Wireless Communication.pdf

Anna University Solved Question Papers B.E./B.Tech. 7th Semester Electronics and Communication Engineering

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ACKNOWLEDGEMENT

Mr K. Murali Babu

Department of Electronics and Communication, GIT, Vellore

Semester-VII Wireless Communication

M02_ECE_7th-SEMESTER_CH01(Nov-Dec-2013).indd 1

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EEE_Sem-VI_Chennai_FM.indd iv

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B.E./B.Tech. DEGREE EXAMINATION, NOV/DEC 2013 Seventh Semester Electronics and Communication Engineering WIRELESS COMMUNICATION Time: Three hours

Maximum: 100 marks

Answer ALL questions PART A (10 ë 2 = 20 marks) 1. What are three most important effectsof small-scalemultipathpropagation? 2. What is a multiple access technique? 3. State the difference between Narrowband and wideband systems. 4. Find the far-field distance for an antenna with maximum dimension of 1 m and operating frequency of 900 MHz. 5. Give the expression for bit error probability of Gaussian Minimum shift Keying modulation. 6. What is fading and Doppler spread? 7. What is Diversity? 8. What is Equalization? 9. What is a PN sequence? Give its significance in spread spectrum modulation technique. 10. What is DECT?

PART B (5 ë 16 = 80 marks) 11. (a) Discuss the types of services, requirements, spectrum limitations and noise considerations of wireless communications. Or


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B.E./B.Tech. Question Papers

(b) Explain the principle of Cellular Networks and various types of Handoff techniques. 12. (a) (i) Briefly explain the factors that influence small-scale fading. (ii) Ifpower a transmitter produces 50 dBW. W of If power, thetotransmit in units of dBM and 50 Wexpress is applied a unity gain antenna with a 900 MHz carrier frequency, find the received power in dBM at a free space distance of 100 m from the antenna. What is Pr (10 km)? Assume unity gain for the receiver antenna. Or (b) (i) Briefly explain the three basic propagation mechanisms which impact propagation in a mobile communication system. (ii) What is Brewster angle? Calculate the Brewster angle for a wave impinging on ground having a permittivity ofer = 4. 13. (a) (i) Explain the Nyquist criterion for ISI cancellation. (ii) With transfer function, explain the raised cosine roll off filter. Or (b) (i) Explain the QPSK transmission and detection techniques. (ii) Explain the performance of Digital modulation in slow flatfading channels. 14. (a) Explain in detail about: (i) Linear Equalizers. (ii) Non Linear Equalizers. Or (b) (i) With block diagram, explain the operation of a RAKE receiver. (ii) Briefly explain the frequency domain coding of speech signals. 15. (a) Explain in detail about: (i) Direct sequence spread spectrum technique. (ii) Frequency hopped spread spectrum technique. Or (b) Discuss in detail about second generation (2G) and third generation (3G) wireless networks and standards.


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Solutions PART A 1. The three most important effects of small-scale multi-path propagation are (i) Rapid changes in signal strength over a small travel distance or time interval (ii) Random frequency modulation due to varying Doppler shifts on different multi-path signals and (iii) Time dispersion (echoes) caused by multi-path propagation delays. 2. Multiple access technique are used to allow many mobile users to share simultaneously a finite amount of radio spectrum. 3. Narrowband systems – Here the available radio spectrum is divided into large number of narrowband channels. The bandwidth of a single channel is nothing but coherence bandwidth of the channel. Wideband systems – The transmission bandwidth of a single channel is much larger than the coherence bandwidth of the channel. 4. Operating frequency, f = 900 MHz l = c/f = 3 × 108 m/s/900 × 106 Hz = 0.33 m Fraunhofer distance, df = 2D2/l = 2(1)2/0.33 = 6 m Path loss PL(dB) = −10 log [(l2)/(4p)2d 2] = −10 log [(0.33)2/(4 × 3.14)2 × 36] = 47dB 5. The error probability for GMSK is given by



Pe = Q 

 

Eb    N 

2g

O



Where g is a constant related to bandwidth-bit duration product (BT) of GMSK filter. g

0. for GMSK with BT 25 0 = . 68 ≅ 0.85 for simple MSK ( BT = ∞)

6. Fading is used to describe the rapid fluctuations of the amplitudes, phases, or multi-path delays of a radio signal over a short period of time or travel distance.


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Doppler spread is a measure of the spectral broadening caused by the time rate of change of the mobile radio channel. 7. Diversity is technique used to compensate for fading channel impairments and is usually implemented by using two or more receiving antennas. 8. Equalization compensates for inter symbol interference (ISI) created by multipath within time dispersive channels. 9. The pseudo-noise (PN) sequence is the spreading codes which is a binary sequence that appears random but can be reproduced in a deterministic manner by intended receivers. Each user is assigned a unique PN code which is approximately orthogonal to the codes of other users, the receiver can separate each user based on their codes, even though they occupy the same spectrum at all times. 10. The Digital European Cordless Telephone (DECT) isa universal cordless telephone standard developed by the European Telecommunications Standards Institute (ETSI). It is the first pan-European standard for cordless telephones and was finalized in July 1992.

PART B 11. (a) Paging Systems Paging systems are communication systems that send brief messages to a subscriber. Depending on the type of service, the message may be either a numeric message, an alphanumeric message, or a voice message. Paging systems are typically used to notify a subscriber of the need to call a particular telephone number or travel to a known location to receive further instructions. In modern paging systems, news headlines, stock quotations, and faxes may be sent. A message is sent to a paging subscriber via the paging system access number (usually a tollfree telephone number) with a telephone keypad or modem. The issued message is called a page. The paging system then transmits the page throughout the service area using base stations which broadcast the page on a radio carrier. Paging systems vary widely in their complexity and coverage area. While simple paging systems may cover a limited range of 2 to 5 km, or may even be confined to within individual buildings, wide area paging systems can provide worldwide coverage. Though paging receivers are simple and inexpensive, the transmission system required


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Wireless Communication (Nov/Dec 2013)

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is quite sophisticated. Wide area paging systems consist of a network of telephone lines, many base station transmitters, and large radio towers that simultaneously broadcast a page from each base station (this is called simulcasting). Simulcast transmitters may be located within the same service area or in different cities or countries. Paging systems are designed to provide reliable communication to subscribers wherever they are; whether inside a building, driving on a highway, or flying in

Figure 1 A wide area paging system. The paging control center dispatches

pages received from the PSTN throughout several cities at the same time.

an airplane. This necessitates large transmitter powers (on the order of kilowatts) and low data rates (a couple of thousand bits per second) for maximum coverage from each base station. Figure shows a diagram of a wide area paging system. Cordless Telephone Systems

Cordless telephone systems are full duplex communication systems that use radio to connect a portable handset to a dedicated base station, which is then connected to a dedicated telephone line with a specific telephone number on the public switched telephone network (PSTN).


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In first generation cordless telephone systems (manufactured in the 1980s), the portable unit communicates only to the dedicated base unit

Figure 2 A cordless telephone system.

and only over distances of a few tens of meters. Early cordless telephones operate solely as extension telephones to a transceiver connected to a subscriber line on the PSTN and are primarily for in-home use. Second generation cordless telephones have recently been introduced which allow subscribers to use their handsets at many outdoor locations within urban centers such as London or Hong Kong. Modern cordless telephones are sometimes combined with paging receivers so that a subscriber may first be paged and then respond to the page using the cordless telephone. Cordless telephone systems provide the user with limited range and mobility, as it is usually not possible to maintain a call if the user travels outside the range of the base station. Typical second generation base stations provide coverage ranges up to a few hundred meters. Figure 2 illustrates a cordless telephone system. Cellular Telephone Systems

A cellular telephone system provides a wireless connection to the PSTN for any user location within the radio range of the system. Cellular systems accommodate a large number of users over a large geographic area, within a limited frequency spectrum. Cellular radio systems provide high quality service that is often comparable to that of the landline telephone systems. High capacity is achieved by limiting the coverage of each base station transmitter to a small geographic area called acell so that the same radio channels may be reused by another base station located some distance away. A sophisticated switching technique called


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a handoff enables a call to proceed uninterrupted when the user moves from one cell to another. Figure 3 shows a basic cellular system which consists ofmobile stations, base stations and a mobile switching center(MSC). The mobile switching center is sometimes called a mobile telephone switching office (MTSO), since it is responsible for connecting all mobiles to the PSTN in a cellular system. Each mobile communicates via radio with one of the base stations and may be handed-off to any number of base stations throughout the duration of a call. The mobile station contains a transceiver, an antenna, and control circuitry, and may be mounted in a vehicle or used as a portable hand-held unit. The base stations consist of several transmitters and receivers which simultaneously handle full duplex communications and generally have towers which support several transmitting and receiving antennas. The base station serves as a bridge

Figure 3 A cellular system. The towers represent base stations which

provide radio access between mobile users and the mobile switching center (MSC). between all mobile users in the cell and connects the simultaneous mobile calls via telephone lines or microwave links to the MSC. The MSC coordinates the activities of all of the base stations and connects the entire cellular system to the PSTN. A typical MSC handles 100,000 cellular subscribers and 5,000 simultaneous conversations at a time, and accommodates all billing and system maintenance functions, as well. In large cities, several MSCs are used by a single carrier.


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11. (b) Cellular Networks Cellular radio systems rely on an intelligent allocation and reuse of channels throughout a coverage region [Oet83]. Each cellular base station is allocated a group of radio channels to be used within a small geographic area called acell. Base stations in adjacent cells are assigned channel groups which contain completely different channels than neighboring cells. The base station antennas are designed to achieve the desired coverage within the particular cell. By limiting the coverage area to within the boundaries of a cell, the same group of channels may be used to cover different cells that are separated from one another by distances large enough to keep interference levels within tolerable limits. The design process of selecting and allocating channel groups for all of the cellular base stations within a system is called orfrequency reuse or frequency planning [Mac79]. The concept of cellular frequency reuse, where cells labeled with the same letter use the same group of channels. The frequency reuse plan is overlaid upon a map to indicate where different frequency channels are used. The hexagonal cell shape is conceptual and is a simplistic model of the radio coverage for each base station, but it has been universally adopted since the hexagon permits easy and manageable analysis of a cellular system. The actual radio coverage of a cell is known as the foot print and is determined from field measurements or propagation prediction models. Although the real footprint is amorphous in nature, a regular cell shape is needed for systematic system design and adaptation for future growth. While it might seem natural to choose a circle to represent the coverage area of a base station, adjacent circles cannot be overlaid upon a map without leaving gaps or creating overlapping regions. Thus, when considering geometric shapes which cover an entire region without overlap and with equal area, there are three sensible choices—a square, an equilateral triangle, and a hexagon. A cell must be designed to serve the weakest mobiles within the footprint, and these are typically located at the edge of the cell. For a given distance between the center of a polygon and its farthest perimeter points, the hexagon has the largest area of the three. Thus, by using the hexagon geometry, the fewest number of cells can cover a geographic region, and the hexagon closely approximates a circular radiation pattern which would occur for an omnidirectional base station antenna and free space propagation. Of course, the actual cellular footprint is determined by the contour in which a given transmitter serves the mobiles successfully. When using hexagons to model coverage areas, base station transmitters are depicted as either being in the center of the cell (center-excited cells) or on


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three of the six cell vertices (edge-excited cells). Normally, omnidirectional antennas are used in center-excited cells and sectored directional antennas are used in corner-excited cells. Practical considerations usually do not allow base stations to be placed exactly as they appear in the hexagonal layout. Most system designs permit a base station to be positioned up to one-fourth the cell radius away from the ideal location. To understand the frequency reuse concept, consider a cellular system which has a total of S duplex channels available for use. If each cell is allocated a group of k channels (k < S), and if the S channels are divided among N cells into unique and disjoint channel groups which each have the same number of channels, the total number of available radio channels can be expressed as

S = kN (1) The N cells which collectively use the complete set of available frequencies is called a cluster. If a cluster is replicatedM times within the system, the total number of duplex channels, C can be used as a measure of capacity and is given by =

C

=

MkN

MS

(2)

Handoff Technique

When a mobile moves into a different cell while a conversation is in progress, the MSC automatically transfers the call to a new channel belonging to the new base station. This handoff operation not only involves identifying a new base station, butalso requires that the voice and control signals be allocated to channels associated with thenew base station. Processing handoffs is an important task in any cellular radio system. Many handoff strategies prioritize handoff requests over call initiation requests when allocating unused channels in a cell site. Handoffs must be performed successfully and as infrequently as possible, and be imperceptible to the users. In order to meet these requirements, system designers must specify an optimum signal level at which to initiate a handoff. Once a particular signal level is specified as the minimum usable signal for acceptable voice quality at the base station receiver (normally taken as between −90 dBm and −100 dBm), a slightly stronger signal level is used as a threshold at which a handoff is made. This margin, given by ∆ = Pr handoff − Pr minimum usable, cannot be too large or too small. If ∆ is too large, unnecessary handoffs which burden the MSC may occur, and if ∆ is too small, there may be insufficient time to complete a handoff before a call is lost due to weak signal conditions. Therefore, ∆ is chosen carefully to meet these conflicting requirements. Figure 1 illustrates a handoff situation. Figure 1 demonstrates the case where a handoff is not


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made and the signal drops below the minimum acceptable level to keep the channel active. This dropped call event can happen when there is an excessive delay by the MSC in assigning a handoff or when the threshold ∆ is set too small for the handoff time in the system. Excessive delays may occur during high traffic conditions due to computational loading at the MSC or due to the fact that no channels are available on any of the

Figure 1

nearby base stations (thus forcing the MSC to wait until a channel in a nearby cell becomes free). Types: the two types of Handoff techniques are: Hard Handoff: With hard handoff, the link to the prior base station is terminated before or as the user is transferred to the new cell’s base station. That is to say that the mobile is linked to no more than one base station at a given time.


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Soft Handoff: CDMA uses soft handoff. Soft handoff is beneficial

because it reduces interference into other cells and improves performance by using macro diversity. 12. (a) (i) Factors Influencing Small-Scale Fading Many physical factors in the radio propagation channel influence smallscale fading. These include the following: • Multipath propagation — The presence of reflecting objects and scatterers in the channel creates a constantly changing environment that dissipates the signal energy in amplitude, phase, and time. These effects result in multiple versions of the transmitted signal that arrive at the receiving antenna, displaced with respect to one another in time and spatial orientation. The random phase and amplitudes of the different multipath components cause fluctuations in signal strength, thereby inducing small-scale fading, signal distortion, or both. Multipath propagation often lengthens the time required for the baseband portion of the signal to reach the receiver which can cause signal smearing due to intersymbol interference. •

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. Doppler shift will be positive or negative depending on whether the mobile receiver is moving toward or away from the base station. • Speed of surrounding objects — If objects in the radio channel are in motion, they induce a time varying Doppler shift on multipath components. If the surrounding objects move at a greater rate than the mobile, then this effect dominates the small-scale fading. Otherwise, motion of surrounding objects may be ignored, and only the speed of the mobile need be considered. Thecoherence time defines the “staticness” of the channel, and is directly impacted by the Doppler shift. •

The transmission bandwidth of the — If the transmitted radio signal bandwidth is greater than thesignal “bandwidth” of the multipath

channel, the received signal will be distorted, but the received signal strength will not fade much over a local area (i.e., the small scale signal fading will not be significant). As will be shown, the bandwidth of the channel can be quantified by thecoherence bandwidthwhich is related to the specific multipath structure of th e channel. The coherence bandwidth is a measure of the maximum frequency difference for which signals are still strongly correlated in amplitude. If the transmitted signal has a narrow bandwidth as compared to the channel, the amplitude of the


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signal will change rapidly, but the signal will not be distorted in time. Thus, the statistics of small-scale signal strength and the likelihood of signal smearing appearing over small-scale distances are very much related to the specific amplitudes and delays of the multipath channel, as well as the bandwidth of the transmitted signal. 12. (a) (ii) Given: Transmitter power, Pt = 50 W Carrier frequency, fc = 900 MHz Using Equation (4.9), (a) Transmitter power, Pt (dBm) = 10log [Pt (mW)/ (1 mW)] = 10log [50 × 103] = 47.0 dBm. (b) Transmitter power, Pt (dBW) = 10log [Pt (mW)/(1 mW)] = 10log [50] = 17.0 dBW. The received power can be determined using Equation 2 =P

r

t P= Gt G r l ( 4p ) 2 d 2 L

Pr ( dBm)

=

10 log (

2

50(1)(1)(1 /3)2 ( 4p) (2 100 ) ( )1 =×

PmW) = r

(3.5 10

log( 10 . × 35 10 − =

6

−

)W

3 mW) .245 −

=

3.5 ×10 d

3

−

mW

Bm.

The received power at 10 km can be expressed in terms of dBm using Equation, where d0 = 100 m and d = 10 km

) Pr (10

km (= )Pr100 log 20 +

 100  . 5 dB40 − 10000  = −24dBm

= 64.5 dBm. 12. (b) (i) The Three Basic Propagation Mechanisms Reflection, diffraction, and scattering are the three basic propagation mechanisms which impact propagation in a mobile communication system. These mechanisms are briefly explained in this section, and propagation models which describe these mechanisms are discussed subsequently in this chapter. Received power (or its reciprocal, path loss) is generally the most important parameter predicted by large-scale propagation models based on the physics of reflection, scattering, and diffraction. Small-scale fading and multipath propagation (discussed in Chapter 5) may also be described by the physics of these three basic propagation mechanisms.


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Reflection occurs when a propagating electromagnetic wave impinges upon an object which has very large dimensions when compared to the wavelength of the propagating wave. Reflections occur from the surface of the earth and from buildings and walls. Diffraction occurs when the radio path between the transmitter and receiver is obstructed by a surface that has sharp irregularities (edges). The secondary waves resulting from the obstructing surface are present throughout the space and even behind the obstacle, giving rise to a bending of waves around the obstacle, even when a line-of-sight path does not exist between transmitter and receiver. At high frequencies, diffraction, like reflection, depends on the geometry of the object, as well as the amplitude, phase, and polarization of the incident wave at the point of diffraction. Scattering occurs when the medium through which the wave travels consists of objects with dimensions that are small compared to the wavelength, and where the number of obstacles per unit volume is large. Scattered waves are produced by rough surfaces, small objects, or by other irregularities in the channel. In practice, foliage, street signs, and lamp posts induce scattering in a mobile communications system. 12. (b) (ii) Brewster Angle The Brewster angle is the angle at which no reflection occurs in the medium of srcin. It occurs when the incident angle qB is such that the reflection coefficient Γ|| is equal to zero (see Figure). The Brewster angle is given by the value ofqB which satisfies sin(q B )

e1

=

e1

+e

(1) 2

For the case when the first medium is free space and the second medium has a relative permittivity,er Equation (2) can be expressed sin(q B )

=

er

−

1

−

1

2

er

(2)

Note that the Brewster angle occurs only for vertical (i.e. parallel) polarization. In accordance to Equation (2), the Brewster angle can be given by, sin(q B ) = 1/ 5or ,

For er = 15, Brewster angle,

qB

=

sin

−

1


qB

=

sin

1

−

() /1 5

( 14 /224 ) .=14 47

=

26.56°

°

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13. (a) (i) Nyquist Criterion for ISI Cancellation Nyquist was the first to solve the problem of overcoming intersymbol interference while keeping the transmission bandwidth low [Nyq28]. He observed that the effect of ISI could be completely nullified if the overall response of the communication system (including transmitter, channel, and receiver) is designed so that at every sampling instant at the receiver, the response due to all symbols except the current symbol is equal to zero. If heff (t) is the impulse response of the overall communication system, this condition can be mathematically stated as

Figure 1 Power spectral density of (a) unipolar NRZ, (b) bipolar RZ, and (c)

Manchester NRZ line codes.


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Figure 2 Time waveforms of binary line codes: (a) unipolar NRZ;

(b) bipolar RZ; (c) Manchester NRZ.

K

heff ( nTs ) = 

n=0

0 n ≠ 0

(1)

where Ts is the symbol period, n is an integer, andK is a non-zero constant. The effective transfer function of the system can be represented as

hefft ( )

=

dt( ) p *t (h ) *t h c( )t *r ( )

(2)

where p(t) is the pulse shape of a symbol, hc(t) is the channel impulse response, hr(t) and is the receiver impulse response. Nyquist derived transfer functions Heff (f) which satisfy the conditions of Equation (1) [Nyq28]. There are two important considerations in selecting a transfer function Heff( f ) which satisfy Equation (1). First, Heff( t ) should have a fast decay with a small magnitude near the sample values for n ≠ 0 . Second, if the channel is ideal (hc(t) = d (t)), then it should be possible to realize or closely approximate shaping filters at both the transmitter


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and receiver to produce the desired Heff( f ). Consider the impulse response in (3)

heff (t )

sin( =

p

t /)Ts

(3)

(p t ) /Ts

Clearly, this impulse response satisfies the Nyquist condition for ISI cancellation given in Equation (1) (see Figure 3). Therefore, if the overall communication system can be modeled as a filter with the impulse response of Equation (3), it is possible to completely eliminate the effects of ISI. The transfer function of the filter can be obtained by taking the Fourier transform of the impulse response, and is given by

heff ( f ) =

1

fs

 f  f s 

(4)

rect 

This transfer function corresponds to a rectangular “brick-wall” filter with absolute bandwidth fs/2, where fs is the symbol rate. While this transfer function satisfies the zero ISI criterion with a minimum of bandwidth, there are practical difficulties in implementing it, since it corresponds to a noncausal system (heff(t) exists for t < 0) and is thus difficult to approximate. Also, the (sin t)/t pulse has a waveform slope that is 1/t at each zero crossing, and is zero only at exact multiples of Ts, thus any error in the sampling time of zero-crossings will cause significant ISI due to overlapping from adjacent symbols. (A slope of 1/t2 or 1/t3 is more desirable to minimize the ISI due to timing jitter in adjacent samples.)

Figure 3 Nyquist ideal pulse shape for zero inter symbol interference

Nyquist also proved that any filter with a transfer function having a rectangular filter of bandwidth f0 ≥ 1/2Ts, convolved with any arbitrary even function Z( f ) with zero magnitude outside the passband of the


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2.19

rectangular filter, satisfies the zero ISI condition. Mathematically, the transfer function of the filter which satisfies the zero ISI condition can be expressed as

H)(

eff

f

 f = rect )( f 0  ⊗ Z

f

(5)

where Z( f ) = Z(−f ), and Z( f ) = 0 for f ≥ f 0 ≥ 1 / 2Ts . Expressed in terms of the impulse response, the Nyquist criterion states that any filter with an impulse response

heff (t )

sin(

p

t /)Ts

p

t

=

z (t )

(6)

can achieve ISI cancellation. Filters which satisfy the Nyquist criterion are called Nyquist filters (Figure 4). Assuming that the distortions introduced in the channel can be completely nullified by using an equalizer which has a transfer function that is equal to the inverse of the channel response, then the overall transfer function Heff(f ) can be approximated as the product of the transfer functions of the transmitter and receiver filters. An effective end-to-end transfer function of Heff(f ) is often achieved by using filters with transfer functions H eff ( f ) at both the transmitter and receiver. This has the advantage of providing a matched filter response for the system, while at the same time minimizing the bandwidth and inter symbol interference.

Figure 4 Transfer function of a Nyquist pulse-shaping filter at baseband.

13. (a) (ii) Raised Cosine Roll off Filter The most popular pulse shaping filter used in mobile communications is the raised cosine filter. A raised cosine filter belongs to the class of filters


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which satisfy the Nyquist criterion. The transfer function of a raised cosine filter is given by

1 

0 ≤| f |≤

 1 1 + cos  p (| f | ⋅2Ts − 1 + a )       2a 2  0 

H RC (f ) 

(1 − a )

(1 − a ) 2Ts

 

(1 a ) ≤| f |≤ +  2Ts 2Ts   (1 + a ) | f |>  2Ts 

(2)

where a is the rolloff factor which ranges between 0 and 1. This transfer function is plotted in Figure 1 for various values of a. When a = 0, the raised cosine roll off filter corresponds to a rectangular filter of minimum bandwidth. The corresponding impulse response of the filter can be obtained by taking the inverse Fourier transform of the transfer function, and is given by sin

pt

cos

 T  ⋅ s

hRC (t )

p

t

pat

 T 2 4a t   1−   2T  s

(11)

s

The impulse response of the cosine rolloff filter at baseband is plotted in Figure 2 for various values of a. Notice that the impulse response decays much faster at the zero-crossings (approximately as 1/ t3 for t » Ts) when compared to the “brick-wall” filtera = 0). The rapid time rolloff allows it to be truncated in time with little deviation in performance from theory. As seen from Figure 1, as the roll off factor a increases, the bandwidth of the filter also increases, and the time sidelobe levels decrease in adjacent symbol slots. This implies that increasinga decreases the sensitivity to timing jitter, but increases the occupied bandwidth. s The symbol that by can be passed through a baseband raised cosine rolloffrate filter isRgiven

Rs

=

1

Ts

=

2B 1+ a

(3)

where B is the absolute filter bandwidth. For RF systems, the RF passband bandwidth doubles and Rs


B =

1+ a

(4)

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Figure 1 Magnitude transfer function of a raised cosine filter at baseband.

Figure 2 Impulse response of a raised cosine rolloff filter at baseband.

13. (b) (i) QPSK Transmission and Detection Techniques Figure 2 shows a block diagram of a typical QPSK transmitter. The unipolar binary message stream has bit rate Rb and is first converted into a bipolar non-return-to-zero (NRZ) sequence using a unipolar to bipolar converter. The bit stream m(t) is then split into two bit streams mI(t) and mQ(t) (in-phase and quadrature streams), each having a bit rate of Rs = Rb/2. The bit stream mI(t) is called the “even” stream andmQ(t) is called the “odd” stream. The two binary sequences are separately modulated by two carriers f1(t) and f2(t), which are in quadrature.


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The two modulated signals, each of which can be considered to be a BPSK signal, are summed to produce a QPSK signal. The filter at the output of the modulator confines the power spectrum of the QPSK signal within the allocated band. This prevents spill-over of signal energy into adjacent channels and also removes out-of-band spurious signals generated during the modulation process. In most implementations, pulse shaping is done at baseband to provide proper RF filtering at the transmitter output.

Figure 1 Block diagram of a QPSK transmitter.

Figure 2 Block diagram of a QPSK receiver

Figure 2 shows a block diagram of a coherent QPSK receiver. The frontend bandpass filter removes the out-of-band noise and adjacent channel interference. The filtered output is split into two parts, and each part is coherently demodulated using the in-phase and quadrature


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2.23

carriers. The coherent carriers used for demodulation are recovered from the received signal using carrier recovery circuits of the type described in Figure. The outputs of the demodulators are passed through decision circuits which generate the in-phase and quadrature binary streams. The two components are then multiplexed to reproduce the srcinal binary sequence. 13. (b) (ii) Performance of Digital Modulation in Slow Flat-Fading Channels

As discussed in Chapter, flat-fading channels cause a multiplicative (gain) variation in the transmitted signal s(t). Since slow flat-fading channels change much slower than the applied modulation, it can be assumed that the attenuation and phase shift of the signal is constant over at least one symbol interval. Therefore, the received signalr(t) may be expressed as rt ( )

=ta( ) exp( −j t st

n (q t )) +()

(t≤)≤T

0

(1)

where a (t) is the gain of the channel, q (t) is the phase shift of the channel, and n(t) is additive Gaussian noise. Depending on whether it is possible to make an accurate estimate of the phase q (t), coherent or noncoherent matched filter detection may be employed at the receiver. To evaluate the probability of error of any digital modulation scheme in a slow flat-fading channel, one must average the probability of error of the particular modulation in AWGN channels over the possible ranges of signal strength due to fading. In other words, the probability of error in AWGN channels is viewed as a conditional error probability, where the condition is that a is fixed. Hence, the probability of error in slow flat-fading channels can be obtained by averaging the error in AWGN channels over the fading probability density function. In doing so, the probability of error in a slow flat-fading channel can be evaluated as ∞

∫

Pe = Pe( X) ( p) X dX

(2)

0

where Pe (X) is the probability of error for an arbitrary modulation at a specific value of signal-to-noise ratioX, X = a 2Eb/N0, and p(X) is the probability density function of X due to the fading channel. Eb and N0 are constants that represent the average energy per bit and noise power density in a non-fading AWGN channel, and the random variablea 2


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2.24


is used to represent instantaneous power values of the fading channel, with respect to the non-fading Eb/N0. It is convenient to assume α 2 is one, for a unity gain fading channel. Then, p(X) can simply be viewed as the distribution of the instantaneous value of Eb/N0 in a fading channel, and P (X) can be seen to be the conditional probability of bit errors for a givene value of the randomEb/N0 due to fading. For Rayleigh fading channels, the fading amplitude a has a Rayleigh distribution, so the fading power a 2 and consequently X have a chisquare distribution with two degrees of freedom. Therefore, p ( X)

where 2

Γ =

Eb

a

2

N0

=

1 exp

Γ

− X    Γ

X

≥0

(3)

is the average value of the signal-to-noise ratio. For

1, note that Γ corresponds to the average Eb/N for the fading 0 channel. By using Equation (3) and the probability of error of a particular modulation scheme in AWGN, the probability of error in a slow flatfading channel can be evaluated. It can be shown that for coherent binary PSK and coherent binary FSK, Equation (4) evaluates to [Ste87]

a

=

Pe , PSK =

1 Γ  1 −  (coherentbinaryPSK) 2 1+ Γ 

(4)

Pe , FSK =

1 1 − 2

(5)

Γ   (coherentbinaryFSK) 2+Γ 

It can also be shown that the average error probability of DPSK and orthogonal noncoherent FSK in a slow, flat, Rayleigh fading channel are given by Pe , DPSK =

Pe , NCFSK =

1 (differentialbinaryPSK) 2(1 + Γ) 1

2+Γ

(noncoherent orthogonal binary FSK)

(6) (7)

Figure 3 illustrates how the BER for various modulations changes as a function of Eb/N0 in a Rayleigh flat-fading environment. The figure was produced using simulation instead of analysis, but agrees closely with Equations (6) to (9) [Rap91b].


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Figure 3 Bit error rate performance of binary modulation schemes in a

Rayleigh flat-fading channel as compared to a typical performance curve in AWGN. For large values of Eb/N0 (i.e., large values of X), the error probability Equations may be simplified as Pe , PSK = Pe , FSK =

Pe , DPSK = Pe , NCFSK =

1 4Γ 1 2Γ

1 2Γ

1 Γ

(coherent binary PSK)

(8)

(coherent binary FSK)

(9)

(differential PSK)

(10)

(noncoherent orthogonal binary FSK)

(11)

For GMSK, the expression for BER in the AWGN channel is given in Equation (6.112.a) which when evaluated in Equation (2) yields a Rayleigh fading BER of Pe ,GMSK

1 = 1 − 2

1 Γ  ≅ Γ + 1  4d Γ

d d

(coherent GMSK)

(12)

where d

. 068 ≅ 0  .85


for

BT 025 = .

for

BT

=∞

(13)

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2.26


As seen from Equations (22) to (26), for lower error rates all five modulation techniques exhibit an inverse algebraic relation between error rate and mean SNR. This is in contrast with the exponential relationship between error rate and SNR in an AWGN channel. According to these results, it is seen that operating at BERs of 10−3 to 10−6 requires roughly a 30 dB to 60 dB mean SNR. This is significantly larger than that required when operating over a nonfading Gaussian noise channel (20 dB to 50 dB more link is required). However, it can easily be shown that the poor error performance is due to the non-zero probability of very deep fades, when the instantaneous BER can become as low as 0.5. Significant improvement in BER can be achieved by using efficient techniques such as diversity or error control coding to totally avoid the probability of deep fades, as shown in Chapter 7. Work by Yao [Yao92] demonstrates how the analytical technique of Equation (2) may be applied to desired signals as well as interfering signals which undergo Rayleigh, Ricean, or log-normal fading. 14. (a) (i) Linear Equalizers As mentioned in otherwise Section 7.5, a linear equalizer canfilter be .implemented as an FIR filter, known as the transversal This type of equalizer is the simplest type available. In such an equalizer, the current and past values of the received signal are linearly weighted by the filter coefficient and summed to produce the output, as shown in Figure 1. If the delays and the tap gains are analog, the continuous output of the equalizer is sampled at the symbol rate and the samples are applied to the decision device. The implementation is, however, usually carried out in the digital domain where the samples of the received signal are stored in a shift register. The output of this transversal filter before a decision is made (threshold detection) is [Kor85]

dˆ k =

N2

∑ n(cnk *) y

(1)

−

n = − N1 *

where c represents the complex filter coefficients or tap weights,d k is the output at time index k, yi is the input received signal at timet0 + iT, t0 is the equalizer starting time, andN = N1 + N2 is the number of taps. The values N1 and N2 denote the number of taps used in the forward and reverse portions of the equalizer, respectively. The minimum mean squared error E[|e(n)|2] that a linear transversal equalizer can achieve is [Pro89] ˆ

n


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2.27

Figure 1 Structure of a linear transversal equalizer.

E [| e( n) | ]2 =

p

T

2π −

/T

N0

∫ p

| F( e j )|T /T w

2

+N 0

(2)

dw

where F(ejwT) is the frequency response of the channel, and N0 is the noise power spectral density. The linear equalizer can also be implemented as lattice a filter, whose structure is shown in Figure 2. In a lattice filter, the input signal yk is transformed into a set ofN intermediate forward and backward error signals, fn(k) and bn(k) respectively, which are used as inputs to the tap multipliers and are used to calculate the updated coefficients. Each stage of the lattice is then characterized by thefollowing recursive equations [Bin88]:

f 1 ( k)

=

(b)1 k ( ) y k

(3)

=

n

f n ()k( =) y−k

(∑− ) yn +k() n in ()f −(1 −k ) =K

i

K

−1

k b

−1

k 1

(4)

i =1 n

bn)k( ( =yk) n− − ( Ky∑ k n−) +i i

(5)

i =1

=

bn 1k( −

)1K+k f(n)k(1 )

−

−

n −1

where Kn(k) is the reflection coefficient for the nth stage of the lattice. The backward error signals, bn, are then used as inputs to the tap weights, and the output of the equalizer is given by N

dˆ k =

∑c kb( k) n

(n )

(6)

n =1


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Figure 2 The structure of a lattice equalizer [from [Pro91] © IEEE].

Two main advantages of the lattice equalizer is its numerical stability and faster convergence. Also, the unique structure of the lattice filter allows the dynamic assignment of the most effective length of the lattice equalizer. Hence, if the channel is not very time dispersive, only a fraction of the stages are used. When the channel becomes more time dispersive, the length of the equalizer can be increased by the algorithm without stopping the operation of the equalizer. The structure of a lattice equalizer, however, is more complicated than a linear transversal equalizer. (ii) Nonlinear Equalization Nonlinear equalizers are used in applications where the channel distortion is too severe for a linear equalizer to handle, and are commonplace in practical wireless systems. Linear equalizers do not perform well on channels which have deep spectral nulls in the pass band. In an attempt to compensate for the distortion, the linear equalizer places too much gain in the vicinity of the spectral null, thereby enhancing the noise present in those frequencies. Three very effective nonlinear methods have been developed which offer improvements over linear equalization techniques and are used in most 2G and 3G systems. These are [Pro91]: 1. Decision Feedback Equalization (DFE) 2. Maximum Likelihood Symbol Detection 3. Maximum Likelihood Sequence Estimation (MLSE)


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Figure 1 Decision feedback equalizer (DFE). Decision Feedback Equalization (DFE)

The basic idea behind decision feedback equalization is that once an information symbol has been detected and decided upon, the ISI that it induces on future symbols can be estimated and subtracted out before detection of subsequent symbols [Pro89]. The DFE can be realized in either the direct transversal form or as a lattice filter. The direct form is shown in Figure1. It consists of a feed forward filter (FFF) and a feedback filter (FBF). The FBF is driven by decisions on the output of the detector, and its coefficients can be adjusted to cancel the ISI on the current symbol from past detected symbols. The equalizer has N1 + N2 + 1 taps in the feed forward filter and N3 taps in the feedback filter, and its output can be expressed as: N3

N2

dˆk =

∑cy nkn n = − N1

*

F+ d∑ −

i

(1)

k −i

i =1

*

where c and yn are tap gains and the inputs, respectively, to the forward n *

filter, Fi are tap gains forthe feedback filter, and di(i < k) is the previous decision made on the detected signal. That is, once dk is obtained using Equation 1, dk is decided from it. Then, dk along with previous decisions dk−1, dk−2,… are fed back into the equalizer, and dˆk 1 is obtained using Equation (1). ˆ

+


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2.30


The minimum mean squared error a DFE can achieve is [Pro89]

E[| e( )|n]

2

 T π /T  ln ∫− π / T  2π

exp =

min

N0     | F( e jω)|T 2 + N  d ω  0   

(2)

It can be shown that the minimum MSE for a DFE in Equation (2) is always smaller than that of an LTE in Equation unless| F( e j )T| 2 is a constant (i.e., when adaptive equalization is not needed) [Pro89]. If there are nulls in | F( e j )T| 2, a DFE has significantly smaller minimum MSE than an LTE. Therefore, an LTE is well behaved when the channel spectrum is comparatively flat, but if the channel is severely distorted or exhibits nulls in the spectrum, the performance of an LTE deteriorates and the mean squared error of a DFE is much better than a LTE. Also, an LTE has difficulty equalizing a nonminimum phase channel, where the strongest energy arrives after the first arriving signal component. Thus, a DFE is more appropriate for severely distorted wireless channels. The lattice implementation of the DFE is equivalent to a transversal DFE having a feed forward filter of length N1 and a feedback filter of w

w

length N2, where N1 > N2. Another form of DFE proposed by Belfiore and Park [Bel79] is called a predictive DFE, and is shown in Figure 3. It also consists of a feed forward filter (FFF) as in the conventional DFE. However, the feedback filter (FBF) is driven by an input sequence formed by the difference of the output of the detector and the output of the feed forward filter. Hence, the FBF here is called anoise predictor because it predicts the noise and

Figure 3 Predictive decision feedback equalizer.


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the residual ISI contained in the signal at the FFF output and subtracts from it the detector output after some feedback delay. The predictive DFE performs as well as the conventional DFE as the limit in the number of taps in the FFF and the FBF approach infinity. The FBF in the predictive DFE can also be realized as a lattice structure [Zho90]. The RLS lattice algorithm (discussed in Section 7.8) can be used in this case to yield fast convergence. Maximum Likelihood Sequence Estimation (MLSE) Equalizer

The MSE-based linear equalizers described previously are optimum with respect to the criterion of minimum probability of symbol error when the channel does not introduce any amplitude distortion. Yet this is precisely the condition in which an equalizer is needed for a mobile communications link. This limitation on MSE-based equalizers led researchers to investigate optimum or nearly optimum nonlinear structures. These equalizers use various forms of the classical maximum likelihood receiver structure. Using a channel impulse response simulator within the algorithm, the MLSE tests all possible data sequences (rather than decoding each received symbol by itself), and chooses the data sequence with the maximum probability as the output. An MLSE usually has a large computational requirement, especially when the delay spread of the channel is large. Using the MLSE as an equalizer was first proposed by Forney [For78] in which he set up a basic MLSE estimator structure and implemented it with the Viterbi algorithm. This algorithm, described in Section 7.15, was recognized to be a maximum likelihood sequence estimator (MLSE) of the state sequences of a finite state Markov process observed in memoryless noise. It has recently been implemented successfully for equalizers in mobile radio channels. The MLSE can be viewed as a problem in estimating the state of a discrete-time finite state machine, which in this case happens to be the radio channel with coefficients fk, and with a channel state which at any instant of time is estimated by the receiver based on theL most recent input samples. Thus, the channel has ML states, where M is the size of the symbol alphabet of the modulation. That is, an ML trellis is used by the receiver to model the channel overtime. The Viterbi algorithm then tracks the state of the channel by the paths through the trellis and gives at stage k a rank ordering of the ML most probable sequences terminating in the most recent L symbols. The block diagram of a MLSE receiver based on the DFE is shown in Figure 4. The MLSE is optimal in the sense that it minimizes the probability of a sequence error. The MLSE requires knowledge of the


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2.32


channel characteristics in order to compute the metrics for making decisions. The MLSE also requires knowledge of the statistical distribution of the noise corrupting the signal. Thus, the probability distribution of the noise determines the form of the metric for optimum demodulation of the received signal. Notice that the matched filter operates on the continuous time signal, whereas the MLSE and channel estimator rely on discretized (nonlinear) samples.

Figure 4 The structure of a maximum likelihood sequence estimator (MLSE)

with an adaptive matched filter. 14. (b) (i) RAKE Receiver In CDMA spread spectrum systems (see Chapter), the chip rate is typically much greater than the flat-fading bandwidth of the channel. Whereas conventional modulation techniques require an equalizer to undo the intersymbol interference between adjacent symbols, CDMA spreading codes are designed to provide very low correlation between successive chips. Thus, propagation delay spread in the radio channel merely provides multiple versions of the transmitted signal at the receiver. If these multipath components are delayed in time by more than a chip duration, they appear like uncorrelated noise at a CDMA receiver, and equalization is not required. The spread spectrum processing gain makes uncorrelated noise negligible after despreading. However, since there is useful information in the multipath components, CDMA receivers may combine the time delayed versions of the srcinal signal transmission in order to improve the signal-to-noise ratio at the receiver. A RAKE receiver does just this—it attempts to collect the timeshifted versions of the srcinal signal by providing a separate correlation receiver for each of the multipath signals. Each correlation receiver may be adjusted in time delay, so that a microprocessor controller can cause


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2.33

different correlation receivers to search in different time windows for significant multipath. The range of time delays that a particular correlator can search is called a search window. The RAKE receiver, shown in Figure 5, is essentially a diversity receiver designed specifically for CDMA, where the diversity is provided by the fact that the multipath components are practically uncorrelated from one another when their relative propagation delays exceed a chip period. A RAKE receiver utilizes multiple correlators to separately detect the M strongest multipath components. The outputs of each correlator are then weighted to provide a better estimate of the transmitted signal than is provided by a single component. Demodulation and bit decisions are then based on the weighted outputs of the M correlators. The basic idea of a RAKE receiver was first proposed by Price and Green [Pri58]. In outdoor environments, the delay between multipath components is usually large and, if the chip rate is properly selected, the low autocorrelation properties of a CDMA spreading sequence can assure that multipath components will appear nearly uncorrelated with each other. However, the RAKE receiver in IS-95 CDMA has been found to perform poorly in indoor environments, which is to be expected since the multipath delay spreads in indoor channels ≈( 100 ns) are much smaller than an IS-95 chip duration (≈800 ns). In such cases, a RAKE will not work since multipath is unresolveable, and Rayleigh flat-fading typically occurs within a single chip period. To explore the performance of a RAKE receiver, assume M correlators are used in a CDMA receiver to capture theM strongest multipath components. A weighting network is used to provide a linear combination

Figure 5 An M-branch (M-finger) RAKE receiver implementation.

Each correlator detects a time shifted version of the srcinal CDMA transmission, and each finger of the RAKE correlates to a portion of the signal which is delayed by at least one chip in time from the other fingers.


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of the correlator output for bit detection. Correlator 1 is synchronized to the strongest multipathm2. Multipath componentm2 arrives t1 later than component m1 where t2 – t1 is assumed to be greater than a chip duration. The second correlator is synchronized tom2. It correlates strongly with m2, but has low correlation withm1. Note that if only a single correlator is used in the receiver (see Figure), once the output of the single correlator is corrupted by fading, the receiver cannot correct the value. Bit decisions based on only a single correlation may produce a large bit error rate. In a RAKE receiver, if the output from one correlato r is corrupted by fading, the others may not be, and the corrupted signal may be discounted through the weighting process. Decisions based on the combination of the M separate decision statistics offered by the RAKE provide a form of diversity which can overcome fading and thereby improve CDMA reception. The M decision statistics are weighted to form an overall decision statistic as shown in Figure 5. The outputs of theM correlators are denoted as Z1, Z2... and ZM. They are weighted by a1, a2, … and aM, respectively. The weighting coefficients are based on the power or the SNR from each correlator output. If the power or SNR is small out of a particular correlator, it will be assigned a small weighting factor. Just as in the case of a maximal ratio combining diversity scheme, the overall signal Z′ is given by M

Z

′=

∑a

m

Zm

(1)

m =1

The weighting coefficients, am, are normalized to the output signal power of the correlator in such a way that the coefficients sum to unity, as shown in Equation (2) am

=

Z

2 m

(2)

M

∑Z

2 m

m =1

As in the case of adaptive equalizers and diversity combining, there are many ways to generate the weighting coefficients. However, due to multiple access interference, RAKE fingers with strong multipath amplitudes will not necessarily provide strong output after correlation. Choosing weighting coefficients based on the actual outputs of the correlators yields better RAKE performance. 14. (b) (ii) Frequency Domain Coding of Speech Frequency domain coders [Tri79] are a class of speech coders which take advantage of speech perception and generation models without


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2.35

making the algorithm totally dependent on the models used. In this class of coders, the speech signal is divided into a set of frequency components which are quantized and encoded separately. In this way, different frequency bands can be preferentially encoded according to some perceptual criteria for each band, and hence the quantization noise can be contained within bands and prevented from creating harmonic distortions outside the band. These schemes have the advantage that the number of bits used to encode each frequency component can be dynamically varied and shared among the different bands. Many frequency domain coding algorithms, ranging from simple to complex are available. The most common types of frequency domain coding include sub-band coding (SBC) and block transform coding. While a sub-band coder divides the speech signal into many smaller sub-bands and encodes each sub-band separately according to some perceptual criterion, a transform coder codes the short-time transform of a windowed sequence of samples and encodes them with number of bits proportional to its perceptual significance. Sub-band Coding

Sub-band coding can be thought of as a method of controlling and distributing quantization noise across the signal spectrum. Quantization is a nonlinear operation which produces distortion products that are typically broad in spectrum. The human ear does not detect the quantization distortion at all frequencies equally well. It is therefore possible to achieve substantial improvement in quality by coding the signal in narrower bands. In a sub-band coder, speech is typically divided into four or eight sub-bands by a bank of filters, and each sub-band is sampled at a bandpass Nyquist rate (which is lower than the srcinal sampling rate) and encoded with different accuracy in accordance to a perceptual criteria. Band-splitting can be done in many ways. One approach could be to divide the entire speech band into unequal sub-bands that contribute equally to the articulation index. One partitioning of the speech band according to this method as suggested by Crochiere et al. [Cro76] is given below. Sub-band Number

Frequency Range

1

200–700Hz

2

700–1310Hz

3

1310–2020Hz

4

2020–3200Hz


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Another way to split the speech band would be to divide it into equal width sub-bands and assign to each sub-band number of bits proportional to perceptual significance while encoding them. Instead of partitioning into equal width bands, octave band splitting is often employed. As the human ear has an exponentially decreasing sensitivity to frequency, this kind of splitting is more in tune with the perception process. There are various methods for processing the sub-band signals. One obvious way is to make a low pass translation of the sub-band signal to zero frequency by a modulation process equivalent to single sideband modulation. This kind of translation facilitates sampling rate reduction and possesses other benefits that accrue from coding low-pass signals. Figure 6 shows a simple means of achieving this low pass translation. The input signal is filtered with a bandpass filter of width wn for the n th band. w1n is the lower edge of the band and w2n is the upper edge of the band. The resulting signal sn(t) is modulated by a cosine wave cos (w1n t), and filtered using a low pass filter hn (t) with bandwidth (0 − wn). The resulting signal rn (t) corresponds to the low pass translated version sn(t) of and can be expressed as

rn(t) = [sn(t) cos w1nt)] ⊗ hn(t)

(1)

where ⊗ denotes a convolution operation. The signalrn(t) is sampled at a rate 2wn. This signal is then digitally encoded and multiplexed with encoded signals from other channels as shown in Figure 6. At the receiver, the data is demultiplexed into separate channels, decoded, and bandpass translated to give the estimate of rn(t) for the n th channel. The low pass translation technique is straightforward and takes advantage of a bank of nonoverlapping bandpass filters. Unfortunately, unless we use sophisticated bandpass filters, this approach will lead to perceptible aliasing effects. Estaban and Galand proposed [Est77] a scheme which avoids this inconvenience even with quasiperfect, subband splitting. Filter banks known as quadrature mirror filters (QMFs) are used to achieve this. By designing a set of mirror filters which satisfy certain symmetry conditions, it is possible to obtain perfect alias cancellation. This facilitates the implementation of sub-band coding without the use of very high order filters. This is particularly attractive for real time implementation as a reduced filter order means a reduced computational load and also a reduced latency. Sub-band coding can be used for coding speech at bit rates in the range 9.6 kbps to 32 kbps. In this range, speech quality is roughly equivalent to that of ADPCM at an equivalent bit rate. In addition, its complexity and relative speech quality at low bit rates make it particularly advantageous


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2.37

for coding below about 16 kbps. However, the increased complexity of subband coding when compared to other higher bit rate techniques does not warrant its use at bit rates greater than about 20 kbps. The CD-900 cellular telephone system uses sub-band coding for speech compression.

Figure 6 Block diagram of a sub-band coder and decoder.

15. (a) (i) Code Division Multiple Access (CDMA) In code division multiple access (CDMA) systems, the narrowband message signal is multiplied by a very large bandwidth signal called the spreading signal. The spreading signal is a pseudonoise code sequence that has a chip rate which is orders of magnitudes greater than the data rate of the message. All users in a CDMA system, as seen from figure 7, use the same carrier frequency and may transmit simultaneously. Each user has its own pseudorandom codeword which is approximately orthogonal to all other codewords.The receiver performs a time correlation operation to detect only the specific desired codeword. All other codewords appear as noise due to decorrelation. For detection of the message signal, the


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receiver needs to know the codeword used by the transmitter. Each user operates independently with no knowledge of the other users. In CDMA, the power of multiple users at a receiver determines the noise floor after decorrelation. If the power of each user within a cell is not controlled such that they do not appear equal at the base station receiver, then the near-far problem occurs. The near–far problem occurs when many mobile users share the same channel. In general, the strongest received mobile signal willcapture the demodulator at a base station. In CDMA, stronger received signal levels raise the noise floor at the base station demodulators for the weaker signals, thereby decreasing the probability that weaker signals will be received. To combat the near–far problem, power control is used in most CDMA implementations. Power control is provided by each base station in a cellular system and assures that each mobile within the base station coverage area provides the same signal level to the base station receiver. This solves the problem of a nearby subscriber overpowering the base station receiver and drowning out the signals of far away subscribers. Power control is implemented at the base station by rapidly sampling the radio signal strength indicator (RSSI) levels of each mobile and then sending a power change command over the forward radio link. Despite the use of power control within each cell, out-of-cell mobiles provide interference which is not under the control of the receiving base station. The features of CDMA including the following: •

Many users of a CDMA system share the same frequency. Either TDD or FDD may be used. • Unlike TDMA or FDMA, CDMA has a soft capacity limit. Increasing the number of users in a CDMA system raises the noise floor in a linear manner. Thus, there is no absolute limit on the number of users in CDMA. Rather, the system performance gradually degrades for all users as the number of users is increased, and improves as the number of users is decreased. Multipath fading may be substantially reduced because the signal is spread over a large spectrum. If the spread spectrum bandwidth is greater than the coherence bandwidth of the channel, the inherent frequency diversity will mitigate the effects of small-scale fading. • Channel data rates are very high in CDMA systems. Consequently, the symbol (chip) duration is very short and usually much less than the channel delay spread. Since PN sequences have low autocorrelation, multipath which is delayed by more than a chip will appear as noise. A RAKE receiver can be used to improve reception by collecting time delayed versions of the required signal. •


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2.39

•

Since CDMA uses co-channel cells, it can use macroscopic spatial diversity to provide soft handoff. Soft handoff is performed by the MSC, which can simultaneously monitor a particular user from two or more base stations. The MSC may chose the best version of the signal at any time without switching frequencies. • Self-jamming is a problem in CDMA system. Self-jamming arises from the fact that the spreading sequences of different users are not exactly orthogonal, hence in the despreading of a particular PN code, non-zero contributions to the receiver decision statistic for a desired user arise from the transmissions of other users in the system. • The near–far problem occurs at a CDMA receiver if an undesired user has a high detected power as compared to the desired user. 15. (a) (ii) Frequency Hopped Multiple Access (FHMA)

Frequency hopped multiple access (FHMA) is a digital multiple access system in which the carrier frequencies of the individual users are varied in a pseudorandom fashion within a wideband channel. Figure 7 illustrates how FHMA allows multiple users to simultaneously occupy the same spectrum at the time, where eachofuser dwells at aon specific narrowband channel at asame particular instance time, based the

Figure 7 Spread spectrum multiple access in

which each channel is assigned a unique PN code which is orthogonal or approximately orthogonal to PN codes used by other users.


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2.40


particular PN code of the user. The digital data of each user is broken into uniform sized bursts which are transmitted on different channels within the allocated spectrum band. The instantaneous bandwidth of any one transmission burst is much smaller than the total spread bandwidth. The pseudorandom change of the channel frequencies of the user randomizes the occupancy of a specific channel at any given time, thereby allowing for multiple access over a wide range of frequencies. In the FH receiver, a locally generated PN code is used to synchronize the receiver’s instantaneous frequency with that of the transmitter. At any given point in time, a frequency hopped signal only occupies a single, relatively narrow channel since narrowband FM or FSK is used. The difference between FHMA and a traditional FDMA system is that the frequency hopped signal changes channels at rapid intervals. If the rate of change of the carrier frequency is greater than the symbol rate, then the system is referred to as a fast frequency hopping system. If the channel changes at a rate less than or equal to the symbol rate, it is called slow frequency hopping. A fast frequency hopper may thus be thought of as an FDMA system which employs frequency diversity. FHMA systems often employ energy efficient constant envelope modulation. Inexpensive receivers may be built to provide noncoherent detection of FHMA. This implies that linearity is not an issue, and the power of multiple users at the receiver does not degrade FHMA performance. A frequency hopped system provides a level of security, especially when a large number of channels are used, since an unintended (or an intercepting) receiver that does not know the pseudorandom sequence of frequency slots must retune rapidly to search for the signal it wishes to intercept. In addition, the FH signal is somewhat immune to fading, since error control coding and interleaving can be used to protect the frequency hopped signal against deep fades which may occasionally occur during the hopping sequence. Error control coding and interleaving can also be combined to guard against erasures which can occur when two or more users transmit on the same channel at the same time. Bluetooth and HomeRF wireless technologies have adopted FHMA for power efficiency and low cost implementation. 15. (b) 2G network: Second Generation (2G) Cellular N etworks

Most of today’s ubiquitous cellular networks use what is commonly called second generation or 2G technologies which conform to the


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2.41

second generation cellular standards. Unlike first generation cellular systems that relied exclusively on FDMA/FDD and analog FM, second generation standards use digital modulation formats and TDMA/FDD and CDMA/FDD multiple access techniques. The most popular second generation standards include three TDMA standards and one CDMA standard: (a) Global System Mobile ( GSM), which supports eight time slotted users for each 200 kHz radio channel and has been deployed widely by service providers in Europe, Asia, Australia, South America, and some parts of the US (in the PCS spectrum band only) [Gar99]; (b) Interim Standard 136 (IS-136), also known as North American Digital Cellular (NADC), which supports three time slotted users for each 30 kHz radio channel and is a popular choice for carriers in North America, South America, and Australia (in both the cellular and PCS bands); (c) Pacific Digital Cellular (PDC), a Japanese TDMA standard that is similar to IS-136 with more than 50million users; and (d) the popular 2G CDMA standard Interim Standard 95 Code Division Multiple Access (IS-95), also known as cdmaOne, which supports up to 64 users that are orthogonally coded and simultaneously transmitted on each 1.25 MHz channel. CDMA is widely deployed by carriers in North America (in both cellular and PCS bands), as well as in Korea, Japan, China, South America, and Australia [Lib99, ch. 1], [Kim00], [Gar00]. The 2G standards mentioned above represent the first set of wireless air interface standards to rely on digital modulation and sophisticated digital signal processing in the handset and the base station. As discussed in Chapter 11, second generation systems were first introduced in the early1990s, and evolved from the first generation of analog mobile phone systems (e.g., AMPS, ETACS, and JTACS). Today, many wireless service providers use both first generation and second generation equipment in major markets and often provide customers with subscriber units that can support multiple frequency bands and multiple air interface standards. For example, in many countries it is possible to purchase a single trimode cellular handset phone that supports CDMA in the cellular and PCS bands in addition to analog first generation technology in the cellular band. Such tri-mode phones are able to automatically sense and adapt to whichever standard is being used in a particular market. Figure 1 illustrates how the world subscriber base was divided between the major 1G and2G technologies as of late 2001. Table 1 highlights the key technical specifications of the dominant GSM, CDMA, and IS-136/PDC second generation standards.


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2.42


Figure 1 Worldwide subscriber base as a function of cellular technology in

late 2001. In many countries, 2G wireless networks are designed and deployed for conventional mobile telephone service, as a high capacity replacement for, or in competition with, existing older first generation cellular telephone systems. Modern cellular systems are also being installed to provide fixed (non-mobile) telephone service to residences and businesses in developing nations—this is particularly cost effective for providing plain old telephone service (POTS) in countries that have poor telecommunications infrastructure and are unable to afford the installation of copper wire to all homes. Since all 2G technologies offer at least a three-times increase in spectrum efficiency(and thus at least a 3X increase in overall system capacity) as compared to first generation analog technologies, the need to meet a rapidly growing customer base justifies the gradual, ongoing change out of analog to digital 2G technologies in any growing wireless network. In mid-2001, several major carriers such as AT&T Wireless and Cingular in the US and NTT in Japan announced their decisions to eventually abandon the IS-136 and PDC standards as long term technology options in favor of emerging third generation standards based on the GSM TDMA platform. Simultaneously, international wireless carrier Nextel announced its decision to upgrade its iDen air interface standard to support up to five times the number of current users based on a data compression methodology using Internet protocol (IP) packet data. Most other carriers throughout the world had already committed to adopting


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a 3G standard based on either GSM or CDMA prior to 2001. Decisions like these have set the stage for the inevitability of two universal and competing third generation (3G) cellular mobile radio technologies, one based on the philosophy and backward compatibility of GSM, and the other based on the philosophy and backward compatibility of CDMA. 3G network Third Generation (3G) Wireless Networks

3G systems promise unparalleled wireless access in ways that have never been possible before. Multi-megabit Internet access, communications using Voice over Internet Protocol (VoIP), voice-activated calls, unparalleled network capacity, and ubiquitous “always-on” access are just some of the advantages being touted by 3G developers. Companies developing 3G equipment envision users having the ability to receive live music, conduct interactive web sessions, and have simultaneous voice and data access with multiple parties at the same time using a single mobile handset, whether driving, walking, or standing still in an office setting. As mentioned in Chapter 1, and described in detail in [Lib99], the International Telecommunications Union (ITU) formulated a plan to implement a global frequency band in the 2000MHz range that would support a single, ubiquitous wireless communication standard for all countries throughout the world. This plan, called International Mobile Telephone 2000 (IMT-2000), has been successful in helping to cultivate active debate and technical analysis for new high speed mobile telephone solutions when compared to 2G. However, as can be seen in Figures, the hope for a single worldwide standard has not materialized, as the worldwide user community remains split between two camps: GSM/IS-136/ PDC and CDMA. The eventual 3G evolution for CDMA systems leads to cdma2000. Several variants of CDMA 2000 are currently being developed, but they all are based on the fundamentals of IS-95and IS-95B technologies. The eventual 3G evolution for GSM, IS-136, and PDC systems leads to Wideband CDMA (W-CDMA), also called Universal Mobile Telecommunications Service(UMTS). W-CDMA is based on the network fundamentals of GSM, as well as the merged versions of GSM and IS136 through EDGE. It is fair to say that these two major 3G technology camps, cdma2000 and W-CDMA, will remain popular throughout the early part of the 21st century. Table 2 illustrates the primary worldwide proposals that were submitted for IMT-2000 in 1998. Since 1998, many standards proposals


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2.47

have capitulated and joined with either thecdma2000 or UMTS (W-CDMA) camps. Commercial grade 3G equipment is expected to be available in the 2002–2003 time frame. Two good Internet sources for up-to-the-minute 3G developments can be found at GSM World (www.gsmworld.com) and the CDMA Developers Group (www.cdg. org). The ITU IMT-2000 standards organizations are currently separated into two major organizations reflecting the two 3G camps: 3GPP (3G Partnership Project for Wideband CDMA standards based on backward compatibility with GSM and IS-136/PDC) and 3GPP2 (3GPartnership Project for cdma2000 standards based on backward compatibility with IS-95). Countries throughout the world are currently determining new radio spectrum bands to accommodate the 3G networks that will likely be deployed in the 2004–2005 time frame. ITU’s2000 World Radio Conference established the 2500–2690 MHz, 1710–1885 MHz, and 806– 960MHZ bands as candidates for 3G. In the US, additional spectrum in the upper UHF television bands near 700 MHz is also being considered for 3G. Given the economic downturn of the telecommunications industry during 2001, many governments throughout the world, including the US, had postponed their 3G auctions and spectrum decisions as of late 2001. Some European governments, however, auctioned off radio spectrum for 3G well before the telecommunications industry depression of 2001. The sale price of the spectrum was astounding! England’s first ever spectrum auction netted $35.5 Billion USD in April 2000 for five nationwide 3G licenses. Germany’s 3G auction generated $46 Billion USD later the same year,for four competing nationwide licenses [Buc00, pp. 32-36.].


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B.E./B.Tech. DEGREE EXAMINATION, MAY/JUNE 2013 Seventh Semester Electronics and Communication Engineering WIRELESS COMMUNICATION Time: Three hours

Maximum: 100 marks

Answer ALL questions PART A (10 ë 2 = 20 marks) 1. Define: Frequency reuse. 2. State the operating principle of adhoc networks. 3. State the differencesbetween small-scaleand large-scalefading. 4. Define: Snells law. 5. Mention any two criteria for choosing a modulation technique fora specific wireless application. 6. Draw the Structure of generic optimum receiver. 7. Define: Hamming distance. 8. State the principle of diversity. 9. Define: Direct Sequence-SpeedSpectrum. 10. State the goals ofa standard IMT -2000.

PART B (5 ë 16 = 80 marks) 11. (a) (i) Explain the methods for increasing the capacity of wireless cellular networks. (ii) Brief about the principle of Time Division Multiple Access (TDMA). Or

M02_ECE_7th-SEMESTER_CH02(May-Jun-2013).indd 1

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2.2


(b) (i) Describe in detail about the effects of multipath propagation in wireless environment. (ii) A Communication system has the following parameters:

P1 = 5 W, Gt (dB) = 13 dB, Gr (dB) = 17 dB, d = 80 km, f = 3 GHz, Determine the value of the received power. 12. (a) (i) Explain the time-variant two-path model of a wireless propagation channel. (ii) Brief about the properties of Rayleigh distribution. Or (b) (i) Explain the narrow band modeling methods for Short scale fading and Long scale fading. (ii) Brief about the properties of Nakagami distribution. 13. (a) (i) Explain the principle of p/4 – Differential Quadrature-Phase Shift Keying from a signal space diagram. (ii) Derive the expression for probability of error in Flat-Fading channels. Or (b) (i) Explain the principle of Minimum Shift Keying (MSK) modulation and derive the expression for power spectral density. (ii) Derive the expression for probability of error in FrequencyDispersive Fading Channels. 14. (a) (i) Explain any two diversity techniques to combat small-scale fading. (ii) Describe any two adaptation algorithms for Mean Square Error Equalizers. Or (b) (i) Write short notes on Linear Predictive voCoder. (ii) The generator matrix for a linear binary code is 0 0 0 1 1 1  G = 0 1 00111  1 0 0 1 1 1 0


1

   

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Wireless Communication (May/June 2013)

(1) Express

2.3

G in systematic [I/P] form.

(2) Determine the Parity check matrix H for the code. (3) Construct the table of syndromes for the code. (4) Determine the minimum distance of the code. 15. (a) (i) Explain the principle of cellular code division multiple access systems. (ii) Brief about the properties of spreading codes used in CDMA systems. Or (b) (i) Describe in detail about the operation of OFDM transceiver structures. (ii) Explain the physical layer features of WCDMA systems.


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Solutions PART A 1. Physical separation of two cells is sufficiently wide; the same subset of frequencies can be used in both cells. This is the concept of frequency reuse. The ability of frequency is to expand the total system capacity without the need of high power transmitters. This is called frequency reuse or frequency planning. 2. MANET - Mobile Ad-hoc –Network It is a self configuring network of mobile stations connected by wireless links. It is defined as Independent Basic Service Set in IEEE 802.11 standard. 3. In large – scale fading, it characterize the signal strength over large transmitter and receiver separation distances (several hundreds or thousands of meters). In small – scale fading, it characterize the rapid fluctuations of the received signal strength over very short distances between transmitter and receiver (a few meters) or short time duration (seconds). 4. Snell’s law states “how light ray reacts when it meets the interface of two media having difference indexes of refraction”.

Refractive model for Snell’s law Let the two media have refractive indexes n1 and n2, where n1 > n2 and f1 and f2 be the angles of incidence and angle of refraction respectively. Snell’s law stated mathematically is

n1 sin f1 = n2 sin f2


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2.5

5. The criteria for choosing a modulation technique for a wireless applications are (i) Fading and multi-path conditions in the mobile radio channel (ii) Data rate and (iii) Low bit rate. 6.

An M-branch (M-finger) RAKE receiver implementation. Each correlator detects a time shifted version of the srcinal CDMA transmission, and each finger of the RAKE correlates to a portion of the signal which is delayed by at least one chip in time from the other fingers. 7. Hamming distance are nothing but the difference between two code words. Ex. 1100 1000. Here hamming distance are 1. 8. Diversity is technique used to compensate for fading channel impairments and is usually implemented by using two or more receiving antennas. 9. In direct sequence spread spectrum (DS-SS), the narrow band message signal is multiplied by pseudo-random noise code sequence called the spreading signal. Each user has its own pseudorandom code word which is approximately orthogonal to all other code words. 10. International Mobile Telecommunication (IMT-2000) formerly called future public land mobile telecommunication system (FPLMTS) tried to establish a common world wide communication system that allowed for terminal and user mobility, supporting the data of Universal Personal Telecommunication (UPT).


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2.6


PART B 11. (a) (i) As the demand for wireless service increases, the number of channels assigned to a cell eventually becomes insufficient to support the required number of users. At this point, cellular design techniques are needed to provide more channels per unit coverage area. Techniques such as cell splitting, sectoring, and coverage zone approaches are used in practice to expand the capacity of cellular systems. Cell splitting allows an orderly growth of the cellular system. Sectoring uses directional antennas to further control the interference and frequency reuse of channels. The zone microcell concept distributes the coverage of a cell and extends the cell boundary to hard-to-reach places. While cell splitting increases the number of base stations in order to increase capacity, sectoring and zone microcells rely on base station antenna placements to improve capacity by reducing co-channel interference. Cell splitting and zone microcell techniques do not suffer the trunking inefficiencies experienced by sectored cells, and enable the base station to oversee all handoff chores related to the microcells, thus reducing the computational load at the MSC. 11. (a) (ii) Time Division Multiple Access (TDMA) systems divide the radio spectrum into time slots, and in each slot only one user is allowed to either transmit or receive. It can be seen from Figure 1 that each user occupies a cyclically repeating time slot, so a channel may be thought of as a particular time slot that reoccurs every frame, where N time slots comprise a frame. TDMA systems transmit data in a buffer-and-burst

Code Channel N

t lo eS m Ti

s Channel 3 Channel 2 Channel 1

Frequency

Time

Figure 1 TDMA scheme where each channel occupies a cyclically repeating time slot


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2.7

One TDMA Frame Preamble

Slot 1 Slot 2

TrailBits

Sync. bits

InformationMessage

Slot 3

TrailBits

Slot

InformationData

Guard Bits

Figure 2 TDMA frame structure. The frame is cyclically repeated over time

method, thus the transmission for any user is noncontinuous. This implies that, unlike in FDMA systems which accommodate analog FM, digital data and digital modulation must be used with TDMA. The transmission from various users is interlaced into a repeating frame structure as shown in Figure 2. It can be seen that a frame consists of a number of slots. Each frame is made uphalf of a of preamble, information message, and tail bits. In TDMA/TDD, the timeanslots in the frame information message would be used for the forward link channels and half would be used for reverse link channels. In TDMA/FDD systems, an identical or similar frame structure would be used solely for either forward or reverse transmission, but the carrier frequencies would be different for the forward and reverse links. In general, TDMA/FDD systems intentionally induce several time slots of delay between the forward and reverse time slots for a particular user, so that duplexers are not required in the subscriber unit. In a TDMA frame, the preamble contains the address and synchronization information that both the base station and the subscribers use to identify each other. Guard times are utilized to allow synchronization of the receivers between different slots and frames. Different TDMA wireless standards have different TDMA frame structures, and some are described in Chapter 11. The features of TDMA include the following: •

TDMA shares a single carrier frequency with several users, where each user makes use of nonoverlapping time slots. The number of time slots per frame depends on several factors, such as modulation technique, available bandwidth, etc.

•

Data transmission for users of a TDMA system is not continuous, but occurs in bursts. This results in low battery consumption, since the subscriber transmitter can be turned off when not in use (which is most of the time).


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2.8


•

Because of discontinuous transmissions in TDMA, the handoff process is much simpler for a subscriber unit, since it is able to listen for other base stations during idle time slots. An enhanced link control, such as that provided by mobile assisted handoff (MAHO) can be carried out by a subscriber by listening on an idle slot in the TDMA frame.

•

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

•

Adaptive equalization is usually necessary in TDMA systems, since the transmission rates are generally very high as compared to FDMA channels.

•

In TDMA, the guard time should be minimized. If the transmitted signal at the edges of a time slot are suppressed sharply in order to shorten the guard time, the transmitted spectrum will expand and cause interference to adjacent channels.

•

High synchronization overhead is required in TDMA systems because of burst transmissions. TDMA transmissions are slotted, and this requires the receivers to be synchronized for each data burst. In addition, guard slots are necessary to separate users, and this results in the TDMA systems having larger overheads as compared to FDMA.

•

TDMA has an advantage in that it is possible to allocate different numbers of time slots per frame to different users. Thus, bandwidth can be supplied on demand to different users by concatenating or reassigning time slots based on priority.

Efficiency of TDMA — The efficiency of a TDMA system is a measure of the percentage of transmitted data that contains information as opposed to providing overhead for the access scheme. The frame efficiency h f , is the percentage of bits per frame which contain transmitted data. Note that the transmitted data may include source and channel coding bits, so the raw end user efficiency of a system is generally less than h f . The frame efficiency can be found as follows. The number of overhead bits per frame is [Zie92],

bOH =N +rb + r +N tb p

N t gb

Nb

rg

(1)

where Nr is the number of reference bursts per frame, Nt is the number of traffic bursts per frame, br is the number of overhead bits per reference


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2.9

burst, bp is the number of overhead bits per preamble in each slot, and bg is the number of equivalent bits in each guard time interval. The total number of bits per frame, bT , is

bT = T f R

(2)

where Tf is the frame duration, and R f is the channel bit rate. The frame efficiency hf is thus given as hf

 b  = 1 − OH  ×100  bT 

%

(3)

Number of channels in TDMA system — The number of TDMA channel slots that can be provided in a TDMA system is found by multiplying the number of TDMA slots per channel by the number of channels available and is given by m( Btot 2 Bguard ) N (4) Bc −

=

Wherechannel. m is theNote maximum number TDMA users supported radio that two guardofbands, one at the low endonofeach the allocated frequency band and one at the high end, are required to ensure that users at the edge of the band do not “bleed over” into an adjacent radio service. 11. (b) (i) Refer Q. no. 12. (a) (i) from Nov/Dec 2013 11. (b) (ii) Given Data:

Pt = 5 W

Transmitter power,

Frequency, f = 3 GHz

Gt = 13 dB Gr = 17 dB d = 80 km usually, L = 1 km To find: Received power,

Pr

Pr (10 km) Solution: l


=

c f

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2.10


3 × 10

8

3 × 10

9

=

l

1 =

or 0.1

10

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( 4p )

2

d L

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2

2

2

0( 1. ) 2

×

)× 80

12 ×

11.05 =

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=

10.94

µW

12. (a) (i) Refer Q. no. 12. (a) (ii) from April/May 2012 12. (a) (ii) Rayleigh Fading Distribution In mobile radio channels, the Rayleigh distribution is commonly used to describe the statistical time varying nature of the received envelope of a flat fading signal, or the envelope of an individual multi path component. It is well known that the envelope of the sum of two quadrature Gaussian noise signals obeys a Rayleigh distribution. Figure 1 shows a Rayleigh distributed signal envelope as a function of time. The Rayleigh distribution has a probability density function (pdf) given by

r  r2  exp  2  − 2s 2  p( r ) =  s 

( 0 ≤ r ≤ ∞) (1) (r

0

< 0)

where s is the rms value of the received voltage signal before envelope detection, and s 2 is the time-average power of the received signal before envelope detection. The probability that the envelope of the received signal does not exceed a specified value R is given by the corresponding cumulative distribution function (CDF)

P ( )R

R

=(Pr ≤ r) = R () ∫0 p=−r−exp dr


1

 

R2  2s 2 

(2)

4/25/2014 11:35:52AM


2.11

Figure 1 A typical Rayleigh fading envelope at 900 MHz [from [Fun93] © IEEE] r

The mean value

mean

of the∞Rayleigh distribution is given by

rmean = =E r][

∫

)(rp = r dr=

.s

0

p

2

1 2533s

and the variance of the Rayleigh distribution is given by represents the ac power in the signal envelope ∞

r s2

= E [−r]2 =[E] 2

∫

(− ) r 2 p r dr

r

0

p = s 2  2 −  = 0.4292s 2  2

(3) 2

sr

which

2

s p

2 (4)

The rms value of the envlope is the square root of the mean square, or 2s , where s is the standard deviation of the srcinal com plex Gaussian signal prior to envelope detection. The median value of r is found by solving 1

and is

2 rmedian

rmedian

=

∫

P (r d) r

(5)

0

= 1.177s

(6)

Thus the mean and the median differ by only 0.55 dB in a Rayleigh fading signal. Note that the median is often used in practice, since fading


4/25/2014 11:35:54AM

2.12


data are usually measured in the field and a particular distribution cannot be assumed. By using median values instead of mean values, it is easy to compare different fading distributions which may have widely varying means. 12. (b) (i) Propagation models have traditionally fo cused on predicting the average received signal strength at a given distance from the transmitter, as well as the variability of the signal strength in close spatial proximity to a particular location. Propagation models that predict the mean signal strength for an arbitrary transmitter–receiver (T–R) separation distance are useful in estimating the radio coverage area of a transmitter and are called large-scale propagation models, since they characterize signal strength over large T–R separation distances (several hundreds or thousands of meters). On the other hand, propagation models that characterize the rapid fluctuations of the received signal strength over very short travel distances (a few wavelengths) or short time durations (on the order of seconds) are called small-scale or fading models. As a mobile moves over very small distances, the instantaneous received signal strength may fluctuate rapidly giving rise to small-scale fading. The reason for this is that the received signal is a sum of many contributions coming from different directions, as described in Chapter 5. Since the phases are random, the sum of the contributions varies widely; for example, obeys a Rayleigh fading distribution. In small-scale fading, the received signal power may vary by as much as three or four orders of magnitude (30 or 40 dB) when the receiver is moved by only a fraction of a wavelength. As the mobile moves away from the transmitter over much larger distances, the local average received signal will gradually decrease, and it is this local average signal level that is predicted by largescale propagation models. Typically, the local average received power is computed by averaging signal measurements over a measurement track of 5 l to 40 l. For cellular and PCS frequencies in the 1 GHz to 2 GHz band, this corresponds to measuring the local average received power over movements of 1 m to 10 m. Figure 1 illustrates small-scale fading and the more gradual largescale variations for an indoor radio communication system. Notice in the figure that the signal fades rapidly (small-scale fading) as the receiver moves, but the local average signal changes much more gradually with distance. This chapter covers large-scale propagation and presents a number of common methods used to predict received power in mobile communication systems.


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2.13

Figure 1 Small-scale and large-scale fading

13. (a) (i) The

p

4

o

/4 QPSK

shifted QPSK modulation is a quadrature phase shift keying

technique which offers a compromise between OQPSK and QPSK in terms of the allowed maximum phase transitions. p It may be demodulated in a coherent or noncoherent fashion. In QPSK, the maximum 4

phase change is limited to ±135°, as compared to 180° for QPSK and 90° for OQPSK. Hence, the bandlimited

p

QPSK signal preserves

4

the constant envelope property better than bandlimited QPSK, but is more susceptible to envelope variations than OQPSK. An extremely attractive feature of

p

4

QPSK is that it can be noncoherently detected,

which greatly simplifies receiver design. Further, it has been found that in the presence of multipath spread and fading, than OQPSK [Liu89]. Very often,

p

4

p

4

QPSK performs better

QPSK signals are differentially

encoded to facilitate easier implementation of differential detection or coherent demodulation with phase ambiguity in the recovered carrier. When differentially encoded, In a

p

4

p

4

QPSK is called

p

4

DQPSK.

QPSK modulator, signaling points of the modulated signal

are selected from two QPSK constellations which are shifted by

p

4

with respect to each other. Figure 1 shows the two constellations along with the combined constellation where the links between two signal points indicate the possible phase transitions. Switching between two


4/25/2014 11:35:55AM

2.14


constellations, every successive bit ensures that there is at least a phase shift which is an integer multiple of

p

4

radians between successive

symbols. This ensures that there is a phase transition for every symbol, which enables a receiver to perform timing recovery and synchronization. 13. (a) (ii) Refer Q. no. 13. (b) (ii) from Nov/Dec 2013 13. (b) (i) Minimum Shift Keying (MSK) Minimum shift keying (MSK) is a special type of continuous phasefrequency shift keying (CPFSK) wherein the peak frequency deviation is equal to 1/4 the bit rate. In other words, MSK is continuous phase FSK with a modulation index of 0.5. The modulation index of an FSK signal is similar to the FM modulation index, and is defined as KFSK = (2∆F)/ Rb, where ∆F is the peak RF frequency deviation and Rb is the bit rate. A modulation index of 0.5 corresponds to the minimum frequency spacing that allows two FSK signals to be coherently orthogonal, and the name minimum shift keying implies the minimum frequency separation (i.e., bandwidth) that allows orthogonal detection. Two FSK signals vH (t) and

vL (t) are said to be orthogonal if T

∫v

H

(t )v L(t )dt

=0

(1)

0

MSK is sometimes referred to as fast FSK, as the frequency spacing used is only half as much as that used in conventional noncoherent FSK [Xio94]. MSK is a spectrally efficient modulation scheme and is particularly attractive for use in mobile radio communication systems. It possesses properties such as constant envelope, spectral efficiency, good BER performance, and self-synchronizing capability.

Block diagram of noncoherent FSK receiver


4/25/2014 11:35:55AM


2.15

An MSK signal can be thought of as a special form of OQPSK where the baseband rectangular pulses are replaced with half-sinusoidal pulses [Pas79]. These pulses have shapes like the St. Louis arch during a period of 2Tb. Consider the OQPSK signal with the bit streams offset as shown in Figure. If half-sinusoidal pulses are used instead of rectangular pulses, the modified signal can be defined as MSK and for an N–bit stream is given by N −1

sMSK ( t ) =

∑ m( )1 (t

p t − )2cos iTb

2p f c t +

(2)

i=0 N −1

∑ mQt()pt(

iT− 2T

b

−) sin b ft

2p

c

i=0

where

  pt  sin p(t ) =   2Tb   0

 ≤ t ≤ 2Tb    elsewhere  0

(3)

and where mI(t) and mQ(t) are the “odd” and “even” bits of the bipolar data stream which have values of ±1 and which feed the in-phase and quadrature arms of the modulator at a rate of Rb/2. It should be noted that there are a number of variations of MSK that exist in the literature [Sun86]. For example, while one version of MSK uses only positive half-sinusoids as the basic pulse shape, another version uses alternating positive and negative half-sinusoids as the basic pulse shape. However, all variations of MSK are continuous phase FSK employing different techniques to achieve spectral efficiency [Sun86]. The MSK waveform can be seen as a special type of a continuous phase FSK if Equation (2) is rewritten using trigonometric identities as

sMSK ()t =

2 Eb cos Tb

 2p f t (−) m() t m t p t + f   c I Q k 2Tb  

(4)

where fk is 0 or p depending on whether mI(t) is 1 or −1. From Equation (4), it can be deduced that MSK has a constant amplitude. Phase continuity at the bit transition periods is ensured by choosing the carrier frequency to be an integral multiple of one fourth the bit rate, 1/4T. Comparing Equation (4) with Equation (6.97), it can be concluded that the MSK signal is an FSK signal with binary signaling frequencies of fc +1/4T and fc −1/4T.


4/25/2014 11:35:56AM

2.16


13. (b) (ii) Frequency Selective Fading If the channel possesses a constant-gain and linear phase response over a bandwidth that is smaller than the bandwidth of transmitted signal, then the channel creates frequency selective on the received signal. Under such conditions, the channel impulse response has a multipath delay spread which is greater than the reciprocal bandwidth of the transmitted message waveform. When this occurs, the received signal includes multiple versions of the transmitted waveform which are attenuated (faded) and delayed in time, and hence the received signal is distorted. Frequency selective fading is due to time dispersion of the transmitted symbols within the channel. Thus the channel induces intersymbol interface (ISI). Viewed in the frequency domain, certain frequency components in the received signal spectrum have greater gains than others. Frequency selective fading channels are much more difficult to model than flat fading channels since each multipath signal must be modeled and the channel must be considered to be a linear filter. It is for this reason that wideband multipath measurements are made, and models are developed from these measurements. When analyzing mobile communication systems, statistical impulse response models such as the two-ray Rayleigh fading model (which considers the impulse response to be made up of two delta functions which independently fade and have sufficient time delay between them to induce frequency selective fading upon the applied signal), or computer generated or measured impulse responses, are generally used for analyzing frequency selective small-scale fading. Figure 2 illustrates the characteristics of a frequency selective fading channel.

Figure 1 Flat fading channel characteristics


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2.17

Figure 2 Frequency selective fading channel characteristics.

For frequency selective fading, the spectrum S(f ) of the transmitted signal has a bandwidth which is greater than the coherence bandwidthBC of the channel. Viewed in the frequency domain, the channel becomes frequency selective, where the gain is different for different frequency components. Frequency selective fading is caused by multipath delays which approach or exceed the symbol period of the transmitted symbol. Frequency selective fading channels are also known aswide band channels since the bandwidth of the signal s(t) is wider than the bandwidth of the channel impulse response. As time varies, the channel varies in gain and phase across the spectrum of s(t), resulting in time varying distortion in the received signal r(t). To summarize, a signal undergoes frequency selective fading if

BS > B C

(1)

TS < st

(2)

and

≥

A common rule thumb isT that is flat if TS on 10the st and a channel is of frequency < 10a schannel , although thisfading is dependent S t specific type of modulation used. 14. (a) (i) Refer Q. no. 14. (a) (i) from April/May 2012 14. (b) (i) Linear Predictive Coders LPC Vocoders

Linear predictive coders (LPCs) [Sch85a] belong to the time domain class of vocoders. This class of vocoders attempts to extract the significant features of speech from the time waveform. Though LPC coders are


4/25/2014 11:35:57AM

2.18


computationally intensive, they are by far the most popular among the class of low bit rate vocoders. With LPC, it is possible to transmit good quality voice at 4.8 kbps and poorer quality voice at even lower rates. The linear predictive coding system models the vocal tract as an all pole linear filter with a transfer function described by H ( z) =

G M

(1)

1 + ∑ bk Z − k k =1

where G is a gain of the filter and z−1 represents a unit delay operation. The excitation to this filter is either a pulse at the pitch frequency or random white noise depending on whether the speech segment is voiced or unvoiced. The coefficients of the all pole filter are obtained in the time domain using linear prediction techniques [Mak75]. The prediction principles used are similar to those in ADPCM coders. However, instead of transmitting quantized values of the error signal representing the difference between the predicted and actual waveform, the LPC system transmits only selected characteristics of the error signal. The parameters include the gain factor, pitch information, and the voiced/unvoiced decision information, which allow approximation of the correct error signal. At the receiver, the received information about the error signal is used to determine the appropriate excitation for the synthesis filter. That is, the error signal is the excitation to the decoder. The synthesis filter is designed at the receiver using the received predictor coefficients. In practice, many LPC coders transmit the filter coefficients which already represent the error signal and can be directly synthesized by the receiver. Figure 1 shows a block diagram of an LPC system [Jay86].

Figure 1 Block diagram of a LPC coding system.


4/25/2014 11:35:57AM


2.19

Determination of Predictor Coefficients— The linear predictive coder uses a weighted sum of p past samples to estimate the present sample, where p is typically in the range of 10–15. Using this technique, the current sample sn can be written as a linear sum of the immediately preceding samples sn–k p

(2)

sn = ∑ka nk sn − + e k =1

where en is the prediction error (residual). The predictor coefficients are calculated to minimize the average energy E in the error signal that represents the difference between the predicted and actual speech amplitude

E

N N  p = ∑ en = ∑  ∑knk a n= n=  k = 2

1

1

s

−

0

 

2

(3)

where a0 = −1 Typically, the error is computed for a time window of 10 ms, which corresponds to a value of N = 80. To minimize E with respect to am , it is required to set the partial derivatives equal to zero ∂E ∂am

p

N

=

∑ 2sn m ∑ ak sn −

n =1 p

=

−k

=0

(4)

k =0 N

∑ ∑ sn m ns k −

−k

a =0

(5)

k = 0 n =1

The inner summation can be recognized as the correlation coefficient Crm and hence the above equation can be rewritten as p

∑ Cmk ak

=0

(6)

k =0

After determining the correlation coefficients Crm, Equation (6) can be used to determine the predictor coefficients. Equation (6) is often expressed in matrix notation and the predictor coefficients calculated using matrix inversion. A number of algorithms have been developed to speed up the calculation of predictor coefficients. Normally, the predictor coefficients are not coded directly, as they would require 8 bits to 10 bits per coefficient for accurate representation [Del93].The accuracy requirements are lessened by transmitting the reflection coefficients (a closely related parameter), which have a smaller dynamic range. These reflection coefficients can be adequately represented by 6 bits per coefficient. Thus, for a 10th order predictor, the total number of bits assigned to the model


4/25/2014 11:35:59AM

2.20


parameters per frame is 72, which includes 5 bits for a gain parameter and 6 bits for the pitch period. If the parameters are estimated every 15 ms to 30 ms, the resulting bit rate is in the range of 2400 bps to 4800 bps. The coding of the reflection coefficient can be further improved by performing a nonlinear transformation of the coefficients prior to coding. This nonlinear transformation reduces the sensitivity of the reflection coefficients to quantization errors. This is normally done through a logarea ratio (LAR) transform which performs an inverse hyperbolic tangent mapping of the reflection coefficients,Rn (k)

LAR n ( k) =tanh −(1

R(n )k)l og

10

 1 + R n (k )    1 − R n ( k ) 

(1)

Various LPC schemes differ in the way they recreate the error signal (excitation) at the receiver. Three alternatives are shown in Figure [Luc89]. The first one shows the most popular means. It uses two sources at the receiver, one of white noise and the other with a series of pulses at the current pitch rate. The selection of either of these excitation methods is based on the voiced/unvoiced decision made at the transmitter and communicated to the receiver along with the other information. This technique requires that the transmitter extract pitch frequency information which is often very difficult. Moreover, the phase coherence between the harmonic components of the excitation pulse tends to produce a buzzy twang in the synthesized speech. These problems are mitigated in the other two approaches: Multipulse excited LPC and stochastic or code excited LPC. Multipulse Excited LPC

Atal showed [Ata86] that, no matter how well the pulse is positioned, excitation by a single pulse per pitch period produces audible distortion. Therefore, he suggested using more than one pulse, typically eight per period, and adjusting the individual pulse positions and amplitudes sequentially to minimize a spectrally weighted mean square error. This technique is called the multipulse excited LPC (MPE-LPC) and results in better speech quality, not only because the prediction residual is better approximated by several pulses per pitch period, but also because the multipulse algorithm does not require pitch detection. The number of pulses used can be reduced, in particular for high pitched voices, by incorporating a linear filter with a pitch loop in the synthesizer. 15. (a) Refer Q. no. 15. (a) from April/May 2012.


4/25/2014 11:35:59AM

B.E./B.Tech. DEGREE EXAMINATION, NOV/DEC 2012 Seventh Semester Electronics and Communication Engineering WIRELESS COMMUNICATION Time: Three hours

Maximum: 100 marks

Answer ALL questions PART A (10 ë 2 = 20 marks) 1. What is flat fading? 2. Define signal to self-interferenceratio. 3. Distinguishbetween narrowbandand wideband systems. 4. What is Link Budget calculation? 5. Find the 3-dB bandwidth for a Gaussian low pass filter usedto produce0.25 GMSK with a channel data rate of Rb 270 kbps. What is the 90% power bandwidth in the RF channel? =

6. What is slotted frequency hopping? 7. Assume four branch diversity is used , where each branch receives an independent Rayleigh fading signal. If the average 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. 8. Define coding gain. 9. What is duplexing? 10. What is the speechcode used in IS-95 system?Why?

PART B (5 ë 16 = 80 marks) 11. (a) (i) Explain about the factors that influence small scale fading.


4/11/2014 5:15:44PM


2.9

(ii) Find the average fade duration for threshold levels? 0.01, ? 0.1 and ? 1, when the Doppler frequency is 200 Hz. =

=

=

Or (b) (i) Write a note on Noise and Interference Limited Systems. (ii) Discuss the principles of Cellular Networks. 12. (a) (i) How the received signal strength is predicted using the free space propagation model? Explain. (ii) Find the far field distance for an antenna with maximum dimension of 1 m and operating frequency of 900 MHz. Or (b) (i) With system theoretic description, explain the characteristics of Time Dispersive channels. (ii) Explain the three basic propagation mechanisms in a mobile communication system. 13. (a) (i) Briefly explain the structure of a wireless communication link. (ii) With block diagram, explain the MSK transmitter and receiver. Derive an expression for MSK and its power spectrum. Or (b) Derive an expression for (i) M-ary phase shift keying and M-ray quadrature amplitude modulation. (ii) Also derive an expression for their bit error probability. 14. (a) Explain in detail about (i) Polarization diversity (ii) Time diversity (iii) Frequency diversity Or (b)

(i) Explain the basic idea about linear and decision feedback equalisers and derive an expression for its minimum mean square error. (ii) With a suitable diagram, explain the channel coding and speech coding techniques.


4/11/2014 5:15:44PM

2.10


15. (a) (i) Discuss in detail about cellular code division multiple access systems with neat diagrams. (ii) Write short notes on transceiver implementation. Or (b) (i) Explain with neat diagram of orthogonal frequency division multiplexing. (ii) Write a note on second generation and third generation wireless networks and standards.


4/11/2014 5:15:44PM

Solutions PART A 1. If the mobile radio channel has a constant gain and linear phase response over a bandwidth which is greater than the bandwidth of the transmitted signal, then the received signal will undergo flat fading. The conditions for the flat fading are (i) BW of the signal < BW of the channel (ii) Delay spread < Symbol period 2. In general, the signal-to-interference ratio for cellular network can be written as

SR =

pdesired

∑ Pinterference, i i

Here P desired is the signal strength from the desired BS and PInterference is the signal strength from the ith interfering BS. 3. Narrowband systems – Here the available radio spectrum is divided into large number of narrowband channels. The bandwidth of a single channel is nothing but coherence bandwidth of the channel. Wideband systems – The transmission bandwidth of a single channel is much larger than the coherence bandwidth of the channel. 4. Link Calculation: In designing a system for reliable communications, we must perform a link budget calculation to ensure that sufficient power is available at the receiver to close the link and meet the SNR Requirement.

5. Figure 1 Block diagram of a GMSK transmitter using direct FM generation


4/23/2014 6:20:56PM


2.5

Figure 2 Block diagram of a GMSK receiver

From the problem statement T =

Solving for

1

Rb

1

=

270 ×10

3

= 3.7µs

=

67.567kHz

B, where BT = 0.25, B

0.25 =

0 25 . =

T

3.7 ×10

−6

Thus the 3-dB bandwidth is 67.567 kHz. To determine the 90% power band-width, use Table 6.3 to find that 0.57 Rb is the desired value. Thus, the occupied RF spectrum for a 90% power bandwidth is given by RF BW

=

0.57 R=b

×57 = 0 . × 270 10

3

153kHz 9 .

6. A form of spread spectrum in which the signal is broadcast over a seemingly random series of radio frequencies, hopping from frequency to frequency at fixed intervals. 7. Given: Five branches diversity, specific threshold, g = 10 dB, Γ = 20 dB. Hence γ /Γ = 0.1 P1(10 dB) = (1 − e−0.1)1 = 0.095, when M = 1 (no diversity) P5(10 dB) = (1 − e−0.1)5 = 0.0000078, whenM = 5 i.e. five branches are used. Mean SNR, 1 + 1 /2+= /1 3/ 1/ 4 1 5=) ). (20 . 2 28 dB 45 6 >>> g= + (+ 20 dB i.e. due to selection diversity average SNR improves multifold. Also, without diversity, the SNR drops below the specific threshold with the probability, i.e. four orders of magnitude greater than, if five branch diversity is used.


4/23/2014 6:20:57PM

2.6


8. Coding gain or Efficiency h

Number of message bits ( k ) =

Number of bits in code word ( n)

9. Duplexing is the process of transmitting information in both directions simultaneously between the sender and the receiver. 10. The speech coder used in the IS-95 system is the Qualcomm 9600 bps Code Excited Linear Predictive (QCELP) coder. The srcinal implementation of this vocoder detects voice activity, and reduces the data rate to 1200 bps during silent periods.

PART B 11. (a) (i) Refer 12. (a) (i) from Nov/Dec 2013 11. (a) (ii) 11. (b) (i) 11. (b) (ii) Refer 11. (a) from Nov/Dec 2013 12. (a) (i) Refer Q. no. 12. (a) (i) from April/May 2012 12. (a) (ii) Operating frequency, f = 900 MHz l = c/f = 3 × 108 m/s/900 × 106 Hz = 0.33 m Fraunhofer distance, df = 2D2/l = 2(1)2/0.33 = 6 m Path loss PL(dB) = −10 log [(l2)/(4p)2 d2] = −10 log [(0.33)2 / (4 × 3.14)2 × 36] = 47dB 12. (b) (i) Time Dispersion Parameters In order to compare different multipath channels and to develop some general design guidelines for wireless systems, parameters which grossly quantify the multipath channel are used. The mean excess delay, rms delay spread, and excess delay spread ( X dB) are multipath channel parameters that can be determined from a power delay profile. The time dispersive properties of wide band multipath channels are most commonly quantified by their mean excess delay (t ) and rms delay spread (st ). The mean excess delay is the first moment of the power delay profile and is defined to be

∑a t

∑ P(t

2

k

t

=

k

∑a

2

k

k

k

=

k

)t k

k

∑ P(t

k

)

(1)

k

The rms delay spread is the square root of the second central moment of the power delay profile and is defined to be


4/23/2014 6:20:58PM


st

t

=

2

−

(t ) 2

2.7

(2)

where

∑a t = 2 ∑a 2

k

t

2

k

k

k

2

k

∑ P(t )t = ∑ P(t ) k

2

k

k

(3)

k

k

These delays are measured relative to the first detectable signal arriving at the receiver at t0 = 0. Equations (1)−(3) do not rely on the absolute power level of P(t), but only the relative amplitudes of the multipath components within P(t). Typical values of rms delay spread are on the order of microseconds in outdoor mobile radio channels and on the order of nanoseconds in indoor radio channels. Table 1 shows the typical measured values of rms delay spread. It is important to note that the rms delay spread and mean excess delay are defined from a single power delay profile which is the temporal or spatial average of consecutive impulse response measurements collected and averaged over a local area. Typically, many measurements are made at many local areas in order to determine a statistical range of multipath channel parameters for a mobile communication system over a largescale area [Rap90]. The maximum excess delay (X dB) of the power delay profile is defined to be the time delay during which multipath energy falls to X dB below the maximum. In other words, the maximum excess delay is defined as tx − t0 where t0 is the first arriving signal and tX is the maximum delay at which a multipath component is within X dB of the strongest arriving multipath signal (which does not necessarily arrive at t0). Figure 1 illustrates the computation of the maximum excess delay for multipath components within 10 dB of the maximum. The maximum excess delay (X dB) defines the temporal extent of the multipath that is above a particular threshold. The value of tx is sometimes called the excess delay spread of a power delay profile, but in all cases must be specified with a threshold that relates the multipath noise floor to the maximum received multipath component. 12. (b) (ii) Refer Q. no. 11 (b) (i) from Nov/Dec 2013 13. (a) (i) 13. (a) (ii) Refer Q. no. 13 (b) (i) from April/May 2013


4/23/2014 6:20:59PM

2.8


13. (b) (i) M-ary Phase Shift Keying (MPSK) In M-ary PSK, the carrier phase takes on one of M possible values, namely, qi = 2(i − 1) p /M, where i = 1, 2, … M. The modulated waveform can be expressed as

si)(t

=

cos 2 Es Ts

 2p f c t( +), 2Mp i − 1 0

tT≤,i,

≤

21M =

s

(1)

…

Where Es = (log2 M) Eb is the energy per symbol andTs = (log2 M) Tb is the symbol period. The above equation can be rewritten in quadrature form as 2 Es 2p   cos ( ) i − 1cos(  Ts M 

si ()t =

−

2Es

Ts

 

sin ( i −)1

) 2p f c t , , , i = 1 2 … M

2p  sin( M 

2p)f c t

By choosing orthogonal basis signals f2 ()t

(2)

f1 ()t

=

2 cos( 2p )f c t , and Ts

2 sin( 2p )f c t defined over the interval 0 ≤ t ≤ Ts, the M-ary Ts PSK signal set can be expressed as =

s M − PSK ()t

 = 

( Ecos s

p  − ( Es )  )i − 1 (2),  f1 t sin

p   (i)− 1 2  f 2

t

  

(3)

i = 1, 2,…, M Since there are only two basis signals, the constellation of M-ary PSK is two dimensional. The M-ary message points are equally spaced on a circle of radius E s centered at the srcin. M-ary Quadrature Amplitude Modulation (QAM) In M-ary PSK modulation, the amplitude of the transmitted signal was constrained to remain constant, thereby yielding a circular constellation. By allowing the amplitude to also vary with the phase, anew modulation scheme called quadrature amplitude modulation (QAM) is obtained. Figure 1 shows the constellation diagram of 16-ary QAM. The constellation consists of a square lattice of signal points. The general form of an M-ary QAM signal can be defined as

s i ()t

=

2E min

Ts

cos( ai

0≤t≤T


) 2p f c t

+

2min E sin( bi ) Ts

2p f c t

i = 1, 2,…, M

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2.9

where Emin is the energy of the signal with the lowest amplitude, andai and bi are a pair of independent integers chosen according to the location of the particular signal point. Note that M-ary QAM does not have constant energy per symbol, nor does it have constant distance between possible symbol states. It reasons that particular values ofsi(t) will be detected with higher probability than others. 13. (b) (ii) 14. (a) Polarization Diversity At the base station, space diversity is considerably less practical than at the mobile because the narrow angle of incident fields requires large antenna spacings [Vau90]. The comparatively high cost of using space diversity at the base station prompts the consideration of using orthogonal polarization to exploit polarization diversity. While this only provides two diversity branches, it does allow the antenna elements to be co-located. In the early days of cellular radio, all subscriber units were mounted in vehicles and used vertical whip antennas. Today, however, over half of the subscriber units are portable. This means that most subscribers are no longer using vertical polarization due to hand-tilting when the portable cellular phone is used. This recent phenomenon has sparked interest in polarization diversity at the base station. Theoretical Model for Polarization Diversity It is assumed that the signal is transmitted from a mobile with vertical (or horizontal) polarization. It is received at the base station by a polarization diversity antenna with two branches. Figure 1 shows the theoretical model and the system coordinates. As seen in the figure, a polarization diversity antenna is composed of two antenna elements V1 and V2, which make a ±a angle (polarization angle) with the Y axis. A mobile station is located in the direction of offset angle b from the main beam direction of the diversity antenna as seen in Figure 1b.

Some of the vertically signals are convertedThe to the horizontal polarized polarized signal because oftransmitted multipath propagation. signal arriving at the base station can be expressed as

x = r1 cos (wt + f1)

(1)

y = r2 cos (wt + f2)

(2)

where x and y are signal levels which are received when b = 0. It is assumed that r1 and r2 have independent Rayleigh distributions, andf1 and f2 have independent uniform distributions.


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2.10


Y V1

V2

X

(a) Z X Multipath

Main Beam

Mobile

(b)

Figure 1 Theoretical model for base station polarization diversity based on

[Koz85]: (a) x-y plane; (b) x–z plane The received signal values at elements V1 and V2 can be written as

V1 = (ar1 cos f1 + r2b cos f2) cos w t − (ar1 sin f1 + r2b sin f2) sin w t

(3)

V2 = (−ar1 cos f1 + r2b cos f2) cos w t − (−ar1 sin f1 + r2b sin f2) sin w t (4) where a

= sina cosb and b = cos a

The correlation coefficient r can be written as

 tan 2( a)c os 2 ( b) − Γ  r= 2  tan ( a)c os 2( b) + Τ  where

2

(5)

2

X

=

R2

(6)

2

R1

and 2

2

r2 b

2

2

2+r21r ab cos( f 1 2

f)

(7)

2

2

r2 b

2

2

2+r21r ab cos( 1 2f

f)

(8)

R1

=

r+ a+ 1

R2

=

r1+ a−


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2.11

Here, X is the cross polarization discrimination of the propagation path between a mobile and a base station. Frequency Diversity Frequency diversity is implemented by transmitting information on more than one carrier frequency. The rationale behind this technique is that frequencies separated by more than the coherence bandwidth of the channel will be uncorrelated and will thus not experience the same fades [Lem91]. Theoretically, if the channels are uncorrelated, the probability of simultaneous fading will be the product of the individual fading probabilities. Frequency diversity is often employed in microwave line-of-sight links which carry several channels in a frequency division multiplex mode (FDM). Due to tropospheric propagation and resulting refraction, deep fading sometimes occurs. In practice, 1: N protection switching is provided by a radio licensee, wherein one frequency is nominally idle but is available on a stand-by basis to provide frequency diversity switching for any one of the N other carriers (frequencies) being used on the same

link, each carrying independent traffic. When diversity is needed, the appropriate traffic is simply switched to the backup frequency. This technique has the disadvantage that it not only requires spare bandwidth but also requires that there be as many receivers as there are channels used for the frequency diversity. However, for critical traffic, the expense may be justified. New OFDM modulation and access techniques exploit frequency diversity by providing simultaneous modulation signals with error control coding across a large bandwidth, so that if a particular frequency undergoes a fade, the composite signal will still be demodulated. Time Diversity Time diversity repeatedly transmits information at time spacings that

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


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2.12


14. (b) (i) Refer Q. no. 14 (a) (ii) from April/May 2012 14. (b) (ii) Refer Q. no. 14 (b) from April/May 2012 15. (a) (i) Refer Q. no. 15 (a) from April/May 2012 15. (a) (ii) 15. (b) (i) 15. (b) (ii) Refer Q. no. 15 (b) from Nov/Dec 2013


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B.E./B.Tech. DEGREE EXAMINATION, MAY/JUNE 2012 Seventh Semester Electronics and Communication Engineering WIRELESS COMMUNICATION Time: Three hours

Maximum: 100 marks

Answer ALL questions PART A (10 ë 2 = 20 marks) 1. What are the different types of multiple Access schemes? 2. Mention the significance of frequency reuse in Cellular Networks. 3. List the different types of wireless channels. 4. What is frequency selectivefading? How to avoid fadingproblem? 5. List the advantagesof QPSK. 6. Differentiatebetween MSK and GMSK. 7. List the different types of Speech coding techniques. 8. State the significanceof linear and decision feedback equalizer. 9. State effects of multipath propagationon CDMA. 10. List a few wireless network standards.

PART B (5 ë 16 = 80 marks) 11. (a) (i) Explain in detail Wide Area Data Services and Broadband Wireless Access services offered to Wireless networks. (ii) What are Paging Systems? Explain. Or


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2.2


(b) (i) With a neat block diagram, explain the Cellular Network Architecture. (ii) Explain any one type of Multiple Access scheme. 12. (a) (i) Explain the free space path loss and derive the gain expression. (ii) Describe in detail Two Ray Model propagation mechanism. Or (b) (i) Define the following Auto-correlation, Cross correlation and power spectral density for narrow band fading model. (ii) What is the need for link calculation? Explain with suitable example. 13. (a) Explain with neat signal diagrams, the modulation and demodulation technique of QPSK. Or (b) (i) Describe with a block diagram Offset-Quadrature Phase Shift Keying and its advantages. (ii) Explain the concept of GMSK and mention its advantages. 14. (a) (i) With a neat block diagram, explain the principle of diversity. (ii) Explain in detail Decision feedback equalizer. Or (b) (i) Explain any one method of channel coding. (ii) What are the advantages of speech coding? Explain any one technique of speech coding 15. Explain: (a) Code Division Multiple Access (CDMA) and compare its performance with TDMA. Or (b) What is orthogonal frequency division multiplexing? Explain OFDM technique and mention its merits, demerits and application.


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Solutions PART A 1. The different types of multiple access schemes are (i) Frequency Division Multiple Access (FDMA) (ii) Time Division Multiple Access (TDMA) (iii) Code Division Multiple Access (CDMA) and (iv) Space Division Multiple Access (FDMA) 2. Physical separation of two cells is sufficiently wide; the same subset of frequencies can be used in both cells. This is the concept of frequency reuse. The ability of frequency is to expand the total system capacity without the need of high power transmitters. This is called frequency reuse or frequency planning. 3. Different types of wireless channels are (i) Flat frequency (or) time selective channels and (ii) Frequency selective channels 4. A signal undergoes frequency selective fading if (i) BW of signal (BS) > BW of channel (BC) and (ii) Symbol period )(TS

<

Delay period ()

st

5. Advantages of QPSK (i) It has twice the bandwidth efficiency of BPSK, since two bits are transmitted in a single modulation symbol. (ii) Higher data rate. 6. MSK: (i) Minimum shift keying is a special type of continuous phasefrequency shift keying (CPFSK) wherein the peak frequency deviation is equal to ¼ the bit rate. (ii) MSK is a spectrally efficient modulation scheme and is particularly attractive for use in mobile radio communication systems. (iii) Main lobe of MSK is wide. So, it is not suitable for multi-user communications.


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2.4


Where as, GMSK: (i) Gaussian refers to the shape of a filter. It is a simple binary modulation scheme. (ii) The side lobe levels of the spectrum are much reduced by passing the modulating NRZ data waveform through a premodulation Gaussian pulse-shaping filter. (iii) GMSK has excellent power efficiency and spectral efficiency than conventional FSK. 7. Different types of speech coding techniques are (i) Linear Predictive Coders (LPCs) (ii) Adaptive Predictive Coding (APC) (iii) Adaptive Differential Pulse Code Modulation (ADPCM) (iv) Sub-Band Coding (SBC) and (v) Adaptive Transform Coding (ATC) 8. Linear equalizer is a simple one and use FIR filter structure. Decision feedback equalizer is a nonlinear equalizer that are used in applications where the channel distortions too severe for linear equalizer to handle. The main advantage for this, once an information symbol has been detected and decided upon, the ISI that it induces on future symbols can be estimated. 9. Effects of multi-path propagation on CDMA: (i) In CDMA, the power of multiple users at a receivers determines the noise floor after decorrelation. If the power of each user within a cell is not controlled such that they do not appear equal at the base station receiver. (ii) The near-far problem occurs when many users share the same channel. 10.

(i) Digital European Cordless Telephone (DECT) (ii) Advanced Mobile Phone System (AMPS) (iii) Global System for Mobile Communication (GSM) and (iv) Cellular Digital Packet Data (CDPD)


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2.5

PART B 11. (a) (i) Refer Q. no. 11 (a) from Nov/Dec 2013 11. (a) (ii) Refer Q. no. 11 (a) from Nov/Dec 2013 (only paging systems) 11. (b) (i) Refer Q. no. 11 (b) from Nov/Dec 2013 11. (b) (ii) Time Division Multiple Access (TDMA)

Time Division MultipleAccess (TDMA) systems divide the radio spectrum into time slots, and in each slot only one user is allowed to either transmit or receive. It can be seen from Figure 1 that each user occupies a cyclically repeating time slot, so a channel may be thought of as a particular time slot that reoccurs every frame, where N time slots comprise a frame. TDMA systems transmit data in a buffer-and-burst method, thus the transmission for any user is noncontinuous. This implies that, unlike in FDMA systems which accommodate analog FM, digital data and digital modulation must be used with TDMA. The transmission from various users is interlaced into a repeating frame structure as shown in Figure 2. It can be seen that a frame consists of a number of slots. Each frame is made ofof a preamble, an information message, and tail bits. Inwould TDMA/ TDD, up half the time slots in the frame information message be used for the forward link channels and half would be used for reverse link channels. In TDMA/FDD systems, an identical or similar frame structure would be used solely for either forward or reverse transmission, but the carrier frequencies would be different for the forward and reverse links. In general, TDMA/FDD systems intentionally induce several time slots of delay between the forward and reverse time slots for a particular user, so that duplexers are not required in the subscriber unit. Code Channel N

t lo me S i T

s Channel 3 Channel 2 Channel 1

Frequency

Time

Figure 1 TDMA scheme where each channel occupies a cyclically repeating time slot


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2.6


In a TDMA frame, the preamble contains the address and synchronization information that both the base station and the subscribers use to identify each other. Guard times are utilized to allow synchronization of the receivers between different slots and frames. Different TDMA wireless standards have different TDMA frame structures, and some are described in Chapter 11. The features of TDMA include the following: One TDMA Frame Preamble

Slot 1 Slot 2

TrailBits

Sync. bits

InformationMessage

Slot 3

TrailBits

Slot

InformationData

Guard Bits

Figure 2 TDMA frame structure. The frame is cyclically repeated over time

• TDMA shares a single carrier frequency with several users, where each user makes use of nonoverlapping time slots. The number of time slots per frame depends on several factors, such as modulation technique, available bandwidth, etc.

• Data transmission for users of a TDMA system is not continuous, but occurs in bursts. This results in low battery consumption, since the subscriber transmitter can be turned off when not in use (which is most of the time).

• Because of discontinuous transmissions in TDMA, the handoff process is much simpler for a subscriber unit, since it is able to listen for other base stations during idle time slots. An enhanced link control, such as that provided by mobile assisted handoff (MAHO) can be carried out by a subscriber by listening on an idle slot in the TDMA frame.

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

• Adaptive equalization is usually necessary in TDMA systems, since the transmission rates are generally very high as compared to FDMA channels.


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2.7

• In TDMA, the guard time should be minimized. If the transmitted signal at the edges of a time slot are suppressed sharply in order to shorten the guard time, the transmitted spectrum will expand and cause interference to adjacent channels. High synchronization overhead is required in TDMA systems

• because of burst transmissions. TDMA transmissions are slotted, and this requires the receivers to be synchronized for each data burst. In addition, guard slots are necessary to separate users, and this results in the TDMA systems having larger overheads as compared to FDMA.

• TDMA has an advantage in that it is possible to allocate different numbers of time slots per frame to different users. Thus, bandwidth can be supplied on demand to different users by concatenating or reassigning time slots based on priority.

Efficiency of TDMA — The efficiency of a TDMA system is a measure of the percentage of transmitted data that contains information as opposed to providing overhead for the access scheme. The frame efficiency,hf , is the bitsmay per frame which contain data.bits, Noteso that the percentage transmittedofdata include source and transmitted channel coding the raw end-user efficiency of a system is generally less thanhf . The frame efficiency can be found as follows. The number of overhead bits per frame is [Zie92], bOH =N+b + +Nb rr tP

Nt gb

N b rg

(1)

where Nr is the number of reference bursts per frame, Nt is the number of traffic bursts per frame, br is the num ber of overhead bits per reference burst, bP is the number of overhead bits per preamble in each slot, and bg is the number of equivalent bits in each guard time interval. The total number of bits per frame,bT , is

bT = T f R

(2)

where Tf is the frame duration, and R is the channel bit rate. The frame efficiency hf , is thus given as hf

 b  = 1 − OH  ×100%  bT 

(3)

Number of channels in TDMA system — The number of TDMA channel slots that can be provided in a TDMA system is found by


4/25/2014 9:38:05PM

2.8


multiplying the number of TDMA slots per channel by the number of channels available and is given by N

m( Btot =

−

2 Bguard )

Bc

(4)

Where m is the maximum number of TDMA users supported on each radio channel. Note that two guard bands, one at the low end of the allocated frequency band and one at the high end, are required to ensure that users at the edge of the band do not “bleed over” into an adjacent radio service. 12. (a) (i) The free space propagation model is used to predict received signal strength when the transmitter and receiver have a clear, unobstructed line-of-sight path between them. Satellite communication systems and microwave line-of-sight radio links typically undergo free space propagation. As with most large-scale radio wave propagation models, the free space model predicts that received power decays as a function of the T–R separation distance raised to some power (i.e. a power law function). The free space power received by a receiver antenna which is separated from a radiating transmitter antenna by a distance d, is given by the Friis free space equation, Pr (d )

=

PG Grl t t

2

( 4p )2 d 2 L

(1)

Where Pt is the transmitted power, Pr(d) is the received power which is a function of the T–R separation, Gt is the transmitter antenna gain, G r is the receiver antenna gain, d is the T–R separation distance in meters, L is the system loss factor not related to propagation (L ≥ 1) and l is the wavelength in meters. The gain of an, antenna is related to its effective aperture Ae, by G

4p A e

=

l

2

(2)

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

=

c f

2l c =

wc

(3)

where f is the carrier frequency in Hertz, wc is the carrier frequency in radians per second, and c is the speed of light given in meters/s. The values for Pt or Pr must be expressed in the small units, and Gt and Gr are dimensionless quantities. The miscellaneous losses L (L ≥ 1) are usually due to transmission line attenuation, filter losses, and antenna losses in


4/25/2014 9:38:06PM


2.9

the communication system. A value of L = 1 indicates no loss in the system hardware. The Friis free space equation of (1) shows that the received power falls off as the square of the T–R separation distance. This implies that the received power decays with distance at a rate of 20 dB/decade. An isotropic radiator is an ideal antenna which radiates power with unit gain uniformly in all directions, and is often used to reference antenna gains in wireless systems. The effective isotropic radiated power (EIRP) is defined as and represents the maximum radiated power available from a transmitter in the direction of maximum antenna gain, as compared to an isotropic radiator.

EIRP = Pt Gt

(4)

In practice, effective radiated power (ERP) is used instead of EIRP to denote the maximum radiated power as compared to a half-wave dipole antenna (instead of an isotropic antenna). Since a dipole antenna has a gain of 1.64 (2.15 dB above an isotrope), the ERP will be 2.15 dB smaller than the EIRP for the same transmission system. In practice, antenna are or given units dBirespect (dB gain respectdipole) to an isotropic gains antenna) dBdin (dB gainofwith to awith half-wave [Stu81]. The path loss, which represents signal attenuation as a positive quantity measured in dB, is defined as the difference (in dB) between the effective transmitted power and the received power, and may or may not include the effect of the antenna gains. The path loss for the free space model when antenna gains are included is given by PL (dB)

= 10log

Pt Pr

G G l2  = 10 − log  t r2 2   ( 4p ) d 

(5)

When antenna gains are excluded, the antennas are assumed to have unity gain, and path loss is given by PL (dB)

= 10log

Pt Pr

 l2  = 10 − log  2 2   ( 4p ) d 

(6)

The Friis free space model is only a valid predictor for Pr for values of d which are in the far-field of the transmitting antenna. The far-field, or Fraunhofer region, of a transmitting antenna is defined as the region beyond the far-field distance df , which is related to the largest linear dimension of the transmitter antenna aperture and the carrier wavelength. The Fraunhofer distance is given by


4/25/2014 9:38:07PM

2.10


df

=

2D

2

(7)

l

where D is the largest physical linear dimension of the antenna. Additionally, to be in the far-field region, df must satisfy

df » D

(8)

df » l

(9)

and

Furthermore, it is clear that Equation (1) does not hold for d = 0. For this reason, large-scale propagation models use a close-in distance, d0, as a known received power reference point. The received power, Pr(d), at any distance d > d0, may be related to Pr at d0. The value Pr(d0) may be predicted from Equation (1), or may be measured in the radio environment by taking the average received power at many points located at a close-in radial distance d0 from t he transmitter. The reference distance must be chosen such that it lies in the far-field region, that is, 0 and d0 is chosen to be smaller anyusing practical distance d df ,mobile in ≥the communication system.than Thus, Equation (1),used the

received power in free at a distance greater than d0 is given by

 d0   d 

2

Pr ( d ) = Pr ( d0 ) 

d

≥ d0 ≥ d f

(10)

In mobile radio systems, it is not uncommon to find that Pr may change by many orders of magnitude over a typical coverage area of several square kilometers. Because of the large dynamic range of received power levels, often dBm or dBW units are used to express received power levels. Equation (10) may be expressed in units of dBm or dBW by simply taking the logarithm of both sides and multiplying by 10. For example, if Pr is in units of dBm, the received power is given by where Pr(d0) is in units of watts.

Pr ( d ) dBm = 10log

 Pr ( d0 )   0.001 W +  

 d0   d 

20 log 

d

≥ d0 ≥ d f

(11)

The reference distance d0 for practical system using low-gain antennas in the 1–2 GHz region is typically chosen to be 1 m in indoor environments and 100 m or 1 km in outdoor environments, so that the numerator in Equations (10) and (11) is a multiple of 10. This makes path loss computations easy in dB units. Operating frequency,


f = 900 MHz

4/25/2014 9:38:08PM


l

Fraunhofer distance, d f

3 10 ×m s 8/ /

=

cf /

=

2D 2 / l

=

=

2.11

900 10× Hz 6 0.33 = m

2(1) 2 / 0.33 = 6 m

path loss PL (dB ) = −10 log [(l 2 )/(4p ) 2 d 2 ] = −

2

×

10l og [(0.33) /(4 3.14)

2

×

36] = 47d B

12. (a) (ii) Reflection from Perfect Conductors Since electromagnetic energy cannot pass through a perfect conductor a plane wave incident on a conductor has all of its energy reflected. As the electric field at the surface of the conductor must be equal to zero at all times in order to obey Maxwell’s equations, the reflected wave must be equal in magnitude to the incident wave. For the case when E-field polarization is in the plane of incidence, the boundary conditions require that [Ram65]

qi = qr

(1)

and (E-field in plane of incidence) E =E i

(2)

r

Similarly, for the case when the E-field is horizontally polarized, the boundary conditions require that

qi = qr

(4.31)

and normal to plane of incidence) Ei = E(E-field r

(4.32)

Referring to Equations (1) to (4), we see that for a perfect conductor, Γ|| = 1, and Γ⊥ = –1 regardless of incident angle. Elliptical polarized waves may be analyzed by using superposition. Ground Reflection (Two-Ray) Model In a mobile radio channel, a single direct path between the base station

and a mobile is seldom the only physical means for propagation, and hence the free space propagation model of Equation (3) is in most cases inaccurate when used alone. The two-ray ground reflection model shown in Figure 3 is a useful propagation model that is based on geometric optics, and considers both the direct path and a ground reflected propagation path between transmitter and receiver. This model has been found to be reasonably accurate for predicting the large-scale signal strength over distances of several kilometers for mobile radio systems that use tall towers (heights which exceed 50 m), as well as for line-ofsight microcell channels in urban environments [Feu94].


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2.12


In most mobile communication systems, the maximum T–R separation distance is at most only a few tens of kilometers, and the earth may be assumed to be flat. The total received E-field, ETOT, is then a result of the direct line-of-sight component, ELOS, and the ground reflected component, Eg.

Figure 1 Two-ray ground reflection model.

Referring to Figure 1, h is the height of the transmitter and h is the r height of the receiver. If Et 0 is the free space E-field (in units of V/m) at a reference distance d0 from the transmitter, then for d > d0, the free space propagating E-field is given by E (d , t )

=

E 0d 0 d

 

 − d   c  

cos  w c  t

(d

> d0 )

(5)

where E(d, t) = E0d0/d represents the envelope of the E-field at d meters from the transmitter. Two propagating waves arrive at the receiver: the direct wave that travels a distance d′; and the reflected wave that travels a distanced″. The E-field due to the line-of-sight component at the receiver can be expressed as E d E LOS (d, ) t

′

=

0 cos

0

d

′

d

 w  t − c′  

(6)

c

and the E-field for the ground reflected wave, which has a propagation distance of d″, can be expressed as

E g ( d ,″) t

=Γ

E 0 d0 cos d″

  d″  w c  t − c  

(7)

According to laws of reflection in dielectrics given in Section 4.5.1

qi = q0


(8)

4/25/2014 9:38:09PM


2.13

and

Eg = Γ Ei

(9)

Et = (1 + Γ) Ei

(10)

where Γ is the reflection coefficient for ground. For small values of qi (i.e., grazing incidence), the reflected wave is equal in magnitude and 180° out of phase with the incident wave, as shown in Example 4.4. The resultant E-field, assuming perfect horizontal E-field polarization and ground reflection (i.e., Γ⊥ = -1 and Et = 0), is the vector sum of ELOS and Eg, and the resultant total E-field envelope is given by

|ETOT| = |ELOS + Eg|

(11)

The electric field ETOT (d, t) can be expressed as the sum of Equations (6) and (7) ETOT (,)d t

Using the

E d

0 =0 cos d′

c

  d0′  0 −1  w  t − c( )  + cos

E d c

d

′′

  d″  w  t − c  

(12)

method of images, which is demonstrated by the geometry of

Figure 2, the path difference, ∆, between the line-of-sight and the ground reflected paths can be expressed as ∆ = d−″ d= ′ h+ h( +t rd)−

2 2

h− h( + )d t r

2 2

(13)

When the T–R separation distance d is very large compared to ht + hr Equation (13) can be simplified using a Taylor series approximation ∆ = d″ − d′ ≈

2ht hr

d

(14)

Figure 2 The method of images is used to find the path difference between the line-of-sight and the ground reflected paths


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2.14


Once the path difference is known, the phase difference q∆ between the two E-field components and the time delay td between the arrival of the two components can be easily computed using the following relations 2p∆

q∆ =

∆w

=

l

(15)

c

c

and ∆

td =

c

=

q∆

(16)

2p f c

It should be noted that as d becomes large, the difference between the distances d′ and d″ becomes very small, and the amplitudes of ELOS and Eg are virtually identical and differ only in phase. That is E 00d

E d

00

≈

d

d′

00

≈

E d

(17)

d″

If the received E-field is evaluated at some time, say at t = d″/c, Equation (12) can be expressed as a phasor sum ETOT

d  d , t d ″  = 0E0cos   c  d′

= ≈

E 00d d

c

  d ″ 0− d0 ′   E d  w  c   −cos d ′′

∠q ∆ 00−

′

E 0d 0 d

[ ∠q ∆

0



(18)

E d d

′′

− 1]

where d is the distance over a flat earth between the bases of the transmitter and receiver antennas. Referring to the phasor diagram of Figure 5 which shows how the direct and ground reflected rays combine, the electric field (at the receiver) at a distanced from the transmitter can be written as ) d E(TOT

=

 E0 d0  (cos



d



2

)

q

E d  − 1 2 +sin 0 0 ∆  d 

2 2

(19)

q ∆

Figure 3 Phasor diagram showing the electric field components of the line-of-sight, ground reflected, and total received E-fields, derived from Equation (18)


4/25/2014 9:38:10PM


E0 d 0

() d = ETOT

d

cos 2−2

2.15

q∆

(20)

Note that if the E-field is assumed to be in the plane of incidence (i.e., vertical polarization) then Γ|| = 1 and Equation (20) would have a “ +” instead of a “–”. Using trigonometric identities, Equation (20) can be expressed as

=2

() d ETOT

q∆    2

E0 d 0

sin d

(21)

Equation (21) is an important expression, as it provides the exact received E-field for the two-ray ground reflection model. One notes that for increasing distance from the transmitter, ETOT (d) decays in an oscillatory fashion, with local maxima being 6 dB greater than the free space value and the local minima plummeting to – ∞ dB (the received E-field cancels out to zero volts at certain values of, although in reality this never happens). Once the distance d is sufficiently large, q∆ becomes ≤p and the received E-field, ETOT (d), then falls off asymtotically with increasing distance. Note that Equation (21) may be simplified whenever (q∆/2) ≈ q∆/2. This occurs when q∆/2 is less than 0.3 radian. Using Equations (14) and (15) q∆ 2

≈

2pht hr

ld

< 0.3 rad

(22)

which implies that Equation (21) may be simplified whenever d

hh 20p >

hh20 tr

tr

≈

3l

(23)

l

Thus, as long as d satisfies (23), the received E-field can be approximated as 2E 0d 0 2p ht hr k V/m ETOT (d ) (24) 2 ≈

d

≈

ld

d

Where k is a constant related to E0, the antenna heights, and the wavelength. This asymptotic behavior is identical for both the E-field in the plane of incidence or normal to the plane of incidence. The free space power received at d is related to the square of the electric field through Equation (24). Combining Equations (24), the received power at a distance d from the transmitter for the two-ray ground bounce model can be expressed as 2

Pr


=

PG G

t t r

2

ht hr d

4

(25)

4/25/2014 9:38:12PM

2.16


As seen from Equation (25) at large distances (d » ht hr ), the received power falls off with distance raised to the fourth power, or at a rate of 40 dB/decade. This is a much more rapid path loss than is experienced in free space. Note also that at large values of d, the received power and path loss become independent of frequency. The path loss for the two-ray model (with antenna gains) can be expressed in dB as PL (dB)

=

40 log− (d 10 log +

log G +10 t

+

20 Glog tr

r

20log ) h

h

(26)

At small T–R separation distances, Equation (12) must be used to compute the total E-field. When Equation (15) is evaluated for q ∆ = p then d ( 4 ht hr ) /l is where the ground appears in the first Fresnel zone between the transmitter and receiver (Fresnel zones are treated in Section 4.7.1). The first Fresnel zone distance is a useful parameter in microcell path loss models [Feu94]. ,

=

12. (b) (i) This

∫

t

2p f Dn dt

assumption

the

Doppler

phase

shift

is3 fn

(t ) =

= 2p f Dn t , and the phase of the nth multipath component

n n becomes (t) = a2key − 2p f Dnt −we p fctassumption: f0.assume that for the nth multipath We nowfmake component the term fctn in fn (t) changes rapidly relative to all other phase terms in this expression. This is a reasonable assumption since fc is large and hence the term fctn can go through a 360 degree rotation for a small change in multipath delay tn. Under this assumption fn (t) is uniformly distributed on [−p, p]. Thus

t E [ r(1 )]

= Et  ∑ acosf ()  n

n

n

∑] fE[cos

t a =[



n

E ()]

,

n

=0

n

Consider now the autocorrelation of the in-phase and quadrature components. Using the independence of an and f the independence of fn and fm, n ≠ m, and the uniform distribution of fn we get that

trEt[ r1() ()] Q

cos (f)m tm asinf( ) = Et  ∑ n a n ∑m  n

 

(1)

= ∑ ∑ E [a na m] E[cos fn (t)s in fm( t)] n

=∑

m

E [a n2] E[ cos fn( )s t in f(m )]t

n

= 0. Thus r1 (t) and rQ (t) are uncorrelated and, since they are jointly Gaussian processes, this means they are independent.


4/25/2014 9:38:13PM


2.17

Following a similar derivation as in (1) we obtain the autocorrelation of r1 (t) as A r1 (t,

a [∑f]E[cos f+n2t (E) cos (n

t) = [E(=)r1+(t r1t )] t

)].

t

n

(2)

t

n

Now making the substitution fn (t ) ptn2cf p =

E [cos fn( t)cfostt (n

+

)] .=p Dt5[cos E f

2

.5E [cos( 4p tf c

D ft2f

−

−

0

. We get

]

n

+

n

p 4 tf D n

n +

n −

4p f D n t

(3)

-p 2 fD t n f− 2 )]. 0

Since fctn changes rapidly relative to all other phase terms and is uniformly distributed, the second expectation term in (3) goes to zero, and thus 2 A r1 (t, )t .= 5∑ [a E] [cos( Et n p

2 )]f D n

(4)

n

= .5∑ E[ a]cos( put 2 cosq n 2

n

/l ),

n =

u cos q / l

A

t

t

n since f Dn is assumed fixed. Note that r1 ( , ) depends only on (A),r1 t and thus r1 (t ) is a wide-sense stationary (WSS) t , A r(1 )t,t random process. Using a similar derivation we can show that the quadrature component is also WSS with autocorrelation ArQ (t) A ( r)1 t . In addition, the cross correlation between the in-phase and quadrature components depends only on the time difference t and is given by =

=

A t r t +t (,t )t = A (1 ,)Q Qt [ (=) E ( r1 )] r r1 , Q r r = 5 ∑ E [an]sin( 2put cos q n /l ) 2

= − E [ rQ( )t (r1 t +)]. t

(5)

S The densities ) ofbyr1taking (t) andthe by Sr1 (of rQ (Fourier t), denoted f) and power respectively, are (PSD obtained transform SrQ (f),spectral their respective autocorrelation functions relative to the delay parameter t. Since these autocorrelation functions are equal, so are the PSDS. Thus

S r1f( ) S=f (rQ) FA[= ( )]

r1 t

1  pr  2pfD = 1 − (f /fD 2 )  0 else


(6)

f ≤ fD

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2.18


12. (b) (ii) Most radio propagation models are derived using a combination of analytical and empirical methods. The empirical approach is based on fitting curves or analytical expressions that recreate a set of measured data. This has the advantage of implicitly taking into account all propagation factors, both known and unknown, through actual field measurements. However, the validity of an empirical model at transmission frequencies or environments other than those used to derive the model can only be established by additional measured data in the new environment at the required transmission frequency. Over time, some classical propagation models have emerged, which are now used to predict large-scale coverage for mobile communication systems design. By using path loss models to estimate the received signal level as a function of distance, it becomes possible to predict the SNR for a mobile communication system 13. (a) Refer 13. (b) (i) from Nov/Dec 2013 13. (b) (i) Decision Circuit

LPF

Received Signal

BPF

Carrier Recovery

Symbol

Recovered

Timing

Circuit

Multiplex

Signal

Recovery

90°

LPF

Decision Circuit

Figure 1 Block diagram of a QPSK receiver

Figure 1 shows a block diagram of a coherent QPSK receiver. The frontend bandpass filter removes the out-of-band noise and adjacent channel interference. The filtered output is split into two parts, and each part is coherently demodulated using the in-phase and quadrature carriers. The coherent carriers used for demodulation are recovered from the received signal using carrier recovery circuits of the type described in Figure 1. The outputs of the demodulators are passed through decision circuits which generate the in-phase and quadrature binary streams. The two components are then multiplexed to reproduce the srcinal binary sequence.


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2.19

Offset QPSK The amplitude of a QPSK signal is ideally constant. However, when QPSK signals are pulse shaped, they lose the constant envelope property. The occasional phase shift of p radians can cause the signal envelope to pass through zero for just an instant. Any kind of hardlimiting or nonlinear amplification of the zero-crossings brings back the filtered sidelobes since the fidelity of the signal at small voltage levels is lost in transmission. To prevent the regeneration of sidelobes and spectral widening, it is imperative that QPSK signals that use pulse shaping be amplified only using linear amplifiers, which are less efficient. Amodified form of QPSK, called offset QPSK (OQPSK) or staggered QPSK is less susceptible to these deleterious effects [Pas79] and supports more efficient amplification. That is, OQPSK ensures there are fewer baseband signal transitions applied to the RF amplifier, which helps eliminate spectrum regrowth after amplification. OQPSK signaling is similar to QPSK signaling, as represented by Equation (6.77), except for the time alignment of the even and odd bit streams. In QPSK signaling, the bit transitions of the even and odd bit

streams occur at the same time instants, but in OQPSK signaling, the even and odd bit streams, and mI(t) and mQ(t), are offset in their relative alignment by one bit period (half-symbol period). This is shown in the waveforms of Figure 2.

Figure 2 The time offset waveforms that are applied to the in-phase and quadrature arms of an OQPSK modulator. Notice that a half-symbol offset is used.


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2.20


Due to the time alignment of mI (t) and mQ (t) in standard QPSK, phase transitions occur only once every Ts = 2 Tbs, and will be a maximum of 180° if there is a change in the value of both mI (t) and mQ (t). However, in OQPSK signaling, bit transitions (and, hence, phase transitions) occur every Tb s. Since the transition instants of mI (t) and mQ (t) are offset, at any given time only one of the two bit streams can change values. This implies that the maximum phase shift of the transmitted signal at any given time is limited to ±90°. Hence, by switching phases more frequently (i.e., every Tb s instead of 2 Tb s) OQPSK signaling eliminates 180°phase transitions. Since 180 ° phase transitions have been eliminated, bandlimiting of (i.e., pulse shaping) OQPSK signals does not cause the signal envelope to go to zero. Obviously, there will be some amount of ISI caused by the bandlimiting process, especially at the 90 ° phase transition points. But the envelope variations are considerably less, and hence hardlimiting or nonlinear amplification of OQPSK signals does not regenerate the high frequency sidelobes as much as in QPSK. Thus, spectral occupancy is significantly reduced, while permitting more efficient RF amplification. The spectrum of an OQPSK signal is identical to that of a QPSK signal, hence both signals occupy the same bandwidth. The staggered alignment of the even and odd bit streams does not change the nature of the spectrum. OQPSK retains its bandlimited nature even after nonlinear amplification, and therefore is very attractive for mobile communication systems where bandwidth efficiency and efficient nonlinear amplifiers are critical for low power drain. Further, OQPSK signals also appear to perform better than QPSK in the presence of phase jitter due to noisy reference signals at the receiver [Chu87]. 13. (b) (ii)

Figure 1 Block diagram of an MSK receiver


4/25/2014 9:38:16PM


2.21

Figure 2 Power spectral density of a GMSK signal [from [Mur81] © IEEE]

Figure 3 Block diagram of a GMSK transmitter using direct FM generation.

Figure 4 Block diagram of a GMSK receiver


4/25/2014 9:38:17PM

2.22


14. (a) (i)

Figure 1 Graph of probability distributions of SNR =g threshold for M branch selection diversity. The term Γ represents the mean SNR on each branch [from [Jak71] © IEEE]

14. (a) (ii) Decision Feedback Equalization (DFE) The basic idea behind decision feedback equalization is that once an information symbol has been detected and decided upon, the ISI that it induces on future symbols can be estimated and subtracted out before detection of subsequent symbols [Pro89]. The DFE can be realized in either the direct transversal form or as a lattice filter. The direct form is shown in Figure 1. It consists of a feed forward filter (FFF) and a feedback filter (FBF). The FBF is driven by decisions on the output of the detector, and its coefficients can be adjusted to cancel the ISI on the current symbol from past detected symbols. The equalizer has N1 + N2 + 1 taps in the feed forward filter and N3 taps in the feedback filter, and its output can be expressed as: N2

dˆ k =

∑Cy nnk n = −N1


N3 *

F d+ ∑ −

i

k −i

(1)

i =1

4/25/2014 9:38:17PM


2.23

Figure 1 Decision feedback equalizer (DFE) *

where Cn and yn are tap gains and the inputs, respectively, to the forward filter, Fi are tap gains for the feedback filter, and di(i < k) is the previous decision made on the detected signal. That is, once d k is obtained using Equation (1), dk is decided from it. Then, dk along with previous decisions dk-1, dk-2, …. are fed back into the equalizer, and dˆ k 1 is obtained using Equation (1). The minimum mean squared error a DFE can achieve is [Pro89] *

ˆ

+

E [| e( )|n ] 2

exp = min

 T p /T  ln∫  2p -p /T

N0     | F( e jwT)| 2 + N  dw    0 

(2)

It can be shown that the minimum MSE for a DFE in Equation (2) is always smaller than that of an LTE in Equation (2) unless F (ejwT) is a constant (i.e., when adaptive equalization is not needed) [Pro89]. If there are nulls in |F (ejwT)|, a DFE has significantly smaller minimum MSE than an LTE. Therefore, an LTE is well behaved when the channel spectrum is comparatively flat, but if the channel is severely distorted or exhibits nulls in the spectrum, the performance of an LTE deteriorates and the mean squared error of a DFE is much better than a LTE. Also, an LTE has difficulty equalizing a nonminimum phase channel, where the strongest energy arrives after the first arriving signal component. Thus, a DFE is more appropriate for severely distorted wireless channels.


4/25/2014 9:38:18PM

2.24


The lattice implementation of the DFE is equivalent to a transversal DFE having a feed forward filter of length N1 and a feedback filter of length N2, where N1 > N2. Another form of DFE proposed by Belfiore and Park [Bel79] is called a predictive DFE, and is shown in Figure. It also consists of a feed forward filter (FFF) as in the conventional DFE. However, the feedback filter (FBF) is driven by an input sequence formed by the difference of the output of the detector and the output of the feed forward filter. Hence, the FBF here is called a noise predicator because it predicts the noise and the residual ISI contained in the signal at the FFF output and subtracts from it the detector output after some feedback delay. The predictive DFE performs as well as the conventional DFE as the limit in the number of taps in the FFF and the FBF approach infinity. The FBF in the predictive DFE can also be realized as a lattice structure [Zho90]. The RLS lattice algorithm can be used in this case to yield fast convergence. (b) (i) Block Codes and Finite Fields 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. 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 [Lin83]. The block code is referred to as an ( n, k) code, and the rate of the code is defined as Rc = k/n and is equal to the rate of information divided by the raw channel rate. The ability of a block code to correct errors is a function of the code distance. Many families of codes exist that provide varying degrees of error protection [Cou93], [Hay94], [Lin83], [Skl93], and [Vit79]. Besides the code rate, other important parameters are the code distance and the weight of particular codewords. These are defined below. Distance of a Code — The distance between two codewords is the number of elements in which two codewords Ci and Cj differ N

( Ci , j dC

) =Cl∑ C, ⊕ i

(mod ulo q

j ,l

)

(1)

l =1


4/25/2014 9:38:19PM


2.25

where d is the distance between the codewords and q is the total number of possible values of Ci and Cj. The length of each codeword is N elements or characters. If the code used is binary, the distance is known as the Hamming distance. The minimum distance dmin is the smallest distance for the given codeword set and is given as

dmin

=

Min{ d( C, Ci )} j

(2)

Weight of a Code — The weight of a codeword of length N is given by the number of nonzero elements in the codeword. For a binary code, the weight is basically the number of 1s in the codeword and is given as N

w (C i )

=

∑C ,

(3)

i l

l =1

Properties of Block Codes Linearity — Suppose Ci and Cj are two codewords in an (n, k) block code. Let a1and a2 be any two elements selected from the alphabet. Then the code is said to be linear if and only if a1C1 + a2C2 is also a code word. A linear code must contain the all-zero code word. Consequently, a constant-weight code is nonlinear. Systematic — A systematic code is one in which the parity bits are appended to the end of the information bits. For an ( n, k) code, the first k bits are identical to the information bits, and the remaining n-k bits of each code word are linear combinations of the k information bits. Cyclic — Cyclic codes are a subset of the class of linear codes which satisfy the following cyclic shift property: If C = [Cn-1, Cn-2, ……, C0] is a codeword 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. As a consequence of the cyclic property, the codes possess a considerable amount of structure which can be exploited

to greatly simplify the encoding and decoding operations. 14. (b) (ii) Refer Q. no. 14 (b) from Nov/Dec 2013 15. (a) The number of simultaneous users that can be accommodated in GSM is given as N

25 MHz =

( 200 KHz)/8

=

1000

Thus, GSM can accommodate 1000 simultaneous users.


4/25/2014 9:38:19PM

2.26


1 = 3.692 µs. 270.833 kbps (b) The time duration of a slot, Tslot = 156.25 × Tb = 0.577 ms. (c) The time duration of a frame, T = 8 × Tslot = 4.615 ms. (a) The time duration of a bit, Tb =

f

(d) A user has to wait 4.615 ms, the arrival time of a new frame, for its next transmission A time slot has 6 + 8.25 + 26 + 2(58) = 156.25 bits (a) Number of overhead bits, boh = 8(6) + 8(8.25) + 8(26) = 322 bits (b) Number of bits/frame = 8 × 156.25 = 1250 bits/frame (c) Frame rate: 270.833 kbps/1250 bits/frame = 216.66 frame/sec (d) Time duration of a slot = 156.25 × 1/270.833 kbps = 576.92 µs (e) Frame efficiency = hf = [1 – (322/1250)] = 74.24% Spread Spectrum Multiple Access

Spread Spectrum Multiple Access (SSMA) uses signals which have a transmission bandwidth that is several orders of magnitude greater than the minimum required RF bandwidth. A pseudo-noise (PN) sequence (discussed in Chapter 6) converts a narrowband signal to a wideband noise-like signal before transmission. SSMA also provides immunity to multipath interference and robust multiple access capability. SSMA is not very bandwidth efficient when used by a single user. However, since many users can share the same spread spectrum bandwidth without interfering with one another, spread spectrum systems become bandwidth efficient in a multiple user environment. It is exactly this situation that is of interest to wireless system designers. There are two main types of spread spectrum multiple access techniques;frequency hopped Multiple Access (FH) and direct sequence multiple access (DS). Direct sequence multiple access is also called code division multiple access(CDMA). Frequency Hopped Multiple Access (FHMA) (FHMA) is a digital multiple

Frequency Hopped Multiple Access

access system in which the carrier frequencies of the individual users are varied in a pseudorandom fashion within a wideband channel. Figure 1 illustrates how FHMA allows multiple users to simultaneously occupy the same spectrum at the same time, where each user dwells at a specific narrowband channel at a particular instance of time, based on the particular PN code of the user. The digital data of each user is broken into uniform sized bursts which are transmitted on different channels within the allocated spectrum band. The instantaneous bandwidth of any one transmission burst is much smaller than the total spread bandwidth. The pseudorandom change of the channel frequencies of the


4/25/2014 9:38:20PM


2.27

Figure 1 Spread spectrum multiple access in which each channel is assigned a unique PN code which is orthogonal or approximately orthogonal to PN codes used by other users

user randomizes the occupancy of a specific channel at any given time, thereby allowing for multiple access over a wide range of frequencies. In the FH receiver, a locally generated PN code is used to synchronize the receiver’s instantaneous frequency with that of the transmitter. At any given point in time, a frequency hopped signal only occupies a single, relatively narrow channel since narrowband FM or FSK is used. The difference between FHMA and a traditional FDMA system is that the frequency hopped signal changes channels at rapid intervals. If the rate of change of the carrier frequency is greater than the symbol rate, then the system is referred to as a fast frequency hopping system. If the channel changes at a rate less than or equal to the symbol rate, it is called

slow frequency hopping. A fast frequency hopper may thus be thought

of as an FDMA system which employs frequency diversity. FHMA systems often employ energy efficient constant envelope modulation. Inexpensive receivers may be built to provide noncoherent detection of FHMA. This implies that linearity is not an issue, and the power of multiple users at the receiver does not degrade FHMA performance. A frequency hopped system provides a level of security, especially when a large number of channels are used, since an unintended (or an intercepting) receiver that does not know the pseudorandom sequence of frequency slots must retune rapidly to search for the signal it wishes to


4/25/2014 9:38:20PM

2.28


intercept. In addition, the FH signal is somewhat immune to fading, since error control coding and interleaving can be used to protect the frequency hopped signal against deep fades which may occasionally occur during the hopping sequence. Error control coding and interleaving can also be combined to guard against erasures which can occur when two or more users transmit on the same channel at the same time. Bluetooth and HomeRF wireless technologies have adopted FHMA for power efficiency and low cost implementation.


4/25/2014 9:38:20PM

B.E./B.Tech. DEGREE EXAMINATION, NOVEMBER/DECEMBER 2011 (Unsolved Question Paper) Seventh Semester

Electronics and Communication Engineering WIRELESS COMMUNICATION Time: Three hours

Maximum: 100 marks

Answer ALL questions PART A (10 × 2 = 20 marks) 1. Differentiate Cellular telephony and Cordless telephony. 2. When does a WLAN become a personal Area Network (PAN)? 3. Compute the Rayleigh distance of a square Antenna with 20 dB gain. 4. List out any two properties of wideband channel. 5. Draw the mathematical link model for analysis of modulation schemes. 6. What is OQPSK? 7. Mention any four common methods of micro diversity. 8. Define Hamming distance and Euclidean distance between two codes. 9. Discuss the principle of OFDM modulation scheme. 10. Give three important functional blocks of GSM system.

PART B (5 × 16

=

80 marks)

11. (a) (i) Compare and contrast wired and wireless communication. (ii) Discuss briefly about the requirements of services for a wireless system.

WC_Question_Nov-Dec 2011.indd 3

(8) (8)

7/26/2012 4:38:11 PM

3.4


Or

(b) (i) Discuss in detail the constructive and destructive interference.

(8)

(ii) Explain how inter symbol interference is caused and how it is eliminated.

(8)

12. (a) (i) Describe any two methods of diffraction by multiple screens. (ii) Discuss about ultra wide band channel.

(8) (8)

Or

(b) (i) Compare Coherence Bandwidth and Coherence time

(8)

(ii) Discuss the mathematical formulation for narrowband and wideband system, with relevant figures.

(8)

13. (a) Compare the ratio of signal power to adjacent channel interference when using (i) raised cosine pulses (ii) root raised cosine pulses with a = 0.5, when two considered signals have center frequencies 0 and 1.25/T.

(16)

Or

(b) (i) Discuss in detail any two demodulation techniques of minimum shift keying method.

(8)

(ii) Explain in detail about optimum receiver structure for Non-coherent detection.

(8)

14. (a) (i) Explain the Viterbi decoding scheme if the decoder input sequence is ‘010 000 100 001 011 110 001’.

(16)

Or (b) (i) With a neat block diagram discuss the structure of a decision feedback equalizer. (ii) Discuss linear predictive vocoder with block diagram.


(8) (8)

7/26/2012 4:38:11 PM


3.5

15. (a) (i) Explain the principle of direct sequence spread spectrum technique.

(8)

(ii) Discuss some methods to increase the capacity of wireless communication system.

(8)

Or (b) (i) Explain in detail about the GSM logical channels. (ii) Explain the block diagram of IS-95 transmitter.


(8) (8)

7/26/2012 4:38:11 PM

B.E./B.Tech. DEGREE EXAMINATION, APRIL/MAY 2011 (Unsolved Question Paper) Seventh Semester


Maximum: 100 marks

Answer ALL questions PART A (10 × 2 = 20 marks) 1. What is the consequence of increased link performance in CDMA systems? 2. Define signal to interface ratio in terms of distance between cells centres and radius of cell in a cellular scenario. 3. How does microzone concept reduce hard off? 4. Differentiate fast fading and slow fading with respect to coherence time and doppler spread. 5. Specify the type of model that can be used to predict path loss for GSM frequency range. 6. What is the probability of error (BER) in QPSK modulation systems? Comment on the performance. 7. Write the DPSK sequence for the data stream 110110. 8. Differentiate between PCM and DPCM with respect to error performance, which source coding is suitable for speech signals. 9. What type of channel coding is used for QSM network? 10. Give the specification of ideal uplink and downlink characteristics of W-CDMA.

WC_Question_Apr-May 2011.indd 6

7/26/2012 4:37:20 PM

Wireless Communication (Apr/May 2011)

PART B (5 × 16

=

3.7

80 marks)

11. (a) Explain the different technique to reduce CCI and ACI in a seven cell reuse cellular system. Derive the signal to interference for the worst conditions. Also explain the effect of pathloss on interference

(16)

Or

(b) (i) Derive the probability of blocking in a trusted Lost Call Cleared (LCC) and Last Call Delayed (LCD) systems.

(6)

(ii) In a cellular system there are 12 clusters available with 120 cells with 10,000 subscribers. Each subcribers uses the cell phone for one hour everyday. On an average 30 minutes of time is used during peak hours. Find the average and peak traffic for the entire cellular system and also for one cell considering all callers are distributed evenly over the system.

(10)

12. (a) (i) What are the different indoor propagation models available for path loss calculation? Explain the log-distance path loss model(s).

(8)

(ii) Calculate the propagation loss for a ratio signal at 850 mHz. The transmitter antenna height is 30 m over a distance of 11 km. Use mobile propagation model and free space model and compare them.

(8)

Or

(b) (i) Explain the different statistical models available for multipath fading channels (ii) Explain the fading effects due to Doppler spread.

(8) (8)

13. (a) How does maximal ratio combining techniques combat deep fades in wireless environment? Derive the expression for the maximum gain of the transmitting antennas.

(16)

Or

(b) Derive the BER for MSK with respect to symbol energy and symbol time period. Why MSK is preferred in mobile


7/26/2012 4:37:20 PM

3.8


environment. Draw the spectral characteristics of the MSK modulation scheme.

(16)

14. (a) (i) Derive the general expression that relates SNR due to quantisation as a function of number of bits. (ii) Explain the different type of vocoders.

(8) (8)

Or (b) Derive the radio capacity for FDMA in terms of path loss, total allocated spectrum, channel Bandwidth and Carrier to interface (CLP) ratio. Compare with capacity of cellular CDMA.

(16)

15. (a) Explain the functions of DECT. What are the various entities? List the specifications for DECT Radio.

(16)

Or (b) What are the various GSM control channels? Explain the various control channels by srcinal a cellular call through the GSM network. Draw the timing diagrams. (16)


7/26/2012 4:37:20 PM

B.E./B.Tech. DEGREE EXAMINATION, NOVEMBER/DECEMBER 2010 (Unsolved Question Paper) Seventh Semester


Maximum: 100 marks

Answer ALL questions PART A (10 × 2 = 20 marks) 1. Draw the frequency management allocation for GSM network. 2. What is meant by grade service? 3. What are the data rates for the cordless phone? 4. If S/T is a cellular network in 18 dB, what is the frequency reuse factor and cluster size for a path loss component of 3? 5. What are the various methods for improving the capacity of cellular networks? 6. Define large scale propagation? 7. Why is QPSK preferred for wireless communication? 8. What is the need for diversity in multipath propagation? 9. Find the number of channels available in a FDMA system, if the cellular network is allocated 12.5 mHz for each simplex band, total bandwidth is 12.5 mHz guard bandwidth is 10 kHz and channel bandwidth is 30 kHZ. 10. Draw the frame structure for the GSM network.


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3.10


PART B (5 × 16

=

80 marks)

11. (a) Derive the tracking capacity in a cellular network which employ cell splitting and cell sectoring techniques comes on the improvement in CCI and ACI.

(16)

Or

(b) Prove that the frequency reuse ratio is 3 N . Where ‘N’ is the size of the cluster. Also derive the worst case S/I ratio as 17 dB for a reuse ratio of 4.6.

(16)

12. (a) Derive the path loss for small scale propagation and large scale propagation in a multipath wireless environment. What is Doppler spread? (16) Or

(b) Derive any one of the path loss models for outdoor environment and explain how it may be used to obtain the path loss in urban and semi urban areas.

(16)

13. (a) What are the different digital modulation techniques employed in wireless communication? Derive the BER for BPSK digital modulation scheme.

(16)

Or

(b) What is the necessity of diversity techniques? Explain space and time diversity and how it can reduce fading in a multipath propagation.

(16)

14. (a) Explain PCM and ADPCM and how it is utilized for encoding and decoding in wireless networks.

(16)

Or (b) What are the multiple access techniques used in AMPS, GSM and CDMA networks? Explain how capacity is improved using special spectrum techniques.


(10)

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3.11

15. (a) Draw the architecture of the GSM and explain the different air channels.

(16)

Or (b) Compare the operational features of WLL and Bluetooth.


(16)

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B.E./B.Tech. DEGREE EXAMINATION, APRIL/MAY 2010 (Unsolved Question Paper) Seventh Semester Electronics and Communication Engineering WIRELESS COMMUNICATION Time: Three hours

Maximum: 100 marks

Answer ALL questions PART A (10 × 2 = 20 marks) 1. List any four advantages of third generation (3G) mobile networks. 2. Write down the formula for co-channel reuse ratio. 3. What is doppler shift? 4. What is meant by coherence bandwidth? 5. Name the four space diversity reception techniques? 6. Define absolute bandwidth. 7. What is meant by vocoders? 8. What is self-jamming in CDMA? 9. What is the purpose of SIM? 10. What are the five functional entities of a DECT system?

PART B (5 × 16

=

80 marks)

11. (a) (i) Explain in detail the various cellular components. (ii) Explain the fundamentals of Digital cellular systems.

(8) (8)

Or


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Wireless Communication (Apr/May 2010)

3.13

(b) (i) Explain the process of operation of paging systems.

(8)

(ii) Describe the steps involved in making a cellular telephone call.

(8)

12. (a) Derive the impulse response model of a multipath channel.

(16)

Or

(b) Explain in detail about type of small scale fading.

(16)

13. (a) (i) Write a note on RAKE Receiver, with block diagram.

(10)

(ii) Explain frequency diversity and time diversity.

(6)

Or

(b) Explain the concept of minimum shift keying and Gaussian MSK.

(16)

14. (a) Explain TDMA and discuss the time division multiple access frame structure.

(16)

Or (b) (i) Draw the block diagram of RELP encoder. Explain the function of coder.

(6)

(ii) Enumerate the procedure for selecting the optimum excitation signal.

(10)

15. (a) (i) Discuss the features and services of GSM. (ii) Explain the GSM system architecture with neat sketch.

(8) (8)

Or (b) Discuss the channels and cellular operation in advanced mobile phone systems (AMPS). (16)


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7/27/2012 4:43:59 PM

Wireless Communication.pdf

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