W
This is due to the fact that the Fourier Transform of a rectangular pulse is a sinc
function. The receiver can be visualized as consisting of a bank of demodulators,
translating each subcarrier down to DC, then integrating the resulting signal over a
symbol period to recover the raw data.
TU
But the OFDM symbol structure so generated at the transmitter end needs to
be modified. Since inter symbol interference (ISI) is introduced by the transmission
channel due to multipaths and also due to the fact that when the bandwidth of
OFDM signal is truncated, its effect in the time domain is to cause symbol spreading
such that a part of the symbol overlaps with the adjacent symbols. In order to cope
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with ISI as discussed previously the OFDM symbol duration can be increased. But
this might not be feasible from the implementation point of view specifically in terms
of FFT size and Doppler shifts.
A different approach is to keep a guard time interval between two OFDM symbols
in which part of the symbol is copied from the end of the symbol to the front and is
ll
popularly known as the cyclic-prefix. If we denote the guard time interval as Tg and
Ts be the useful symbol duration, then after this cyclical extension the total symbol
duration becomes T = Tg + T s . When the guard interval is longer than the length
A
of the channel impulse response, or the multipath delay, then ISI can be eliminated.
However the disadvantage is the reduction in data rate or throughput and greater
power requirements at the transmitting end. The OFDM transmitter and receiver
126
ld
or
W
Figure 6.13: OFDM Transmitter and Receiver Block Diagram.
In this chapter, a major chunk has been devoted to digital communication systems
which obviously have certain distinction in comparison to their analog counterpart
due to their signal-space representation. The important modulation techniques for
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chapter.
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127
Techniques to Mitigate Fading
ld
Effects
or
Introduction
Apart from the better transmitter and receiver technology, mobile communications
require signal processing techniques that improve the link performance. Equaliza-
W
tion, Diversity and channel coding are channel impairment improvement techniques.
Equalization compensates for Inter Symbol Interference (ISI) created by multipath
within time dispersive channels. An equalizer within a receiver compensates for
the average range of expected channel amplitude and delay characteristics. In other
TU
words, an equalizer is a filter at the mobile receiver whose impulse response is inverse
of the channel impulse response. As such equalizers find their use in frequency selec-
tive fading channels. Diversity is another technique used to compensate fast fading
and is usually implemented using two or more receiving antennas. It is usually em-
ployed to reduce the depths and duration of the fades experienced by a receiver in
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a flat fading channel. Channel coding improves mobile communication link perfor-
mance by adding redundant data bits in the transmitted message.At the baseband
portion of the transmitter, a channel coder maps a digital message sequence in to
another specific code sequence containing greater number of bits than original con-
tained in the message. Channel Coding is used to correct deep fading or spectral
ll
null. We discuss all three of these techniques in this chapter. A general framework
of the fading effects and their mitigation techniques is shown in Figure 7.1.
A
128
ld
or
Figure 7.1: A general framework of fading effects and their mitigation techniques.
Equalization
W
ISI has been identified as one of the major obstacles to high speed data transmission
over mobile radio channels. If the modulation bandwidth exceeds the coherence
bandwidth of the radio channel (i.e., frequency selective fading), modulation pulses
TU
are spread in time, causing ISI. An equalizer at the front end of a receiver compen-
sates for the average range of expected channel amplitude and delay characteristics.
As the mobile fading channels are random and time varying, equalizers must track
the time-varying characteristics of the mobile channel and therefore should be time-
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129
the channel and estimate filter coefficients to compensate for the channel.
Tracking Mode:
• When the training sequence is finished the filter coefficients are near optimal.
ld
• When the data of the users are received, the adaptive algorithms of the equal-
izer tracks the changing channel.
or
A Mathematical Framework
W
x(t) = d(t) ∗ h (t) + nb (t)
where d(t) is the transmitted signal, h(t) is the combined impulse response of the
transmitter,channel and the RF/IF section of the receiver and nb (t) denotes the
baseband noise.
(7.1)
TU
If the impulse response of the equalizer is heq (t), the output of the equalizer is
ŷ (t) = d (t) ∗ h (t) ∗ heq (t) + nb (t) ∗ heq (t) = d (t) ∗ g (t) + nb (t) ∗ heq (t) . (7.2)
However, the desired output of the equalizer is d(t) which is the original source data.
Assuming nb (t)=0, we can write y(t) = d(t), which in turn stems the following
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equation:
The main goal of any equalization process is to satisfy this equation optimally. In
frequency domain it can be written as
ll
Heq (f ) H (f ) = 1 (7.4)
which indicates that an equalizer is actually an inverse filter of the channel. If the
A
channel is frequency selective, the equalizer enhances the frequency components with
small amplitudes and attenuates the strong frequencies in the received frequency
131
spectrum in order to provide a flat, composite received frequency response and
linear phase response. For a time varying channel, the equalizer is designed to track
the channel variations so that the above equation is approximately satisfied.
In a zero forcing equalizer, the equalizer coefficients cn are chosen to force the samples
ld
of the combined channel and equalizer impulse response to zero. When each of the
delay elements provide a time delay equal to the symbol duration T, the frequency
response Heq (f ) of the equalizer is periodic with a period equal to the symbol rate
or
1/T. The combined response of the channel with the equalizer must satisfy Nyquist’s
criterion
uation, so it is rarely used for wireless link except for static channels with high SNR
such as local wired telephone. The usual equalizer model follows a time varying or
adaptive structure which is given next.
The basic structure of an adaptive filter is shown in Figure 7.2. This filter is called
the transversal filter, and in this case has N delay elements, N+1 taps and N+1
tunable complex multipliers, called weights. These weights are updated continuously
by an adaptive algorithm. In the figure the subscript k represents discrete time
index. The adaptive algorithm is controlled by the error signal ek . The error signal
ll
is derived by comparing the output of the equalizer, with some signal dk which is
replica of transmitted signal. The adaptive algorithm uses ek to minimize the cost
A
function and uses the equalizer weights in such a manner that it minimizes the cost
function iteratively. Let us denote the received sequence vector at the receiver and
132
ld
or
W
Figure 7.2: A generic adaptive equalizer.
T
xk = [xk , xk−1, ....., xk−N ] , (7.6)
Now, the output sequence of the equalizer yk is the inner product of xk and wk, i.e.,
yk = (x k , w k ) = xT wk = wT x k . (7.8)
k k
ek = dk − yk = dk − xT w k . (7.9)
k
ll
Assuming dk and xk to be jointly stationary, the Mean Square Error (MSE) is given
as
A
= E[(dk − x Tk wk )2]
= E[d 2 ] + w T E[x k x T ]wk − 2E[dk x T ]wk (7.10)
k k k k
133
where wk is assumed to be an array of optimum values and therefore it has been
taken out of the E() operator. The MSE then can be expressed as
MSE = ξ = σ2 + wT Rwk − 2pT wk (7.11)
k k
where the signal variance σ2 = E[d2 ] and the cross correlation vector p between the
d k
ld
. .
p = E [dk x k ] = E d k xk dk xk−1 dk xk−2 · · · dk xk−N . (7.12)
where
or
xk xk−1 xk xk−2 ··· xk xk−N
x2k
2
. . xk−1xk xk−1 xk−1 xk−2 · · · xk−1 xk−N
R=E xx T
=E x x x x x2 ···x x . (7.13)
k k k−2 k k−2 k−1 k−2 k−2 k−N
xk
.
−N xk
W .
.
xk−N xk−1
Since an adaptive equalizer compensates for an unknown and time varying channel,
it requires a specific algorithm to update the equalizer coefficients and track the
channel variations. Factors which determine algorithm’s performance are:
Rate of convergence: Number of iterations required for an algorithm, in re-
ll
134
Computational complexity: Number of operations required to make one com-
plete iteration of the algorithm.
Numerical properties: Inaccuracies like round-off noise and representation
errors in the computer, which influence the stability of the algorithm.
Three classic equalizer algorithms are primitive for most of today’s wireless stan-
dards. These include the Zero Forcing Algorithm (ZF), the Least Mean Square Algo-
ld
rithm (LMS), and the Recursive Least Square Algorithm (RLS). Below, we discuss
a few of the adaptive algorithms.
or
Least Mean Square (LMS) Algorithm
LMS algorithm is the simplest algorithm based on minimization of the MSE between
the desired equalizer output and the actual equalizer output, as discussed earlier.
Here the system error, the MSE and the optimal Wiener solution remain the same
W
as given the adaptive equalization framework.
In practice, the minimization of the MSE is carried out recursively, and may be
performed by use of the stochastic gradient algorithm. It is the simplest equalization
algorithm and requires only 2N+1 operations per iteration. The filter weights are
TU
updated by the update equation. Letting the variable n denote the sequence of
iteration, LMS is computed iteratively by
where the subscript k denotes the kth delay stage in the equalizer and µ is the step
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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
characteristics that is greater than the propagation delay through the equalizer, then
the equalizer will be unable to reduce distortion. The convergence rate of the LMS
algorithm is slow due to the fact that there is only one parameter, the step size, that
ll
controls the adaptation rate. To prevent the adaptation from becoming unstable,
the value of µ is chosen from
A
,N
.
0<µ <2 λi (7.17)
i=1
where λi is the i-th eigenvalue of the covariance matrix R.
135
Normalized LMS (NLMS) Algorithm
ld
becomes very small and the correction factor may diverge. So, a small positive
number ε is added to the denominator term of the correction factor. Here, the step
size is time varying and is expressed as
or
β
µ (n) = . (7.18)
"x (n)"2 + ε
Therefore, the NLMS algorithm update equation takes the form of
Diversity
wk (n + 1) = wk (n) +
W β
"x (n)" + ε 2
ek (n) x (n − k) . (7.19)
over independent fading paths. It exploits the random nature of radio propagation
by finding independent signal paths for communication. It is a very simple concept
where if one path undergoes a deep fade, another independent path may have a
strong signal. As there is more than one path to select from, both the instantaneous
JN
and average SNRs at the receiver may be improved. Usually diversity decisions are
made by receiver. Unlike equalization, diversity requires no training overhead as a
training sequence is not required by transmitter. Note that if the distance between
two receivers is a multiple of λ/2, there might occur a destructive interference be-
tween the two signals. Hence receivers in diversity technique are used in such a
way that the signal received by one is independent of the other. Diversity can be of
ll
various forms, starting from space diversity to time diversity. We take up the types
one by one in the sequel.
A
136
ld
or
Figure 7.3: Receiver selection diversity, with M receivers.
Space Diversity
W
A method of transmission or reception, or both, in which the effects of fading are
minimized by the simultaneous use of two or more physically separated antennas,
TU
ideally separated by one half or more wavelengths. Signals received from spatially
separated antennas have uncorrelated envelopes.
Space diversity reception methods can be classified into four categories: selection,
feedback or scanning, maximal ratio combining and equal gain combining.
(a) Selection Diversity:
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The basic principle of this type of diversity is selecting the best signal among all
the signals received from different branches at the receiving end. Selection Diversity
is the simplest diversity technique. Figure 7.3 shows a block diagram of this method
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 branches
ll
channel is called a diversity branch and let each branch has the same average SNR.
The signal to noise ratio is defined as
Eb
SNR = Γ = α2 (7.20)
N0
137
where Eb is the average carrier energy, N0 is the noise PSD, α is a random variable
used to represent amplitude values of the fading channel.
The instantaneous SNR(γi) is usually defined as γi = instantaneous signal power
per branch/mean noise power per branch. For Rayleigh fading channels, α has a
Rayleigh distribution and so α2 and consequently γi have a chi-square distribution
with two degrees of freedom. The probability density function for such a channel is
ld
1 −γi
Γ . (7.21)
p (γi) = e
Γ
The probability that any single branch has an instantaneous SNR less than some
or
defined threshold γ is
γ γ
¸ ¸ −γi −γ
1
Pr [γi ≤ γ] = p (γi) dγi = e Γ dγi = 1 − e Γ = P (Γ). (7.22)
0 0 Γ
W
Similarly, the probability that all M independent diversity branches receive signals
which are simultaneously less than some specific SNR threshold γ is
Pr [γ 1, γ 2, . . . , γ M ≤ γ] = 1 − e
. −γ
Γ
.M
= PM (γ) (7.23)
SNR = γ. Quite clearly, PM (Γ) < P (Γ). If a single branch achieves SNR > γ, then
the probability that SNR > γ for one or more branches is given by
. −γ
.M
Pr [γ i > γ] = 1 − P M (γ) = 1 − 1 − e Γ (7.24)
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which is more than the required SNR for a single branch receiver. This expression
shows the advantage when a selection diversity is used.
To determine of average signal to noise ratio, we first find out the pdf of γ as
d M. .M −1 e−γ/Γ .
− γ/Γ (7.25)
pM (γ) = PM (γ) = 1 −e
dγ Γ
The average SNR, γ̄, can be then expressed as
ll
¸∞ ¸∞ M−1
. −x.
γ̄ = γpM (γ) dγ = Γ Mx 1 − e e−x dx (7.26)
A
0 0
where x = γ/Γ and Γ is the average SNR for a single branch, when no diversity is
used.
138
This equation shows an average improvement in the link margin without requir-
ing extra transmitter power or complex circuitry, and it is easy to implement as
it needed a monitoring station and an antenna switch at the receiver. It is not an
optimal diversity technique as it doesn’t use all the possible branches simultaneously.
(b) Feedback or Scanning Diversity:
Scanning all the signals in a fixed sequence until the one with SNR more than a
ld
predetermined threshold is identified. Feedback or scanning diversity is very similar
to selection diversity except that instead of always using the best of N signals, the N
signals are scanned in a fixed sequence until one is found to be above a predetermined
or
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, but
the advantage is that it is very simple to implement(only one receiver is required).
(c) Maximal Ratio Combining:
W
Signals from all of the m branches are weighted according to their individual
signal voltage to noise power ratios and then summed. Individual signals must be
cophased before being summed, which generally requires an individual receiver and
phasing circuit for each antenna element. Produces an output SNR equal to the
TU
sum of all individual SNR. Advantage of producing an output with an acceptable
SNR even when none of the individual signals are themselves acceptable. Modern
DSP techniques and digital receivers are now making this optimal form, as it gives
the best statistical reduction of fading of any known linear diversity combiner. In
terms of voltage signal,
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m
.
rm = ttiri (7.27)
i=1
where tti is the gain and ri is the voltage signal from each branch.
(d) Equal Gain Combining:
In some cases it is not convenient to provide for the variable weighting capability
required for true maximal ratio combining. In such cases, the branch weights are
ll
all set unity, but the signals from each branch are co-phased to provide equal gain
combining diversity. It allows the receiver to exploit signals that are simultaneously
A
139
ld
or
W
TU
unity, here, m
.
rm = ri . (7.28)
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i=1
Polarization Diversity
Polarization Diversity relies on the decorrelation of the two receive ports to achieve
diversity gain. The two receiver ports must remain cross-polarized. Polarization
Diversity at a base station does not require antenna spacing. Polarization diversity
ll
140
munication base stations since it is less susceptible to the near random orientations
of transmitting antennas.
Frequency Diversity
ld
behind this technique is that frequencies separated by more than the coherence
bandwidth of the channel will be uncorrelated and will thus not experience the same
fades. The probability of simultaneous fading will be the product of the individual
or
fading probabilities. This method is employed in microwave LoS links which carry
several channels in a frequency division multiplex mode (FDM). Main disadvantage
is that it requires spare bandwidth also as many receivers as there are channels used
for the frequency diversity.
Time Diversity
W
In time diversity, the signal representing the same information are sent over the
same channel at different times. Time diversity repeatedly transmits information at
TU
time spacings that exceeds the coherence time of the channel. Multiple repetition
of the signal will be received with independent fading conditions, thereby providing
for diversity. A modern implementation of time diversity involves the use of RAKE
receiver for spread spectrum CDMA, where the multipath channel provides redun-
dancy in the transmitted message. Disadvantage is that it requires spare bandwidth
JN
also as many receivers as there are channels used for the frequency diversity. Two
important types of time diversity application is discussed below.
In CDMA spread spectrum systems, CDMA spreading codes are designed to provide
ll
very low correlation between successive chips, propagation delay spread in the radio
channel provides multiple version of the transmitted signal at the receiver. Delaying
A
multipath components by more than a chip duration, will appear like uncorrelated
noise at a CDMA receiver. CDMA receiver may combine the time delayed versions
of the original signal to improve the signal to noise ratio at the receiver. RAKE
141
ld
or
Figure 7.5: RAKE receiver.
W
receiver collect the time shifted versions of the original signal by providing a sep-
arate correlation receiver for M strongest multipath components. Outputs of each
correlator are weighted to provide a better estimate of the transmitted signal than
provided by a single component. Demodulation and bit decisions are based on the
TU
Application 2: Interleaver
JN
In the encoded data bits, some source bits are more important than others, and
must be protected from errors. Many speech coder produce several important bits
in succession. Interleaver spread these bit out in time so that if there is a deep fade
or noise burst, the important bits from a block of source data are not corrupted
at the same time. Spreading source bits over time, it becomes possible to make
use of error control coding. Interleaver can be of two forms, a block structure or a
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convolutional structure.
A block interleaver formats the encoded data into a rectangular array of m rows
A
and n columns, and interleaves nm bits at a time. Each row contains a word of
source data having n bits. an interleaver of degree m consists of m rows. source bits
are placed into the interleaver by sequentially increasing the row number for each
142
successive bit, and forming the columns. The interleaved source data is then read
out row-wise and transmitted over the channel. This has the effect of separating
the original source bits by m bit periods. At the receiver, de-interleaver stores the
received data by sequentially increasing the row number of each successive bit, and
then clocks out the data row-wise, one word at a time. Convolutional interleavers
are ideally suited for use with convolutional codes.
ld
Channel Coding
or
In channel coding, redundant data bits are added in the transmitted message so
that if an instantaneous fade occurs in the channel, the data may still be recov-
ered at the receiver without the request of retransmission. A channel coder maps
the transmitted message into another specific code sequence containing more bits.
W
Coded message is then modulated for transmission in the wireless channel. Channel
Coding is used by the receiver to detect or correct errors introduced by the channel.
Codes that used to detect errors, are error detection codes. Error correction codes
can detect and correct errors.
TU
In 1948, Shannon showed that by proper encoding of the information, errors induced
by a noise channel can be reduced to any desired level without sacrificing the rate
of information transfer. Shannon’s channel capacity formula is applicable to the
JN
C = B log2 . .
S = B log2
.
P
.
= B log2 . Eb Rb
. (7.29)
1+ N 1 + N0 B 1 + N0 B
where C is the channel capacity (bit/s), B is the channel bandwidth (Hz), P is the
received signal power (W), N0 is the single sided noise power density (W/Hz), Eb is
the average bit energy and Rb is transmission bit rate.
ll
2 N 0 B
B
and the ratio C/B is denoted as bandwidth efficiency. Introduction of redundant
bits increases the transmission bit rate and hence it increases the bandwidth require-
ment, which reduces the bandwidth efficiency of the link in high SNR conditions, but
143
provides excellent BER performance at low SNR values. This leads to the following
two inferences.
Corollary 1 : While dealing within maximum channel capacity, introduction of re-
dundant bits increase the transmitter rate and hence bandwidth requirement also
increases, while decreasing the bandwidth efficiency, but it also decreases the BER.
Corollary 2 : If data redundancy is not introduced in a wideband noisy environment,
ld
error free performance in not possible (for example, CDMA communication in 3G
mobile phones).
A channel coder operates on digital message (or source) data by encoding the source
or
information into a code sequence for transmission through the channel. The error
correction and detection codes are classified into three groups based on their struc-
ture.
1. Block Code
2. Convolution Code
3. Concatenated Code.
Block Codes
W
TU
Block codes are forward error correction (FEC) codes that enable a limited number
of errors to be detected and corrected without retransmission. Block codes can be
used to improve the performance of a communications system when other means of
improvement (such as increasing transmitter power or using a more sophisticated
demodulator) are impractical.
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In block codes, parity bits are added to blocks of message bits to make codewords
or code blocks. In a block encoder, k information bits are encoded into n code bits.
A total of n−k redundant bits are added to the k information bits for the purpose of
detecting and correcting errors. The block code is referred to as an (n, k) code, and
the rate of the code is defined as Rc = k/n and is equal to the rate of information
ll
144
ments in which two codewords Ci and Cj differs denoted by d (Ci, Cj ). If the code
used is binary, the distance is known as ’Hamming distance’. For example d(10110,
11011) is 3. If the code ’C’ consists of the set of codewords, then the minimum
distance of the code is given by dmin = min{d (Ci, Cj )}.
(c) Code Weight (w): Weight of a codeword is given by the number of nonzero
elements in the codeword. For a binary code, the weight is basically the number of
ld
1s in the codeword. For example weight of a code 101101 is 4.
Ex 1: The block code C = 00000, 10100, 11110, 11001 can be used to represent two
bit binary numbers as:
or
• 00 – 00000
• 01 – 10100
• 10 – 11110
• 11 – 11001
W
Here number of codewords is 4, k = 2, and n = 5.
To encode a bit stream 1001010011
TU
(a) Linearity: Suppose Ci and Cj are two code words in an (n, k) block code. Let
α1 and α2 be any two elements selected from the alphabet. Then the code is said to
be linear if and only if α1C1 + α2C2 is also a code word. A linear code must contain
the all-zero code word.
ll
(b) Systematic: A systematic code is one in which the parity bits are appended
to the end of the information bits. For an (n, k) code, the first k bits are identical
A
to the information bits, and the remaining n − k bits of each code word are linear
combinations of the k information bits.
145
(c) Cyclic: Cyclic codes are a subset of the class of linear codes which satisfy the
following cyclic shift property: If C = [Cn−1, Cn−2, ..., C0] is a code word of a cyclic
code, then [Cn−2, Cn−3, ..., C0, Cn−1], obtained by a cyclic shift of the elements of C,
is also a code word. That is, all cyclic shifts of C are code words.
In this context, it is important to know about Finite Field or Galois Field.
Let F be a finite set of elements on which two binary operations – addition (+) and
ld
multiplication (.) are defined. The set F together with the two binary operations is
called a field if the following conditions are satisfied:
1. F is a commutative group under addition.
or
2. The set of nonzero elements in F is a commutative group under multiplication.
3. Multiplication is distributive over addition; that is, for any three elements a, b,
and c in F, a(b + c) = ab + ac
4. Identity elements 0 and 1 must exist in F satisfying a + 0 = a and a.1 = a.
W
5. For any a in F, there exists an additive inverse (−a) such that a + (−a) = 0.
6. For any a in F, there exists an multiplicative inverse a−1 such that a.a−1 = 1.
Depending upon the number of elements in it, a field is called either a finite or an
infinite field. The examples of infinite field include Q (set of all rational numbers),
TU
R (set of all real numbers), C (set of all complex numbers) etc. A field with a finite
number of elements (say q) is called a ’Galois Field’ and is denoted by GF(q). A
finite field entity p(x), called a polynomial, is introduced to map all symbols (with
several bits) to the element of the finite field. A polynomial is a mathematical
expression
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where the symbol x is called the indeterminate and the coefficients p 0, p 1, ..., pm are
the elements of GF(q). The coefficient pm is called the leading coefficient. If pm
is not equal to zero, then m is called the degree of the polynomial, denoted as deg
p(x). A polynomial is called monic if its leading coefficient is unity. The division
ll
algorithm states that for every pair of polynomials a(x) and b(x) in F(x), there
exists a unique pair of polynomials q(x), the quotient, and r(x), the remainder,
A
such that a(x) = q(x)b(x) + r(x), where deg r(x)¡deg b(x). A polynomial p(x) in
F(x) is said to be reducible if p(x)=a(x)b(x), otherwise it is called irreducible. A
monic irreducible polynomial of degree at least one is called a prime polynomial.
146
An irreducible polynomial p(x) of degree ‘m’ is said to be primitive if the smallest
integer ‘n’ for which p(x) divides xn +1 is n = 2 m −1. A typical primitive polynomial
is given by p(x) = xm + x + 1.
A specific type of code which obeys both the cyclic property as well as poly-
nomial operation is cyclic codes. Cyclic codes are a subset of the class of linear
codes which satisfy the cyclic property. These codes possess a considerable amount
ld
of structure which can be exploited. A cyclic code can be generated by using a
generator polynomial g(p) of degree (n-k). The generator polynomial of an (n,k)
cyclic code is a factor of pn + 1 and has the form
or
g (p) = pn−k + gn −k −1pn−k−1 + · · · + g1 p + 1. (7.32)
W
x (p) = xk−1pk−1 + · · · + x 1 p + x0
where (xk−1, . . . , x0) represents the k information bits. The resultant codeword c(p)
can be written as
c (p) = x (p) g (p)
(7.33)
(7.34)
TU
(a) Single Parity Check Code: In single parity check codes (example: ASCII code),
an overall single parity check bit is appended to ’k’ information bits. Let the infor-
mation bit word be: (b1, b2, ..., bk), then parity check bit: p = b1 + b2 + ......... + bk
modulo 2 is appended at the (k+1)th position, making the overall codeword: C =
(b1, b2, ..., bk , p). The parity bit may follow an even parity or an odd parity pattern.
All error patterns that change an odd number of bits are detectable, and all even
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numbered error patterns are not detectable. However, such codes can only detect
the error, it cannot correct the error.
A
147
1, 1, 1, 1, 0, 0, 1), then it can be easily checked by the receiver that now there are
odd number of 1’s in the codeword and hence there is an error. On the other hand,
if there are two errors, say, errors in bit 3 and 5: (0, 1, 1, 1, 0, 0, 0, 1), then error
will not be detected.
After decoding a received codeword, let pc be the probability that the decoder
gives correct codeword C, pe is the probability that the decoder gives incorrect
ld
codeword Ct ƒ= C, and pf is the probability that the decoder fails to give a codeword.
In this case, we can write pc + pe + pf = 1.
If in an n-bit codeword, there are j errors and p is the bit error probability,
then the probability of obtaining j errors in this codeword is Pj = n C j pj (1 − p)n−j .
or
Using this formula, for any (n, n − 1) single parity check block code, we get
• pc = P0,
W
• pf = P1 + P3 + ... + P tn (nt = n − 1 if n is even, otherwise nt = n).
As an example, for a (5,4) single parity check block code, pc = P0, pe = P2 + P4,
and pf = P1 + P3 + P5.
TU
(b) Product Codes: Product codes are a class of linear block codes which pro-
vide error detection capability using product of two block codes. Consider that nine
information bits (1, 0, 1, 0, 0, 1, 1, 1, 0) are to be transmitted. These 9 bits can be
divided into groups of three information bits and (4,3) single parity check codeword
can be formed with even parity. After forming three codewords, those can be ap-
JN
pended with a vertical parity bit which will form the fourth codeword. Thus the
following codewords are transmitted:
C1 = [1 0 1 0]
C2 = [0 0 1 1]
C3 = [1 1 0 0]
ll
C4 = [0 1 0 1].
A
Now if an error occurs in the second bit of the second codeword, the received code-
words at the receiver would then be
C1 = [1 0 1 0]
148
C2 = [0 1 1 1] ←
C3 = [1 1 0 0]
C4 = [0 1 0 1]
and these would indicate the corresponding row and column position of the erroneous
ld
bit with vertical and horizontal parity check. Thus the bit can be corrected. Here
we get a horizontal (4, 3) codeword and a vertical (4, 3) codeword and concatenating
them we get a (16, 9) product code. In general, a product code can be formed as
or
(n1, k1) & (n2, k2) → (n1n2, k1k2).
(c) Repetition Codes: In a (n,1) repetition code each information bit is repeated
n times (n should be odd) and transmitted. At the receiver, the majority decoding
principle is used to obtain the information bit. Accordingly, if in a group of n received
W
bit, 1 occurs a higher number of times than 0, the information bit is decoded as 1.
Such majority scheme works properly only if the noise affects less than n/2 number
of bits. Ex 3: Consider a (3,1) binary repetition code.
• For input bit 0, the codeword is (0 0 0) and for input bit 1, the codeword is
TU
(1 1 1).
(n, k) = (2m − 1, 2m − 1 − m)
ll
(7.35)
where k is the number of information bits used to form a n bit codeword, and m
A
149
codeword is represented by C = [i1, ...in, p1, ..., pn−k]. This is quite a useful code in
communication which is illustrated via the following example.
Ex 4: Consider a (7, 4) Hamming code. With three parity bits we can correct exactly
1 error. The parity bits may follow such a modulo 2 arithmetic:
ld
p 1 = i1 + i2 + i 3 ,
p 2 = i2 + i3 + i 4 ,
p3 = i1 + i3 + i 4 ,
or
which is same as,
p 1 + i1 + i2 + i3 = 0
p 2 + i2 + i3 + i4 = 0
W
p3 + i1 + i3 + i4 = 0.
The transmitted codeword is then C = [i1, i2, ..., i 4, p1, p2, p3].
Syndrome Decoding: For this Hamming code, let the received codeword be V =
[v1, v2, ..., v4, v5, v6, v7]. We define a syndrome vector S as
TU
S = [S1 S2 S3]
S1 = v1 + v2 + v3 + v5
S2 = v2 + v3 + v4 + v6
S3 = v1 + v2 + v4 + v7
JN
It is obvious that in case of no error, the syndrome vector is equal to zero. Corre-
sponding to this syndrome vector, there is an error vector e which can be obtained
from a syndrome table and finally the required codeword is taken as C = V + e. In
a nutshell, to obtain the required codeword, we perform the following steps:
ll
150
2. Let C = [1 1 0 0 0 1 0] and V = [1 1 0 1 0 1 0]. This means S = [0 1 1], from which
we get e = [0 0 0 1 0 0 0] which means a single bit error is there in the received bit
ld
corrected. Therefore a (7,4) Hamming code can correct only single bit error.
(e) Golay Codes: Golay codes are linear binary (23,12) codes with a minimum
distance of seven and a error correction capability of three bits. This is a special,
or
one of a kind code in that this is the only nontrivial example of a perfect code.
Every codeword lies within distance three of any codeword, thus making maximum
likelihood decoding possible.
(f) BCH Codes: BCH code is one of the most powerful known class of linear
W
cyclic block codes, known for their multiple error correcting ability, and the ease
of encoding and decoding. It’s block length is n = 2m − 1 for m ≥ 3 and number
of errors that they can correct is bounded by t < (2m − 1)/2. Binary BCH codes
can be generalized to create classes of non binary codes which use m bits per code
TU
symbol.
(g) Reed Solomon (RS) Codes: Reed-Solomon code is an important subset of
the BCH codes with a wide range of applications in digital communication and data
storage. Typical application areas are storage devices (CD, DVD etc.), wireless
communications, digital TV, high speed modems. It’s coding system is based on
JN
groups of bits, such as bytes, rather than individual 0 and 1. This feature makes it
particularly good at dealing with burst of errors: six consecutive bit errors. Block
length of these codes is n = 2m − 1, and can be extended to 2m or 2 m + 1. Number
of parity symbols that must be used to correct e errors is n − k = 2e. Minimum
distance dmin = 2e + 1, and it achieves the largest possible d min of any linear code.
ll
For US-CDPD, the RS code is used with m = 6. So each of the 64 field elements
is represented by a 6 bit symbol. For this case, we get the primitive polynomial as
A
151
Table 7.1: Finite field elements for US-CDPD
α5 α4 α3 α2 α1 α0
1 0 0 0 0 0 1
α1 0 0 0 1 0 0
α2 0 0 1 0 0 0
ld
. . . . . . .
. . . . . . .
α6 = α + 1 0 0 0 0 1 1
. . . . . . .
or
. . . . . . .
W
The encoding part of the RS polynomial is done as follows:
Information polynomial: d(x) = Cn−1xn−1 + Cn−2x n−2 + ..... + C2tx 2t ,
Parity polynomial: p(x) = C2t−1x2t−1 + ... + C 0,
Codeword polynomial: c(x) = d(x) + p(x).
Since generating an information polynomial is difficult, so a generating polynomial is
TU
used instead. Information polynomial is then the multiple of generating polynomial.
This process is given below.
Since this kind of codes are cyclic codes, we take a generating polynomial g(x)
such that d(x) = g(x)q(x) + r(x) where q(x) is the quotient polynomial and r(x)
is the remainder polynomial. The codeword polynomial would then be given as:
JN
c(x) = g(x)q(x) + r(x) = p(x). If we assign a parity polynomial p(x) = r(x), then
the codeword polynomial c(x) = g(x)p(x) and the entire process becomes easier.
On the decoder side one has to find a specific r(x) = p(x) or vice-versa, but due
to its complexity, it is mainly done using syndrome calculation. The details of such
a syndrome calculation can be found in [1].
ll
Convolutional Codes
A
152
ld
Figure 7.6: A convolutional encoder with n=2 and k=1.
or
and m linear algebraic function generators based on the generator polynomials.
Input data is shifted into and along the shift register, k-bits at a time. Number
of output bits for each k-bit user input data sequence is n bits, so the code rate
Rc = k/n. The shift register of the encoder is initialized to all-zero-state before
W
TU
JN
State Diagram:
Since the output of the encoder is determined by the input and the current
state of the encoder, a state diagram can be used to represent the encoding process.
A
The state diagram is simply a graph of the possible states of the encoder and the
possible transitions from one state to another. The path information between the
states, denoted as b/c1c2, represents input information bit ’b’ and the corresponding
153
ld
or
Figure 7.8: Tree diagram representation of a convolutional encoder.
W
output bits (c1c2). Again, it is not difficult to verify from the state diagram that an
input information sequence b = (1011) generates an encoded sequence c = (11, 10,
00, 01).
Tree Diagram:
TU
The tree diagram shows the structure of the encoder in the form of a tree with the
branches representing the various states and the outputs of the coder. The encoded
bits are labeled on the branches of the tree. Given an input sequence, the encoded
sequence can be directly read from the tree. As an example, an input sequence
(1011) results in the encoded sequence (11, 10, 00, 01).
JN
ll
A
154
ld
Figure 7.10: Block diagram of a turbo encoder.
or
Trellis Diagram:
Tree reveals that the structure repeats itself once the number of stages is greater
than the constraint length. It is observed that all branches emanating from two
W
nodes having the same state are identical in the sense that they generate identical
output sequences. This means that the two nodes having the same label can be
merged. By doing this throughout the tree diagram, we obtain another diagram
called a Trellis Diagram which is more compact representation.
TU
Concatenated Codes
Turbo Codes: A turbo encoder is built using two identical convolutional codes
of special type with parallel concatenation. An individual encoder is termed a com-
ponent encoder. An interleaver separates the two component encoders. The inter-
leaver is a device that permutes the data sequence in some predetermined manner.
Only one of the systematic outputs from the two component encoders is used to form
ll
a codeword, as the systematic output from the other component encoder is only a
permuted version of the chosen systematic output. Figure 7.10 shows the block di-
A
agram of a turbo encoder using two identical encoders. The first encoder outputs
the systematic V0 and recursive convolutional V1 sequences while the second en-
coder discards its systematic sequence and only outputs the recursive convolutional
155
may be binary or m-binary encoder. Encoders are also categorized as systematic
or non-systematic. If the component encoders are not identical then it is called an
asymmetric turbo code.
ld
or
W
TU
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ll
A
156
ld
Equalization, Diversity, and Channel Coding
or
•Introduction
W
•Equalization Techniques
•Algorithms for Adaptive Equalization
•Diversity Techniques
TU
•RAKE Receiver
•Channel Coding
JN
ll
A
NCCU
7-1 Wireless Comm. Lab.
ld
Introduction[1]
or
•Three techniques are used independently or in tandem to improve
W
receiver signal quality
•Equalization compensates for ISI created by multipath with time
dispersive channels (W>BC)
TU
¾Linear equalization, nonlinear equalization
•Diversity also compensates for fading channel impairments, and is
usually implemented by using two or more receiving antennas
JN
¾Spatial diversity, antenna polarization diversity, frequency
diversity, time diversity
ll
A
NCCU
7-2 Wireless Comm. Lab.
ld
Introduction[1]
or
•The former counters the effects of time dispersion (ISI), while the
latter reduces the depth and duration of the fades experienced
W
by a receiver in a flat fading (narrowband) channel
• Channel Coding improves mobile communication link
performance by adding redundant data bits in the transmitted
TU
message
•Channel coding is used by the Rx to detect or correct some (or all)
of the errors introduced by the channel (Post detection
JN
technique)
¾Block code and convolutional code
ll
A
NCCU
7-3 Wireless Comm. Lab.
ld
Equalization Techniques
or
• The term equalization can be used to describe any signal
W
processing operation that minimizes ISI [2]
• Two operation modes for an adaptive equalizer: training
and tracking
TU
•Three factors affect the time spanning over which an
equalizer converges: equalizer algorithm, equalizer
structure and time rate of change of the multipath radio
JN
channel
•TDMA wireless systems are particularly well suited for
equalizers
ll
A
NCCU
7-4 Wireless Comm. Lab.
ld
Equalization Techniques
or
• Equalizer is usually implemented at baseband or at IF in a
receiver (see Fig. 1)
W
y( t ) = x( t ) ∗ f ∗( t ) + n ( t )
b
TU
f*(t): complex conjugate of f(t)
nb(t): baseband noise at the input of the equalizer
heq(t): impulse response of the equalizer
JN
ll
A
NCCU
7-5 Wireless Comm. Lab.
ld
Equalization Techniques
or
W
TU
JN
Fig. 1
ll
A
NCCU
7-6 Wireless Comm. Lab.
ld
Equalization Technologies
or
dˆ (t ) = y (t ) ∗ heq (t )
= x (t ) ∗ f ∗ (t ) ∗ heq (t ) + mb (t ) ∗ heq (t )
W
= δ(t )
∴ F ∗ (− f ) ∗ H eq ( f ) = 1
TU
• If the channel is frequency selective, the equalizer enhances the
frequency components with small amplitudes and attenuates the strong
JN
frequencies in the received frequency response
• For a time-varying channel, an adaptive equalizer is needed to track the
channel variations
ll
A
NCCU
7-7 Wireless Comm. Lab.
ld
Basic Structure of Adaptive Equalizer
or
•Transversal filter with N delay elements, N+1 taps, and N+1 tunable
complex weights
W
TU
JN
NCCU
7-8 Wireless Comm. Lab.
ld
Equalization Techniques
or
•Classical equalization theory : using training sequence to minimize
the cost function
W
E[e(k) e*(k)]
•Recent techniques for adaptive algorithm : blind algorithms
¾Constant Modulus Algorithm (CMA, used for constant envelope
TU
modulation) [3]
¾Spectral Coherence Restoral Algorithm (SCORE, exploits spectral
JN
redundancy or cyclostationarity in the Tx signal) [4]
ll
A
NCCU
7-9 Wireless Comm. Lab.
ld
Solutions for Optimum Weights of Figure 2 (一)
or
•Error signal e k = x k − y kT ω k = x k − ω kT y k
where y k = [ y k y k −1 y k − 2 .... y k − N ]T
W
ω k = [ωk ωk − 1 ωk − 2 .... ωk − N ]T
= x k2 + ω y k y kT ω − 2 x k y kT ω
2
•Mean square error ek T
k k k
TU
•Expected MSE ξ = E ek [ ] = E [x ]+ ω
2 2
k
T
Rω − 2pTω
where ⎡ yk2 yk yk −1 yk yk − 2 .... yk yk − N ⎤
⎢ ⎥
[ ]y y
R = E yk yk = E ⎢ k −1 k
yk2−1 yk −1 yk −2 .... yk −1 yk − N ⎥
JN
*
⎢ .... .... .... .... .... ⎥
⎢ 2 ⎥
⎣ yk − N yk yk − N yk −1 yk − N yk − 2 .... yk − N ⎦
p = E[ x k y k ] = E[ x k y k xk yk −2 .... xk yk − N ]
T
xk yk −1
ll
A
NCCU
7-10 Wireless Comm. Lab.
ld
Solutions for Optimum Weights of Figure 2 (二)
or
•Optimum weight vector
ˆ = R −1p
ω
W
•Minimum mean square error (MMSE)
[ ] T −1
ξmin = E χ κ2 − p R p
= E[χ ] −
TU
2
κ
pΤω̂
•Minimizing the MSE tends to reduce the bit error rate
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ll
A
NCCU
7-11 Wireless Comm. Lab.
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Equalization Techniques
or
•Two general categories - linear and nonlinear
equalization (see Fig. 3)
W
•In Fig. 1, if d(t) is not the feedback path to adapt the equalizer,
the equalization is linear
•In Fig. 1, if d(t) is fed back to change the subsequent outputs
TU
of the equalizer, the equalization is nonlinear
JN
ll
A
NCCU
7-12 Wireless Comm. Lab.
ld
Equalization Techniques
or
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TU
JN
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NCCU
7-13 Wireless Comm. Lab.
ld
Equalizer Techniques
or
•Linear transversal equalizer (LTE, made up of tapped delay lines
as shown in Fig.4)
W
TU
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Fig.4 Basic linear transversal equalizer structure
NCCU
7-14 Wireless Comm. Lab.
ld
Equalizer Techniques
or
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TU
JN
Fig.5 Tapped delay line filter with both feedforward and feedback taps
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A
NCCU
7-15 Wireless Comm. Lab.
ld
Structure of a Linear Transversal Equalizer [5]
or
W
TU
N2
• dˆ k = ∑ C n* y k − n
n=− N1
[ ]
JN
T πT No
• E e(n) = ∫ dω
2
π
2π T F( e ) + N o
− jωt 2
NCCU
7-16 Wireless Comm. Lab.
ld
Structure of a Lattice Equalizer [6-7]
or
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TU
JN
NCCU
7-17 Wireless Comm. Lab.
ld
Characteristics of Lattice Filter
or
•Advantages
¾Numerical stability
W
¾Faster convergence
¾Unique structure allows the dynamic assignment of the most effective
length
TU
•Disadvantages
¾The structure is more complicated
JN
ll
A
NCCU
7-18 Wireless Comm. Lab.
ld
Nonlinear Equalization
or
•Used in applications where the channel distrotion is too severe
•Three effective methods [6]
W
¾Decision Feedback Equalization (DFE)
¾Maximum Likelihood Symbol Detection
¾Maximum Likelihood Sequence Estimator (MLSE)
TU
JN
ll
A
NCCU
7-19 Wireless Comm. Lab.
ld
Nonlinear Equalization--DFE
or
•Basic idea : once an information symbol has been detected and decided
upon, the ISI that it induces on future symbols can be estimated and
W
substracted out before detection of subsequent symbols
•Can be realized in either the direct transversal form (see Fig.8) or as a
lattice filter
TU
N2 N3
• dˆ k = ∑ C y k − n + ∑ F i d k − i
*
n
n=− N1 i=1
JN
[
• E e(n)
2
] T Tπ
= exp{ ∫− π ln[
2π T
No
]dω}
F( e ) + N o
jωT 2
min
ll
A
NCCU
7-20 Wireless Comm. Lab.
ld
Nonlinear Equalizer-DFE
or
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TU
JN
NCCU
7-21 Wireless Comm. Lab.
ld
Nonlinear Equalization--DFE
or
•Predictive DFE (proposed by Belfiore and Park, [8])
•Consists of an FFF and an FBF, the latter is called a noise predictor
W
( see Fig.9 )
•Predictive DFE performs as well as conventional DFE as the limit
in the number of taps in FFF and the FBF approach infinity
TU
•The FBF in predictive DFE can also be realized as a lattice structure [9].
The RLS algorithm can be used to yield fast convergence
JN
ll
A
NCCU
7-22 Wireless Comm. Lab.
ld
Nonlinear Equalizer-DFE
or
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TU
JN
NCCU
7-23 Wireless Comm. Lab.
ld
Nonlinear Equalization--MLSE
or
•MLSE tests all possible data sequences (rather than decoding each
received symbol by itself ), and chooses the data sequence with the
W
maximum probability as the output
•Usually has a large computational requirement
•First proposed by Forney [10] using a basic MLSE estimator
TU
structure and implementing it with the Viterbi algorithm
•The block diagram of MLSE receiver (see Fig.10 )
JN
ll
A
NCCU
7-24 Wireless Comm. Lab.
ld
Nonlinear Equalizer-MLSE
or
W
TU
Fig.10 The structure of a maximum likelihood sequence equalizer(MLSE) with
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an adaptive matched filter
•MLSE requires knowledge of the channel characteristics in order
to compute the matrics for making decisions
•MLSE also requires knowledge of the statistical distribution of
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NCCU
7-25 Wireless Comm. Lab.
ld
Algorithm for Adaptive Equalization
or
•Excellent references [6, 11--12]
•Performance measures for an algorithm
W
¾Rate of convergence
¾Misadjustment
¾Computational complexity
TU
¾Numerical properties
•Factors dominate the choice of an equalization structure and its algorithm
¾The cost of computing platform
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¾The power budget
¾The radio propagation characteristics
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A
NCCU
7-26 Wireless Comm. Lab.
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Algorithm for Adaptive Equalization
or
•The speed of the mobile unit determines the channel fading rate and the
Dopper spread, which is related to the coherent time of the channel
W
directly
•The choice of algorithm, and its corresponding rate of convergence,
depends on the channel data rate and coherent time
TU
•The number of taps used in the equalizer design depends on the maximum
expected time delay spread of the channel
•The circuit complexity and processing time increases with the number of
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taps and delay elements
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A
NCCU
7-27 Wireless Comm. Lab.
ld
Algorithm for Adaptive Equalization
or
•Three classic equalizer algorithms : zero forcing (ZF), least mean squares
(LMS), and recursive least squares (RLS) algorithms
W
•Summary of algorithms (see Table 1)
TU
JN
ll
A
NCCU
7-28 Wireless Comm. Lab.
ld
Summary of algorithms
or
W
TU
JN
NCCU
7-29 Wireless Comm. Lab.
ld
Diversity Techniques
or
•Requires no training overhead
•Can provides significant link improvement with little added cost
W
•Diversity decisions are made by the Rx, and are unknown to the Tx
•Diversity concept
¾If one radio path undergoes a deep fade, another independent path may
TU
have a strong signal
¾By having more than one path to select from, both the instantaneous
and average SNRs at the receiver may be improved, often by as much
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as 20 dB to 30 dB
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A
NCCU
7-30 Wireless Comm. Lab.
ld
Diversity Techniques
or
•Microscopic diversity and Macroscopic diversity
¾The former is used for small-scale fading while the latter for large-scale
W
fading
¾Antenna diversity (or space diversity)
•Performance for M branch selection diversity (see Fig.11)
TU
Pr[SNR > r ] = 1 − Pr [γ 1 , .... , γ M ≤ r ]
= 1 − (1 − e − r/Γ )M
Μ
JN
PM (r) = Pr[SNR≤ r] = (1− e−r/Γ )M−1e−r/Γ
d
dr Γ
M
r 1
Γ
= ∑
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k =1 k
A
NCCU
7-31 Wireless Comm. Lab.
ld
Diversity techniques
or
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TU
JN
selection diversity. The term Γ represents the mean SNR on each branch
A
NCCU
7-32 Wireless Comm. Lab.
ld
Diversity Techniques
or
• Performance for Maximal Ratio Combining Diversity [13]
(see Fig. 12)
W
M
γ M = ∑ Giγ i
M
N T = N ∑ Gi
2
i =1 i =1
TU
γ M2
rM =
2 NT
( r / Γ ) k −1
JN
r M
Pr{rM ≤ r} = ∫ p ( rM )drM = 1 − e −r / Γ
∑
0 k =1 ( k − 1)!
M −1 − rM / Γ
rM
e
P ( rM ) = M
Γ ( M − 1)!
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A
NCCU
7-33 Wireless Comm. Lab.
ld
Diversity Techniques
or
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TU
JN
Fig. 12 Generalized block diagram for space diversity
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A
NCCU
7-34 Wireless Comm. Lab.
ld
Diversity Techniques
or
• Space diversity [14]
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¾ Selection diversity
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¾ Feedback diversity
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¾ Maximal ration combining
NCCU
7-35 Wireless Comm. Lab.
ld
Diversity Techniques
or
• Selection diversity (see Fig. 13)
¾ The receiver branch having the highest instantaneous SNR
W
is connected to the demodulator
¾The antenna signals themselves could be sampled and the
TU
best one sent to a single demodulation
JN
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NCCU
7-36 Wireless Comm. Lab.
ld
Diversity Techniques
or
• Feedback or scanning diversity (see Fig. 14)
¾ The signal, the best of M signals, is received until it falls
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below threshold and the scanning process is again initiated
TU
JN
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NCCU
7-37 Wireless Comm. Lab.
ld
Diversity Techniques
or
• Maximal ratio combining [15] (see Fig. 12)
W
¾ The signals from all of the M branches are weighted
according to their signal voltage to noise power ratios and
then summed
TU
• Equal gain diversity
¾ The branch weights are all set to unity but the signals from
JN
each are co-phased to provide equal gain combining
diversity
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A
NCCU
7-38 Wireless Comm. Lab.
ld
Diversity Techniques
or
• Polarization diversity
¾ Theoretical model for polarization diversity [16] (see Fig.15)
W
x = r1 cos(ωt + φ1 )
the signal arrive at the base station
y = r2 cos(ωt + φ2 )
the correlation coefficient can be written as
TU
2
⎛ tan 2 (α ) cos2 ( β ) − Γ ⎞
ρ = ⎜⎜ 2 ⎟⎟
⎝ tan (α ) cos 2
(β ) + Γ ⎠
JN
R22
Γ =
R12
NCCU
7-39 Wireless Comm. Lab.
ld
Diversity Techniques
or
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TU
JN
ll
Fig. 15 Theoretical Model for base station polarization diversity based on [Koz85]
A
NCCU
7-40 Wireless Comm. Lab.
ld
Diversity Techniques
or
• Frequency diversity
¾ Frequency diversity transmits information on more than one
W
carrier frequency
¾ Frequencies separated by more than the coherence bandwidth
TU
of the channel will not experience the same fads
• Time diversity
JN
¾ Time diversity repeatedly transmits information at time
spacings that exceed the coherence time of the channel
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A
NCCU
7-41 Wireless Comm. Lab.
ld
RAKE Receiver
or
• RAKE Receiver [17]
M Z m2
Z′ = ∑αm Z m αm =
W
M
m =1 ∑ Z m2
m =1
TU
JN
Fig. 16 An M-branch (M-finger) RAKE receiver implementation. Each correlator detects a time shifted
version of the original CDMA transmission, and each finger of the RAKE correlates to a portion of the
ll
signal which is delayed by at least one chip in time from the other finger.
A
NCCU
7-42 Wireless Comm. Lab.
ld
Interleaving
or
W
TU
JN
Fig. 17 Block interleaver where source bits are read into columns and out as n-bit rows
ll
A
NCCU
7-43 Wireless Comm. Lab.
ld
References
or
[1] T. S. Rappaport, Wireless Communications -- Principles and Practice, Prentice Hall Inc., New Jersey, 1996.
[2] S.U.H. Qureshi, “Adaptive equalization, ” Proceeding of IEEE, vol. 37 no.9, pp.1340 -- 1387, Sept. 1985.
[3] J. R. Treichler, and B.G. Agoe, “A new approach to multipath correction of constant modulus signals, ”
W
IEEE Trans. Acoustics, Speech, and Signal Processing, vol. ASSP--31, pp. 459--471, 1983
[4] W. A. Gardner, “Exploitation of spectral redundancy in cyclostationary signals, ” IEEE Signal Processing
Magazine, pp. 14-- 36, April 1991.
[5] I.Korn, Digital Communications, Van Nostrand Reinhold, 1985.
TU
[6] J. Proakis, “Adaptive equalization for TDMA digital mobile radio, ” IEEE Trans. Commun., vol. 40, no.2,
pp.333--341, May 1991.
[7] J. A. C. Bingham, The Theory and Practice of Modem Design, John Wiley & sons, New York.
[8] C. A, Belfiori, and J.H. Park, “Decision feedback equalization, ” Proceedings of IEEE, vol. 67, pp. 1143--
JN
1156, Aug. 1979.
[9] K. Zhou, J.G. Proakis, F. Ling, “Decision feedback equalization of time dispersive channels with coded
modulation, ” IEEE Trans. Commun., vol. 38, pp. 18--24, Jan. 1990.
ll
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NCCU
7-44 Wireless Comm. Lab.
ld
References
or
[10] G. D. Forney, “The Viterbi algorithm, ” Proceedings of the IEEE, vol.61, no.3, pp. 268--278, March 1978.
[11] B. Widrow, and S.D. Stearns, Adaptive Signal Processing, Prentice Hall, 1985.
W
[12] S. Haykin, Adaptive Filter Theory, Prentice Hall, Englewood Cliffs, NJ, 1986.
[13] T. Eng, N. Kong, and L. B. Milstein, “Comparison of Diversity Combining Techniques for Rayleigh-
Fading Channels,” IEEE Trans. Commun., vol. 44, pp. 1117-1129, Sep. 1996.
[14] W. C. Jakes, “A Comparision of Space Diversity Techniques for Reduction of Fast Fading in UHF Mobile
Radio Systems,” IEEE Trans. Veh. Technol., vol. VT-20, No. 4, pp. 81-93,
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Nov. 1971.
[15] L. Kahn, “Radio Square,” Proceedings of IRE, vol. 42, pp. 1074, Nov. 1954.
[16] S. Kozono, et al, “Base Station Polarization Diversity Reception for Mobile Radio,” IEEE Trans. Veh.
Technol., vol VT-33, No. 4, pp. 301-306, Nov. 1985.
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[17] R. Price, P. E. Green, “A Communication Technique for Multipath Channel,” Proceeding of the IRE, pp.
555-570, March 1958
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NCCU
7-45 Wireless Comm. Lab.
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Chapter 7:
Equalization and
or
Diversity
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School of Information Science
TU
and Engineering, SDU
J N
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A
ld
Outline
or
l Introduction
l Fundamentals of Equalization
W
l Survey of Equalization Techniques
l Linear Equalizers
l Nonlinear Equalization
l
l
l
TU
Algorithms for Adaptive Equalization
Fundamentals of diversity
Survey of Diversity Techniques
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l Frequency/Time/Space/Polarization Diversity
l Selection/MRC/EGC Combining
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l RAKE Receiver
l Interleaving
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A
ld
7.1 Introduction
or
l The properties of mobile radio channels:
l Multipath fading -> time dispersion, ISI
W
l Doppler spread -> dynamical fluctuation
These effects have a strong negative impact on the bit error rate of any
modulation.
l
l
TU
Mobile communication systems require signal processing
techniques that improve the link performance in hostile
mobile radio environments.
Three popular techniques:
N
l Equalization: compensates for ISI
J
or
Channel model:
k
W
Received signal: r (t ) = s (t ) ∗ h(t ) = ∑ α k s (t − τ k )
k
Ts Ts
TU α1
N
α3
α4
J
α2
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A
ld
(1) Equalization
or
l If the modulation bandwidth exceeds the coherence bandwidth
of the radio channel, ISI occurs and modulation pulses are
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spread in time.
TU
An equalizer within a receiver compensates for the average range
of expected channel amplitude and delay characteristics.
N
l Equalizers must be adaptive
since the channel is generally unknown and time varying.
J
ll
A
ld
(2) Diversity
or
l Usually employed to reduce the depth and duration of the
fades experienced by a receiver in a flat fading (narrowband)
W
channel.
Without increasing the transmitted power or bandwidth.
l Can be employed at both base station and mobile receivers.
l
TU
Types of diversity:
.antenna polarization diversity
.frequency diversity
N
.time diversity.
For example, CDMA systems often use a RAKE receiver,
J
or
l Used to Improve mobile communication link performance by
adding redundant data bits in the transmitted message.
W
At the baseband portion of the transmitter, a channel coder
maps a digital message sequence into another specific code
sequence containing a greater number of bits than originally
contained in the message.
l
wireless channel.TU
The coded message is then modulated for transmission in the
N
l coding can be considered to be a post detection technique.
Because decoding is performed after the demodulation portion
J
block codes
convolutional codes.
A
ld
or
l Channel coding is generally treated independently from the
type of modulation used
W
but this has changed recently with the use of trellis coded
modulation schemes that combine coding and modulation to
achieve large coding gains without any bandwidth expansion.
Notes
l
TU
The three techniques of equalization, diversity, and channel
N
coding are used to improve radio link performance (i.e. to
minimize the instantaneous bit error rate)
J
systems.
A
7.2 Fundamentals of Equalization
ld
or
l Intersymbol interference (ISI)
W
l caused by multipath propagation (time dispersion) ;
l cause bit errors at the receiver;
l the major obstacle to high speed data transmission over
TU
mobile radio channels.
l Equalization
N
l a technique used to combat ISI;
l can be any signal processing operation that minimizes ISI;
J
or
l Training (first stage)
l A known fixed-length training sequence is sent by the
transmitter so that the receiver's equalizer may average to a
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proper setting.
l The training sequence is designed to permit an equalizer at
the receiver to acquire the proper filter coefficients in the
worst possible channel conditions
TU
The training sequence is typically a pseudorandom binary signal
or a fixed, prescribed bit pattern.
Immediately following the training sequence, the user data is
sent.
N
l The time span over which an equalizer converges is a
function of
J
or
l Tracking (second stage)
Immediately following the training sequence, the user data is sent.
W
l As user data are received, the adaptive algorithm of the
equalizer tracks the changing channel and adjusts its filter
characteristics over time.
l
TU
commonly used in digital communication systems
where user data is segmented into short time blocks.
N
l TDMA wireless systems are particularly well suited for
equalizers.
J
or
l Equalizer can be implemented at baseband or at IF in a receiver.
l Since the baseband complex envelope expression can be used
W
to represent bandpass waveforms and, thus, the channel
response, demodulated signal, and adaptive equalizer
algorithms are usually simulated and implemented at baseband
TU
Block diagram of a simplified communications system using an
adaptive equalizer at the receiver is shown in next page
J N
ll
A
ld
Communication system with an adaptive equalizer
or
Radio
x(t )
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Modulator Transmitter RF Front End
Channel
Detector
nb (t ) + IF Stage
TU
Matched Filter
f (t )
y (t )
Adaptive Decision
N
heq (t ) Equalizer Maker
d (t )
dˆ (t )
J
Σ
ll
e(t )
A
ld
Relevant equations
or
y (t ) = x(t ) ∗ f (t ) + nb (t )
dˆ (t ) = x(t ) ∗ f (t ) ∗ heq (t ) + nb (t ) ∗ heq (t )
W
heq (t ) = ∑ ck δ (t − nTs )
TU
k
heq (t ) ∗ f (t ) = δ (t ) H eq ( f ) =
1
N
F( f )
J
frequencies
A
ld
or
yk yk-1 yk-2
W
Z-1 Z-1 Z-1 Z-1
w0 w1
TU w2 wN
dˆk
Σ
N
Adaptive algorithm that updates the weights
ek
J
Σ
ll
Prior knowledge: d k
A
7.3 A Generic Adaptive Equalizer
ld
or
l A transversal filter with
l N delay elements
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l N+1 taps
l N+1 weights:
l
algorithm TU
These weights are updated continuously by the adaptive
signal.
A
ld
7.3 A Generic Adaptive Equalizer
or
l A cost function is used
the cost function is minimized by using ekThe, and the weights
W
are updated iteratively.
l For example, The least mean squares (LMS) algorithm can
serve as a cost function.
TU
l Iterative operation based on LMS
New weights = Previous weights + (constant) x (Previous error) x (Current input vector)
Where
Previous error = Previous desired output — Previous actual output
N
This process is repeated rapidly in a programming loop while the
equalizer attempts to converge
J
or
l Techniques used to minimize the error
l gradient
W
l steepest decent algorithms
or
Blind algorithms
l more recent class of adaptive algorithms
W
l able to exploit characteristics of the transmitted signal and do
not require training sequences.
provide equalizer convergence without burdening the
transmitter with training overhead
TU
able to acquire equalization through property restoral
techniques of the transmitted signal,
l Two techniques:
l the constant modulus algorithm (CMA)
N
used for constant envelope modulation
forces the equalizer weights to maintain a constant envelope on
J
ld
Receiver
or
l Because noise is present, an equalizer is unable to achieve
perfect performance.
W
l Therefore, the instantaneous combined frequency response
will not always be flat, resulting in some finite prediction error.
l The mean squared error (MSE) E [ek2] is one of the most
l
TU
important measures of how well an equalizer works.
Minimizing MSE E [ek2] tends to reduce the bit error rate.
For wireless communication links, it would be best to minimize
N
the instantaneous probability of error instead of MSE
generally results in nonlinear equations
J
or
l Equalization techniques can be subdivided into two general
categories:
W
l linear equalization
l The output of the decision maker is not used in the feedback
path to adapt the equalizer.
l nonlinear equalization
l
TU
l The output of the decision maker is used in the feedback path
to adapt the equalizer.
or
Equalizer
W
Linear Nonlinear
Types
TU
ML Symbol
DFE MLSE
Detector
N
Transversal
Structures Transversal Lattice Transversal Lattice
Channel Est.
J
RLS
Algorithms RLS Fast RLS Fast RLS
A
or
l made up of tapped delay lines, with the tappings spaced a
symbol period (Ts) apart
W
l the transfer function can be written as a function of the delay
operator − jωTs or Z −1
Assuming that the delay elements have unity gain and delay Ts,
TU
of a linear N
J
ll
A
or
Two types of LTE
W
l finite impulse response (FIR) filter
l The simplest LTE uses only feedforward Z −1 taps
l Transfer function is a polynomial in
l has many zeroes but poles only at z = 0
l
TU
Usually simply called a transversal filter
and zeros.
l tend to be unstable when used in channels where the
strongest pulse arrives after an echo pulse (i.e., leading
ll
echoes)
rarely used.
A
ld
or
W
TU
J N
ll
Tapped delay line filter with both feedforward and feedback taps (IIR)
A
ld
7.6 Linear Equalizers
or
Transversal filter implementation (LTE)
W
Input
TU
J N
Output
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Threshold Detector
A
or
l current and past values of the received signal are linearly
weighted by the filter coefficient and summed to produce the
W
output,
If the delays and the tap gains are analog, the continuous output
of the equalizer is sampled at the symbol rate and the samples are
applied to the decision device.
TU
Implementation is usually carried out in the digital domain where
the samples of the received signal are stored in a shift register.
or
Lattice filter implementation
W
TU
J N
ll
A
or
l Two main advantages of the lattice equalizer
l numerical stability
W
l faster convergence
or
l Linear equalizers do not perform well on channels which have
deep spectral nulls in the passband.
W
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.
TU
Nonlinear equalizers are used in applications where the
channel distortion is too severe for a linear equalizer to
handle.
N
l Three very effective nonlinear equalizer
l Decision Feedback Equalization (DFE)
J
ld
(DFE)
or
Basic idea:
once an information symbol has been detected, the ISI
W
that it induces on future symbols can be estimated and
subtracted out before detection of subsequent symbols.
l
TU
as a lattice filter.
ld
(DFE)
or
Input
W
TU
Feedforward Filter
Output
J N
ll
A
Feedback Filter
7.7.1 Decision Feedback Equalization
ld
(DFE)
or
The output of DFE
W
The minimum mean square error of DFE
TU
• It can be seen that the minimum MSE for a DFE is always
smaller than that of an LTE
N
Unless F (e jωT ) is a constant, where adaptive equalization is not
needed
J
ld
(DFE)
or
Conclusion
W
l an LTE is well behaved when the channel spectrum is
comparatively flat
l a DFE is more appropriate for severely distorted wireless
channels.
l
spectrum
l
TU
If the channel is severely distorted or exhibits nulls in the
channel
where the strongest energy arrives after the first arriving signal
component.
ll
A
ld
Another form of DFE----predictive DFE
or
l also consists of a feed forward filter (FFF) as in the
conventional DFE.
W
l Difference: the feedback filter (FBF) is driven by an input
sequence formed by the difference of the output of the
detector and the output of the feed forward filter.
TU
the FBF here is called a noise predictor because it predicts
the noise and the residual ISI contained in the signal at the FFF
output and subtracts from it
N
l The predictive DFE performs as well as the conventional
DFE as the limit in the number of taps in the FFF and the
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or
W
TU
J N
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A
7.7.2 Maximum Likelihood Sequence
ld
Estimation (MLSE) equalizer
or
The MSE-based linear equalizers are optimum with respect to the
criterion of minimum probability of symbol error when the channel
W
does not introduce any amplitude distortion.
Yet this is precisely the condition in which an equalizer is needed
for a mobile communications link.
l
receiver structure.
TU
MLSE uses various forms of the classical maximum likelihood
ld
Estimation (MLSE) equalizer
or
W
Matched
Filter
TU
Delay
N
Channel
Estimator
J
ll
A
7.7.2 Maximum Likelihood Sequence
ld
Estimation (MLSE) equalizer
or
l The MLSE can be viewed as a problem in estimating the state
of a discrete time finite state machine
W
The channel has ML states, where M is the size of the symbol
alphabet of the modulation.
TU
The Viterbi algorithm then tracks the state of the channel by the
paths through the trellis.
N
l The MLSE is optimal in the sense that it minimizes the
probability of a sequence error.
J
ll
A
7.7.2 Maximum Likelihood Sequence
ld
Estimation (MLSE) equalizer
or
NOTES:
W
l The MLSE requires knowledge of the channel characteristics
in order to compute the metrics for making decisions.
TU
l
distribution of the noise corrupting the signal
the probability distribution of the noise determines the form of the
metric for optimum demodulation of the received signal.
N
l The matched filter operates on the continuous time signal,
whereas the MLSE and channel estimator rely on discretized
J
(nonlinear) samples.
ll
A
ld
7.8 Algorithms for Adaptive Equalization
or
l Equalizer requires a specific algorithm to update the coefficients
and track the channel variations.
W
Since it compensates for an unknown and time-varying channel
or
Factors determining the performance of an algorithm:
W
l Rate of convergence (fast or slow?)
l Defined as the number of iterations required for the algorithm,
in response to stationary inputs, to converge close enough to
the optimum solution.
l
l
TU
A fast rate of convergence allows the algorithm to adapt
rapidly to a stationary environment of unknown statistics.
Furthermore, it enables the algorithm to track statistical
variations when operating in a nonstationary environment.
N
l Misadjustment (precise or not?)
J
or
Factors determining the performance of an algorithm:
W
l Computational complexity (simple or complex?)
l Number of operations required to make one complete
iteration of the algorithm.
l
l TU
Numerical properties (stable or not?)
When an algorithm is implemented numerically, inaccuracies
are produced due to round-off noise and representation
N
errors in the computer.
l These kinds of errors influence the stability of the algorithm.
J
ll
A
ld
7.8 Algorithms for Adaptive Equalization
or
Practical considerations for choice of an equalizer structure and
its algorithm
W
l The cost of the computing platform (affordable or not?)
especially when used in user equipments
l
TU
The power budget (power limited applications or else?)
In portable radio applications, battery drain at the subscriber unit
is a paramount consideration
N
l The radio propagation characteristics (fast fading & time
delay spread?)
J
The speed of the mobile unit determines the channel fading rate
and the Doppler spread, which is directly related to the coherence
ll
or
Three classic equalizer algorithms
W
l Zero Forcing Algorithm (ZF)
l Least Mean Square Algorithm (LMS)
TU
l Recursive Least Squares Algorithm (RLS)
N
Please read references for detailed
J
or
Criterion:
to force the samples of the combined channel and equalizer
W
impulse response to zero at all but one of sample points in the
tapped delay line filter.
Disadvantage:
TU
may excessively amplify noise at frequencies where the
folded channel spectrum has high attenuation.
N
Suitability:
J
Wireline communications
1 1
ll
H eq ( f ) = , f <
H ch ( f ) 2T
A
7.8.2 Least Mean Square (LMS)
ld
Algorithm
or
Criterion:
W
to minimize the mean square error (MSE) between the desired
equalizer output and the actual equalizer output.
Minimize ξ = E[ek ⋅ ek ]
*
TU
Must be solved iteratively
Simplest algorithm, requires only 2N + I operations per iteration.
α
J
ld
Algorithm
or
Disadvantage: low convergence rate.
Because of the only one parameter α
W
Especially when the eigenvalues of the input covariance matrix
RNN have a very large spread, i.e, λmax / λmin >> 1
l
TU
greater than the propagation delay through the equalizer, then
the equalizer will be unable to reduce distortion.
l
order to avoid instability in the equalizer [Hay86].
A
since
7.8.3 Recursive Least Squares (RLS)
ld
Algorithm
or
l RLS is Proposed to improve the convergence rate of LMS algorithm.
W
l Error measures expressed in terms of a time average of the actual
received signal instead of a statistical average.
l
the equalizer.
TU
λ is the weighting coefficient that can change the performance of
l The smaller the λ , the better the tracking ability of the equalizer.
l However, if λ is too small, the equalizer will be unstable
ll
A
7.8.3 Recursive Least Squares (RLS)
ld
Algorithm
or
Advantage: high convergence rate
W
Disadvantage: sometimes unstable
TU
The RLS algorithm described above, called the
Kalman RLS algorithm
N
Uses 2.5N2 + 4.5N arithmetic operations per
iteration.
J
ll
A
ld
7.8.4 Summary of equalization algorithms
or
l There are number of variations of the LMS and RLS algorithms
l RLS algorithms have similar convergence and tracking
W
performances, which are much better than the LMS algorithm.
Usually have high computational requirement and complex
program structures.
Some RLS algorithms tend to be unstable.
About FTF
TU
l Among the RLS algorithms, fast transversal filter (FTF)
algorithm requires the least computation
N
l a rescue variable can be used to avoid instability.
However, rescue techniques tend to be a bit tricky for widely
J
ld
or
W
TU
J N
ll
A
ld
7.9 Fractionally Spaced Equalizers(FSE)
or
l In the presence of channel distortion, the matched filter prior to
the equalizer must be matched to the channel and the corrupted
W
signal.
Usually get the suboptimal result because the channel response is
unknown.
This results in a significant degradation in performance.
l
the Nyquist rate. TU
FSE is based on sampling the incoming signal at least as fast as
N
l The FSE compensates for the channel distortion before aliasing
effects occur due to the symbol rate sampling.
J
ld
Techniques
or
l Random nature of radio propagation:
l Multipath propagation
W
l If one radio path undergoes a deep fade, another
independent path may have a strong signal
l
TU
Diversity exploits the random nature of radio propagation by
finding independent signal paths for communication, so as to
boost the instantaneous SNR at the receiver.
N
Path 2
J
Path 3
A
7.10 Fundamentals of Diversity
ld
Techniques
or
l Diversity is a powerful communication receiver technique that
provides wireless link improvement at relatively low cost.
W
l Requires no training
TU
receiver, and are unknown to the transmitter.
ld
Techniques
or
Microscopic diversity
l Small-scale fades: deep and rapid amplitude fluctuations over
distances of just a few wavelengths.
W
caused by multiple reflections from the surroundings in the vicinity
of the mobile.
results in a Rayleigh fading distribution of signal strength over small
distances.
l
changing signal.
TU
Microscopic diversity techniques can exploit the rapidly
N
For example, use two antennas at the receiver (separated by a
fraction of a meter), one may receive a null while the other receives a
strong signal.
J
ld
Techniques
or
Macroscopic diversity
W
l Large-scale fading: caused by shadowing due to variations in
both the terrain profile and the nature of the surroundings.
In deeply shadowed conditions, the received signal strength at a
mobile can drop well below that of free space.
TU
log-normally distributed with a standard deviation of about 10 dB in
urban environments.
ld
Techniques
or
Macroscopic diversity
W
l Macroscopic diversity is also useful at the base station receiver.
By using base station antennas that are sufficiently separated in
space, the base station is able to improve the reverse link by selecting
the antenna with the strongest signal from the mobile.
l
TU
Used to combat slow fading (shadowing)
ld
Techniques
or
Macro-scope diversity
W
Base station Base station
TU
N
Mobile
J
ll
A
7.10 Fundamentals of Diversity
ld
Techniques
or
l Strategies used in diversity techniques
l Selection diversity
W
l Maximal ratio combining diversity
l Hybrid schemes
l TU
Practical considerations
l effectiveness, complexity, cost, and etc.
J N
ll
A
7.10.1 Derivation of Selection
ld
Diversity improvement
or
l Consider M independent Rayleigh fading channels available
areceiver.
W
Each channel is called a diversity branch.
TU
J N
ll
A
7.10.1 Derivation of Selection
ld
Diversity improvement
or
l Further assume that each branch has the same average SNR
given by
W
Where we assume α = 1 .
2
ld
Diversity improvement
or
l Now, the probability that all M independent diversity branches
receive signals which are simultaneously less than some
W
specific SNR threshold γ is
l TU
This is the probability of all branches failing to achieve SNR = γ i .
If a single branch achieves SNR > γ , then the probability that
SNR > γ for one or more branches is given by
J N
This is the probability of exceeding a threshold when selection
diversity is used.
ll
A
A
ll
JN
TU
W
or
ld
7.10.1 Derivation of Selection
ld
Diversity improvement
or
How to determine the average signal-to-noise ratio of the received
signal when diversity is used?
W
l First of all, find the pdf of γ (the instantaneous SNR when M
branches are used). Thus we compute the derivation of CDF
PM (γ ) ,
l TU
Then, we can compute the average SNR, γ ,
N
where x = γ / Γ .
J
Γ k =1 k
7.10.1 Derivation of Selection
ld
Diversity improvement
or
l Selection diversity offers an average improvement in the link
margin without requiring additional transmitter power or
sophisticated receiver circuitry.
W
The diversity improvement can be directly related to the average
bit error rate for various modulations.
TU
Selection diversity is easy to implement because all that is
needed is a side monitoring station and an antenna switch at
the receiver.
N
l However, it is not an optimal diversity technique because it
J
phased and weighted manner such that the highest achievable SNR
is available at the receiver at all times.
A
7.10.1 Derivation of Selection
ld
Diversity improvement
or
Example
Assume four branch diversity is used, where each branch
receives an independent Rayleigh fading signal. If the average
W
SNR is 20 dB, determine the probability that the SNR will drop
below 10 dB. Compare this with the case of a single receiver
without diversity.
Solution
TU
J N
ll
A
7.10.2 Derivation of Maximal Ratio
ld
Combining Improvement
or
In maximal ratio combining, the voltage signals ri from each of
the M diversity branches are co-phased to provide coherent voltage
W
addition and are individually weighted to provide optimal SNR.
TU
J N
ll
A
7.10.2 Derivation of Maximal Ratio
ld
Combining Improvement
or
1) The SNR out of the diversity combiner:
l If each branch has gain Gi , then the resulting signal envelope
W
applied to the detector is
l
TU
Assuming that each branch has the same average noise power
N, the total noise power NT applied to the detector is simply the
weighted sum of the noise in each branch. Thus
J N
ld
Combining Improvement
or
l Using Chebychev's inequality, γ M is maximized
when Gi = ri / N , which leads to
W
(7-66)
TU
l Conclusion:
The SNR out of the diversity combiner is simply the sum of
the SNRs in each branch.
J N
ll
A
7.10.2 Derivation of Maximal Ratio
ld
Combining Improvement
or
2) The pdf of γ M
W
l According to Chapter 3, γ M is a Chi-square distribution of 2M
Gaussian random variables. Thus, the pdf for γ M is
TU
(7-68)
3) The CDF of γ M
N
l γM
According to the abovementioned pdf, The probability that
is less than some SNR threshold γ is
J
ll
A
7.10.2 Derivation of Maximal Ratio
ld
Combining Improvement
or
γM
4) The average SNR out of the diversity combiner,
W
l γ M can be calculated by using the pdf of γ M (Eq. (7.68)). But the
direct way is to calculate it from Eq. (7-66).
l
TU
That is to say, the average SNR, γ M , is simply the sum of the
individual γ i from each branch.
N
The control algorithms for setting the gains and phases for
J
ld
Considerations
or
l Space diversity (also known as antenna diversity), is one of the
most popular forms of diversity used in wireless systems.
W
l The signals received from spatially separated antennas on the
mobile would have essentially uncorrelated envelopes for
antenna separations of one half wavelength or more.
l
or both.
TU
Space diversity can be used at either the mobile or base station,
N
Since the important scatterers are generally on the ground in the
J
ld
Considerations
or
general block diagram of a space diversity scheme
W
TU
J N
ll
A
7.10.3 Practical Space Diversity
ld
Considerations
or
Space diversity reception methods can be classified into four
categories
W
l 1. Selection diversity
l 2. Feedback diversity
TU
l 3. Maximal ratio combining
l 4. Equal gain diversity
J N
ll
A
7.10.3 Practical Space Diversity
ld
Considerations
or
(1) Selection Diversity
W
l The simplest diversity technique.
l The receiver branch having the highest instantaneous SNR is
connected to the demodulator.
l
l
TU
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.
N
l A practical selection diversity system cannot function on a
truly instantaneous basis, but must be designed so that the
J
ld
Considerations
or
(2) Feedback or Scanning Diversity
l Very similar to selection diversity
W
l The M signals are scanned in a fixed sequence until one is
found to be above a predetermined threshold.
l This signal is then received until it falls below threshold and
TU
the scanning process is again initiated.
l The resulting fading
statistics are
somewhat inferior to
N
those obtained by
the other methods.
J
l Advantage: very
simple to implement
ll
ld
Considerations
or
(3) Maximal Ratio Combining
l The signals from all of the M branches are weighted and then
W
summed.
l The individual signals must be co-phased before being
summed.
TU
requires an individual receiver and phasing circuit for each
antenna element.
l Output SNR equal to the sum of the individual SNRs.
l Advantage: produces an output with an acceptable SNR even
N
when none of the individual signals are themselves acceptable.
l Gives the best statistical reduction of fading of any known
J
ld
Considerations
or
Maximal Ratio Combiner
W
TU
J N
ll
A
7.10.3 Practical Space Diversity
ld
Considerations
or
(4) Equal Gain Combining
W
In certain cases, it is not convenient to provide for the variable
weighting capability required for true maximal ratio combining. In
such cases, the branch
l
TU
Equal gain combining diversity sets all weights to unity but the
signals from each branch are co-phased.
The possibility of producing an acceptable signal from a
N
number of unacceptable inputs is still retained,
l The performance is only marginally inferior to maximal ratio
J
or
At the base station, space diversity is considerably less practical .
W
l polarization diversity only provides two diversity branches, but
allows the antenna elements to be co-located.
l Measured horizontal and vertical polarization paths between a
TU
mobile and a base station are reported to be uncorrelated.
l Decorrelation for the signals in each polarization is caused by
multiple reflections.
l The reflection coefficient for each polarization is different,
N
which results in different amplitudes and phases for each, or at
least some, of the reflections.
J
or
l Transmits information on more than one carrier frequency.
frequencies separated by more than the coherence bandwidth of
W
the channel will not experience the same fades.
l
TU
In practice, 1:N protection switching is provided by a radio licensee,
When diversity is needed, the appropriate traffic is simply switched
to the backup frequency.
N
Disadvantage: not only requires spare bandwidth but also
J
l
requires that there be as many receivers as there are channels
used for the frequency diversity.
ll
or
l New OFDM modulation and access techniques exploit
frequency diversity by providing simultaneous modulation
W
signals with error control coding across a large bandwidth.
TU
J N
ll
A
ld
7.10.6 Time Diversity
or
l Time diversity repeatedly transmits information at time
spacings that exceed the coherence time of the channel
W
Multiple repetitions of the signal will be received with independent
fading conditions.
TU
l One modem implementation of time diversity involves the use
of the RAKE receiver for spread spectrum CDMA, where the
multipath channel provides redundancy in the transmitted
message.
J N
ll
A
ld
7.11 RAKE Receiver
or
l In CDMA spread spectrum systems, the spreading codes are
designed to provide very low correlation between successive
W
chips.
TU
receiver, and equalization is not required.
or
The RAKE receiver is essentially a diversity receiver
designed specifically for CDMA, where the diversity is provided
W
by the fact that the multipath components are practically
uncorrelated from one another when their relative propagation
delays exceed a chip period.
TU
J N
ll
An M branch (M-finger) RAKE receiver implementation. Each correlator detects a time shifted
version of the original CDMA transmission, and each finger of the RAKE correlates to a portion
A
of the signal which is delayed by at least one chip in time from the other fingers.
ld
7.12 Interleaving
or
l Interleaving is used to obtain time diversity in a digital
communications system without adding any overhead.
W
useful technique in all second and third generation digital cellular
systems.
l
TU
"important" bits in succession.
or
Two types of interleaver:
1) Block structure
W
l Formats the encoded data into a rectangular array of m rows
and n columns, and interleaves nm bits at a time.
l Usually, each row contains a word of source data having n bits.
An interleaver of degree m (or depth m) consists of m rows.
TU
J N
ll
A
ld
7.12 Interleaving
or
l Source bits are placed into the interleaver by sequentially
W
increasing the row number for each successive bit, and filling
the columns.
TU
transmitted over the channel.
This has the effect of separating the original source bits by m bit
periods.
N
l At the receiver, the de-interleaver stores the received data by
sequentially increasing the row number of each successive bit,
J
and then clocks out the data row-wise, one word (row) at a time.
ll
A
ld
7.12 Interleaving
or
Delay introduced by interleaving
W
l There is an inherent delay associated with an interleaver .
since the received message block cannot be fully decoded until
all of the nm bits arrive at the receiver and are de-interleaved.
l
than 40 ms occur.
TU
Human speech is tolerable to listen to until delays of greater
or
2) Convolutional structure
W
l Can be used in place of block interleavers in much the same
fashion.
l Ideally suited for use with convolutional codes.
TU
J N
ll
A
7.13 Fundamental of
ld
Channel Coding
or
l Channel coding protects digital data from errors by selectively
introducing redundancies in the transmitted data.
W
l Two types of Channel codes
1) error detection codes
2) error correction codes.
l TU
The basic purpose of Channel Coding:
Introduce redundancies in the data to improve wireless link
performance.
N
l Cost: Increases the bandwidth requirement for a fixed source
J
data rate.
This reduces the bandwidth efficiency of the link in high SNR
ll
conditions.
But provides excellent BER performance at low SNR values.
A
7.13 Fundamental of
ld
Channel Coding
or
W
Note:
TU
J N
ll
A
ld
Wireless Networks
or
W
TU
CSG 250
Spring 2005
Rajmohan Rajaraman
JN
ll
A
ld
Transmission Fundamentals
or
o Analog and digital transmission
W
o Channel capacity
o Antennas, propagation modes, and fading
o Signal encoding techniques
TU
Spread spectrum technology
Coding and error control
Cellular networks
JN
Wireless LANs
o IEEE 802.11
ll
o Bluetooth
A
ld
Mobile IP
or
TCP for wireless
Multihop ad hoc networks
W
o MAC and routing protocols
o Power control and topology control
TU
o Capacity of ad hoc networks
Sensor networks
JN
o Infrastructure, MAC, and routing protocols
o Algorithms for query processing
ll
A
ld
Guglielmo Marconi invented the wireless telegraph in 1896
or
o Communication by encoding alphanumeric characters in analog signal
o Sent telegraphic signals across the Atlantic Ocean
W
Communications satellites launched in 1960s
Advances in wireless technology
o Radio, television, mobile telephone, communication satellites
More recently
TU
o Satellite communications, wireless networking, cellular technology, ad
hoc networks, sensor networks
JN
ll
A
ld
Target information systems: “Anytime, Anywhere, Any
or
form”
Applications: Ubiquitous computing and information
W
access
Market in continuous growth:
o 35-60% annual growth of PCS
TU
o Number of subscribers:
• by 2001: over 700M mobile phones
• by 2003: 1 billion wireless subscribers (source Ericsson)
JN
o 300% growth in wireless data from 1995-1997
Large diversity of standards and products
Confusing terminology
ll
A
ld
Wireless is convenient and less expensive
or
Limitations and political and technical difficulties
W
inhibit wireless technologies
Lack of an industry-wide standard
Device limitations
TU
o E.g., small LCD on a mobile telephone can only
displaying a few lines of text
JN
o E.g., browsers of most mobile wireless devices use
wireless markup language (WML) instead of HTML
ll
A
W
IMT200, WLAN, ho
DAB, GSM, ad
TETRA, ...
TU
JN
Personal Travel Assistant,
PDA, laptop, GSM, cdmaOne,
WLAN, Bluetooth, ...
ll
A
ld
Wireless technologies have gradually migrated to
or
higher frequencies
W
twisted
pair
1 Mm 10 km
coax cable
100 m
TU 1m 10 mm 100 µm
optical transmission
1 µm
JN
300 Hz 30 kHz 3 MHz 300 MHz 30 GHz 3 THz 300 THz
ld
Wireless:
or
o Limited bandwidth
o Broadcast medium: requires multiple access schemes
W
o Variable link quality (noise, interference)
o High latency, higher jitter
o Heterogeneous air interfaces
o
Mobility:
o
TU
Security: easier snooping
ld
Personal communication systems
or
o Focus on voice communication
o Limited bit-rate data transmission
o Large-scale mobility and coverage
W
o Operate over licensed frequency bands
Wireless LANs
o Designed for high bit-rate transmission
o
o
o
IP oriented
TU
Low-scale coverage
Use unlicensed ISM frequency bands
JN
Multihop ad hoc networks
o Have little or no infrastructure
o Low-scale coverage
o Need new routing protocols
ll
o Emerging applications
A
ld
Electromagnetic signals
or
o Time domain
W
o Frequency domain
Data rate and bandwidth
Analog and digital data transmission
TU
Channel capacity
o Nyquist theorem
JN
o Shannon capacity theorem
Transmission media
ll
A
ld
or
W
TU
JN
ll
A
ld
or
W
TU
JN
ll
A
ld
Transmission medium
or
o Physical path between transmitter and receiver
Guided media
W
o Waves are guided along a solid medium
o E.g., copper twisted pair, copper coaxial cable, optical fiber
Unguided media
TU
o Provides means of transmission but does not guide
electromagnetic signals
o Usually referred to as wireless transmission
JN
o E.g., atmosphere, outer space
ll
A
ld
Transmission and reception are achieved by
or
means of an antenna
W
Configurations for wireless transmission
o Directional
o Omnidirectional
TU
JN
ll
A
ld
or
Microwave frequency range
o 1 GHz to 40 GHz
o Directional beams possible
W
o Suitable for point-to-point transmission
o Used for satellite communications
Radio frequency range
o 30 MHz to 1 GHz
TU
o Suitable for omnidirectional applications
Infrared frequency range
JN
o Roughly, 3x1011 to 2x1014 Hz
o Useful in local point-to-point multipoint applications within confined
areas
ll
A
ld
Description of common microwave antenna
or
o Parabolic "dish", 3 m in diameter
o Fixed rigidly and focuses a narrow beam
W
o Achieves line-of-sight transmission to receiving antenna
o Located at substantial heights above ground level
Applications
TU
o Long haul telecommunications service
o Short point-to-point links between buildings
JN
ll
A
ld
Description of communication satellite
or
o Microwave relay station
o Used to link two or more ground-based microwave
W
transmitter/receivers
o Receives transmissions on one frequency band (uplink), amplifies
or repeats the signal, and transmits it on another frequency
(downlink)
Applications
TU
o Television distribution
o Long-distance telephone transmission
JN
o Private business networks
ll
A
ld
Description of broadcast radio antennas
or
o Omnidirectional
o Antennas not required to be dish-shaped
W
o Antennas need not be rigidly mounted to a precise alignment
Applications
o Broadcast radio
TU
• VHF and part of the UHF band; 30 MHZ to 1GHz
• Covers FM radio and UHF and VHF television
JN
ll
A
ld
Beyond the EHF spectrum
or
o 1012 to 1014 Hz
W
Transceivers must be within line of sight or
reachable via reflection
o Does not penetrate walls
TU
JN
ll
A