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CS3282 Digital Communications Examination June'05 \(complete\)

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CS3282 Exam June 2005 1 25/08/2006 BMGC CS3282 Digital Communications Examination June’05 (complete) Electronic calculators ALLOWED. Answer Three questions in TWO hours. Attached: Appendix 1: Graph of complementary error function Q(z) against z. Appendix 2: Fourier transform properties Appendix 3: Trigonometric formulae ------- ------- ------ ------ ----- ----- ---- ----- ---- ---- ---- --- 1. (a) Explain the term 'spread-spectrum multiple access' (SSMA), name two commonly used SSMA techniques and briefly explain how they work.. [8 marks] (b) Explain how the multi-carrier modulation scheme known as 'orthogonal frequency division multiplexing' (OFDM) is employed in IEEE802.11 wireless LAN standards and show how the modulation is achieved using an inverse FFT. [6 marks] (c) Explain the purpose of the cyclic extension in OFDM and state the advantages and disadvantages of OFDM as compared to single carrier modulation for mobile computer networks. [6 marks] 2.(a) State the Shannon-Hartley Theorem with an explanation of its significance. Show that, according to this theorem, arbitrarily low bit-error rates cannot be achieved from a digital channel if E b /N 0 is less than about -1.6 dB where E b is the average energy per bit and N 0 /2 is the two-sided power spectral density of additive white Gaussian noise. [8 marks] According to the Shannon-Hartley Theorem, what is the maximum bit-rate that can be transmitted and received with arbitrarily small bit-error rates over a 20 MHz channel with signalling giving a signal-to- noise ratio of 20 dB? [2 marks] (b) What is meant by inter-symbol interference in a digital transmission system and how does it arise? [2 marks] An appropriately shaped symbol centred on t = 0 with zero-crossings at t = ±T, ±2T, etc. is distorted by the frequency response of the channel such that the received voltages x(t) at t = 0, ±T and ±2T are expected to be as follows in the absence of channel noise : x(-2T) = -0.1, x(-T) = 0.1, x(0) = 1, x(T) = -0.1, x(2T) = 0.1 Inter-symbol interference is to be reduced by a 3–term ”zero-forcing” transversal equaliser. Explain how this may be achieved and give a diagram of the equaliser. [3 marks] Given that the zero-forcing equaliser has coefficients C 0 , C 1 and C 2 , expressed as a column-vector c , show that the required zero-crossings are achieved if Ac = b where = 1 1 . 0 1 . 0 1 . 0 1 1 . 0 1 . 0 1 . 0 1 A and b is a suitably chosen column-vector. [2 marks] Given that the inverse of matrix A is approximately as follows: = 98 . 0 107 . 0 087 . 0 087 . 98 . 0 107 . 0 107 . 0 087 . 0 98 . 0 A 1 calculate the coefficients C 0 , C 1 , and C 2 . [1 mark] Give an expression for the frequency response of the equaliser and comment on how it will affect any additive white Gaussian channel noise. [2 marks]
Transcript
Page 1: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 1 25/08/2006 BMGC CS3282 Digital Communications Examination June’05 (complete) Electronic calculators ALLOWED. Answer Three questions in TWO hours. Attached: Appendix 1: Graph of complementary error function Q(z) against z. Appendix 2: Fourier transform properties Appendix 3: Trigonometric formulae ------- ------- ------ ------ ----- ----- ---- ----- ---- ---- ---- --- 1. (a) Explain the term 'spread-spectrum multiple access' (SSMA), name two commonly used SSMA techniques and briefly explain how they work.. [8 marks] (b) Explain how the multi-carrier modulation scheme known as 'orthogonal frequency division multiplexing' (OFDM) is employed in IEEE802.11 wireless LAN standards and show how the modulation is achieved using an inverse FFT. [6 marks] (c) Explain the purpose of the cyclic extension in OFDM and state the advantages and disadvantages of OFDM as compared to single carrier modulation for mobile computer networks. [6 marks] 2.(a) State the Shannon-Hartley Theorem with an explanation of its significance. Show that, according to this theorem, arbitrarily low bit-error rates cannot be achieved from a digital channel if Eb/N0 is less than about -1.6 dB where Eb is the average energy per bit and N0 /2 is the two-sided power spectral density of additive white Gaussian noise. [8 marks] According to the Shannon-Hartley Theorem, what is the maximum bit-rate that can be transmitted and received with arbitrarily small bit-error rates over a 20 MHz channel with signalling giving a signal-to-noise ratio of 20 dB? [2 marks] (b) What is meant by inter-symbol interference in a digital transmission system and how does it arise?

[2 marks] An appropriately shaped symbol centred on t = 0 with zero-crossings at t = ±T, ±2T, etc. is distorted by the frequency response of the channel such that the received voltages x(t) at t = 0, ±T and ±2T are expected to be as follows in the absence of channel noise :

x(-2T) = -0.1, x(-T) = 0.1, x(0) = 1, x(T) = -0.1, x(2T) = 0.1 Inter-symbol interference is to be reduced by a 3–term ”zero-forcing” transversal equaliser. Explain how this may be achieved and give a diagram of the equaliser. [3 marks] Given that the zero-forcing equaliser has coefficients C0, C1 and C2 , expressed as a column-vector c, show that the required zero-crossings are achieved if Ac = b where

−−

−=

11.01.01.011.01.01.01

A

and b is a suitably chosen column-vector. [2 marks] Given that the inverse of matrix A is approximately as follows:

−−

−=−

98.0107.0087.0087.98.0107.0107.0087.098.0

A 1

calculate the coefficients C0, C1, and C2. [1 mark] Give an expression for the frequency response of the equaliser and comment on how it will affect any additive white Gaussian channel noise. [2 marks]

Page 2: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 2 25/08/2006 BMGC 3(a) Define what is meant by the ‘dynamic range’ of a speech or music quantiser. Why is pseudo-logarithmic companding (A- or mu-law) used when telephone speech is digitised for transmission by pulse-code modulation and how does it increase the dynamic range, as compared with uniform quantisation, without increasing the bit-rate? [7 marks] (b) Monophonic music is to be sampled at 40 kHz, uniformly quantised and transmitted over a 450 kHz bandwidth channel using QPSK with 50% RRC symbols. Calculate the number of bits per sample that may be transmitted and hence estimate the maximum achievable SQNR assuming music to be approximately sinusoidal. Calculate the bandwidth efficiency of the transmission scheme. If the minimum acceptable SQNR is 40 dB, what is the dynamic range of the quantiser? [7 marks] (c) Explain what is meant by coherent detection and state its advantages and disadvantages with respect

to non-coherent detection. Show how a QPSK transmission may be generated using a vector modulator and detected coherently at the receiver. [6 marks]

4. Assuming the Cauchy-Schartz inequality:

∫∫∫ ≤b

a

b

a

b

aduuyduuxduuyux 222

)()()()(

show that if binary signalling at 1/T Baud with equally likely symbols s1(t) and s2(t) is received from a channel affected by additive white Gaussian noise, a matched filter with impulse-response proportional to s1(T-t) - s2(T-t) with an appropriate choice of threshold will minimise the bit-error rate produced by a threshold detector. [8 marks] If the two-sided power spectral density (PSD) of the noise is N0/2 Watts/Hz, and the average energy of the received difference signal s1(t) - s2(t) is Ed Joules, show that the minimised error probability is PB = Q( )2/( 0NEd ). [6 marks] (b) A receiver receives 2 volt and -2 volt NRZ rectangular binary symbols. The transmission is distorted by additive white Gaussian noise with two-sided PSD N0/2 = 1 x 10-3 Watts/Hz. If the received signal is detected with the aid of a matched filter, what is the maximum bit-rate that can be sent with a bit-error rate of less than 1 bit in 1000. [6 marks] A graph of the “complementary error function” Q(z) against z is attached to this examination paper.

Page 3: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 3 25/08/2006 BMGC 5.(a) Define what is meant by:

(i) binary frequency shift keying (binary FSK) [2 marks] (ii) Gaussian minimum shift keying (GMSK) [2 marks]

What are the main advantages of GMSK that led to its adoption for second generation GSM mobile telephones? Show how GMSK may be generated. [3 marks] (b) Why is Gray coding used in multi-level digital modulation schemes? [2 marks] The constellation diagrams for an ‘8-APK’ modulator and an ‘8-PSK’ modulator are shown below:

100

101

111

110

010

011

001

000

R

2A

-2A

-A

A

-2A -A 2A A

111

011

101

110 010 100

‘8-PSK’ ‘8-APK’

Q

I

Q

I

001

000

Assuming all symbols are equally likely, calculate the average energy per bit, Eb, for each of these modulation schemes when the symbol rate is 1 kbaud. Show that if R ≈ 1.58A , the average energy per bit is the same for both schemes. [5 marks] Symbol detection is achieved by choosing the nearest of the eight symbols on the constellation diagram to a received symbol and therefore the minimum distance between symbols is taken as a measure of noise immunity. Which of the two schemes has the greater noise immunity for a given Eb/N0? [2 marks] Sketch the waveform produced by each of the schemes above when the 3 kb/s bit-stream is as follows: 0 0 0 1 1 0 0 1 1 1 0 1 and the carrier frequency is 2 kHz. The effect of band-limiting and pulse-shaping need not be shown. [4 marks]

Page 4: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 4 25/08/2006 BMGC

Graph of Complementary Error function, ∫∞

−=

zdttzQ

2exp

21)(

2

π

Page 5: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 5 25/08/2006 BMGC Appendix 2: Properties of the Fourier Transform (1/2/03)

Property Signal x(t) Fourier Transform X(f)

Transform & inverse: x(t) = X(f) = ∫∞

∞−dfefX jftπ2)( ∫

∞−

− dtetx jftπ2)(

Similarly let the Fourier transforms of y(t), y1(t) & y2(t) be Y(f), Y1(f) & Y2(f) respectively. Rect & sinc rectA(t) = rect(t/A) Asinc1/A(f)=A sinc(Af) sincA(t) = sinc(t/A) Arect1/A(f) = A rect(Af) Delay y(t-τ) e-2πjτY(f) Frequency shift e2πjFt y(t) Y(f-F) Amplitude scaling: Ay(t) AY(f) Time-reversal: y(-t) Y(-f) Superposition: Ay1(t)+By2(t) AY1(f)+BY2(f) Constant: A Aδ(f) Impulse: Aδ(t) A Gaussian ( π /α) exp(−π2 t2 / α2) exp(−α2 f 2 ) Time-scaling: y(At) (1 / |A|) Y(f/A) Differentiation: dmy(t)/dtm (2πjf)m Y(f) Product: y1(t)y2(t) Y1(f) ⊗ Y2(f) = Y Y f d1 2( ) ( )θ θ−

−∞

∫ θ

τ

τ

τ

Convolution: Yy y t d1 2( ) ( )τ τ−−∞

∫ 1(f) Y2(f)

Cross-correlation: Yy y t d1 2( ) ( )τ τ+−∞

∫ 1(f) Y2*(f)

Auto-correlation: |Y(f)|y y t d( ) ( )τ τ+−∞

∫ 2

Repeat: repeatPy(t) (1/P)sample1/PY(f) Sample: sampleTy(t) (1/T)repeat1/TY(f) Properties for real signals:- For all real signals: x*(t)=x(t) X(−f) = X*(f) i.e. |X(−f)| = |X(f)| & φ(−f) = −φ(f) Real and even: x(t) = x(−t) X(f) is purely real & X(−f) = X(f) Real and odd: x(t) = −x(−t) X(f) is purely imaginary & X(−f) = −X(f) Formulae:-

sinc

rect( ) =1 :| |< 0.50.5:| |= 0.5

sinc ( ) sinc( / A) rect ( ) rect( / AA A( )

sin( )( ) :

: :| | .)x

xx x

xx

xxx

x x x x= ≠=

>

= =π

π 01 0 0 05

repeat sample repeatT TPn n

x t x t nP x t x nT t nT x t t ( ) ( ) ( ) ( ) ( ) ( ) ( )= − = − ==−∞

=−∞

∑ ∑ δ δ

Fourier series for repeatPx(t):-

( )C e M k P t A A k P t B k P tkjkt P

k k k kkkk

20

102 2π π θ π π/ cos(( / ) ) cos(( / ) ) sin(( / ) )= + = + +

=

=

=−∞

∑∑∑ 2

k k

C P X k P M C M C C A C A C B Ck k k k k k k= = = = = = =( / ) ( / ) ; | |; | |; arg( ); ; Re ; Im( 1 2 20 0 0 0 2θ

Page 6: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 6 25/08/2006 BMGC Appendix 3: Trigonometric formulae

sin(A ± B) = sin(A) cos(B) ± cos(A)sin(B) cos(A ± B) = cos(A) cos(B) sin(A)sin(B) mtan(A ± B) = ( tan(A) ± tan(B) ) / (1 m tan(A)tan(B) ) sin(2A) = 2 sin(A) cos(A) cos(2A) = 2 cos2(A) – 1 = 1 – 2 sin2(A) tan (2A) = 2 tan(A) / (1 – tan2(A) ) 2cos(A) cos(B) = cos(A + B) + cos(A - B) 2 sin(A) cos(B) = sin(A + B) + sin(A - B) 2 sin(A) sin(B) = cos(A - B) – cos(A + B) cos(A) + cos(B) = 2cos( (A + B)/2 ) cos( (A – B)/2 ) sin(A) + cos(B) = sin(A) + sin(B+π/2) sin(A) + sin(B) = 2sin( (A + B)/2 ) cos( (A - B)/2 ) cos(θ) = (ejθ + e-jθ) / 2 sin(θ) = (ejθ - e-jθ) / (2j) λ cos(θ) + µ sin(θ) = R cos (θ + φ) where R2 = λ2 + µ2 and φ = tan-1(µ/λ) + π if λ<0

Page 7: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 7 25/08/2006 BMGC Solution methods

Question 1(a): (c) SSMA techniques use transmitters which spread the information over a bandwidth several orders of magnitude wider than would be needed with ordinary PSK, FSK or ASK. While this may seem inefficient, it is done in such a way that many users can transmit simultaneously over the same very wide frequency band, their transmissions being separable at a receiver. There are two main types of SSMA: "frequency hopped" (FH) and "direct sequence" (DS). The latter is also known as "code division multiple access" (CDMA). [2] FH-MA ("frequency hopping multiple access") can be applied to transmission schemes such as PSK by varying the carrier frequencies in a pseudo-random fashion within a wide-band channel. The data to be transmitted is split into blocks each of equal duration, and each is transmitted and received with a different carrier frequency. [1] The choice of frequencies must be made according to a known pseudo random sequence and synchronised at transmitter and receiver. FHMA provides security and also considerable immunity to fading since the effect of deep fades at particular frequencies will be spread out among all users each being degraded for just a short period of time (until he "hops" onto another carrier). The effect of the short duration degradation can be minimised by error coding or "diversity" transmissions. [2] CDMA : achieved by multiplying a base-band signal not by a sinusoidal carrier but instead by a very large bandwidth signal called a "spreading signal". The base-band signal represents the data stream to be transmitted ). The spreading signal is created in digital form by generating a pseudo-random sequence of bits at a very high sampling rate called the "chip-rate". The multiplication is done digitally where each data bit modulates about 50 pseudo random "chips". The stream of chips is transmitted as a very wide-band signal. [1] The receiver, knowing the chip sequence can recover each data bit by a cross-correlation process. [1] CDMA has the advantages of a "soft" capacity limit; i.e. as more and more users start to share the same bandwidth, the quality of signal received will degrade slowly. The effects of multi-path fading are substantially reduced because of the spreading. However power control is a difficulty with CDMA due to the well known "near-far" problem . [1]

(b) OFDM has many carrier frequencies evenly spaced out over the given bandwidth. IEEE802.11a takes 64 carrier frequencies over the range fC to fC + 20 MHz. fC + 0, fC + fD, fC + 2fD, … , fC +63fD Do the OFDM modulation in two stages: (1) Apply PSK, QPSK, QAM (or other) to 64 'sub-carriers' of frequencies: 0 , fD, 2fD , …, 63fD (2) Vector-modulate a carrier (eg. 5 GHz) with sum of modulated sub-carriers. [2] Let the 64 carriers be modulated by X0(t), X1(t), …. Adding these together we obtain: 63 x(t) = Σ Xk(t) exp (2πjkfD t ) : -∞<t<∞. [1] k=0 Sampling at 20MHz, (i.e. T = 0.05 us) this becomes: 63 x(nT) = x[n] = Σ Xk(nT) exp (2πjk fD nT ) k=0

Page 8: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 8 25/08/2006 BMGC Make Xk[nT] =Xk : constant for 0<n<79, i.e. 4 us. 63 x[n] = Σ Xk exp(jk(2π/N)n) : 0<n<79 [1] k=0 Generates a set x[0], x[1], …, x[79] of complex numbers which is time-domain OFDM "symbol" . The expression for x[n] is the "inverse FFT" of X0, X1, …., X63. [1] The real part of the symbol multiplies cos(2πfCt) & the imaginary part multiplies sin(2πfCt). Taking the real part of s(t)exp(-2πfCt), where s(t) is the sequence of OFDM symbols, would generate an OFDM signal starting at fC Hz rather than zero. [1] 1 (c) We calculate x[0], x[1], …, x[63], x[64], …, x[79], i.e. 80 time-domain samples rather than just 64. The set of time-domain samples x[64] to x[79] is the "cyclic extension". It may be thought of as providing a "guard interval" between one symbol and the next. It is useful for carrier and symbol synchronisation at receiver. Also, due to the cyclic extension & cyclic nature of DFT and its inverse, even if exact synchronisation is not achieved at the receiver, exact data can still be recovered with a phase shift. [2] OFDM is very good for channels affected by frequency selective fading for several reasons. (1) info can be spread out across sub-carriers so that when some are lost, others will compensate. (2) guard-band allows for ISI; if 4us OFDM symbol rings on, it only affects beginning of next symbol, repeated at end. (Can have cyclic "prefix"). So no pulse-shaping necessary! (3) equalisation much easier than with single carrier systems. OFDM equalisation done by multiplication in frequency-domain after FFT. Easier than adaptive filtering used for single carrier. Works because of cyclic extension. [2] Disadvantage of OFMD is "peak to mean" ratio of symbols which can be very large by nature of FFT & its inverse. Shape of each OFDM symbol is very complex & must be sent & received accurately. Amplitudes can become very large in comparison to the mean. Definitely not "constant envelope". Transmitter & receiver must be linear to preserve shape. Therefore, need ‘class A’ amplifiers: less power efficient than those for constant envelope transmissions. [2]

Page 9: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 9 25/08/2006 BMGC

Question 2(a) This theorem gives us a theoretical maximum bit-rate that can be transmitted with an arbitrarily small bit-error rate (BER), with a given average signal power, over a channel with bandwidth B Hz which is affected by AWGN. By “arbitrarily small BER” this means for any given BER, however small, we can find a coding technique that achieves this BER. The maximum achievable bit-rate (with arbitrary BER) is referred to as the channel capacity C. The Theorem states that:

ondbitsNSBC sec/1log 2

+=

where S/N is the mean-square signal to noise ratio (not in dB).

If we wish to transmit average energy/bit of Eb (Joules per bit) and the AWG noise has 2-sided power

spectral density N0 /2 Watts per Hz, the signal power S = EbR and the noise power N = N0B Watts.

Hence, by the S-H theorem:

+≤

02 1log

NEBR

BRb

R/B is called the bandwidth efficiency in units of bit/second/Hz.. We can now write:

0

12NEBR

b

BR +≤

which means that

BRNE

BR

b12

0−

i.e. BR

NEBR

b12)( min0

−≥

If we draw a graph of (Eb/N0 )min against R/B we see that (Eb/N0 )min never goes less than about 0.69

which is about -1.6 dB. Therefore if our normalised energy per bit is less than -1.6dB, we can never

satisfy the Shannon-Hartley law.

Shannon-H Limit

0.1

1

10

100

0.1 1 10B/W efficiency

(Eb/

No)

min

To see this mathematically, note that

BRee BRBRBR e 693.012 693.0)2(log +≈==

when R/B is small. Therefore when R/B is small,

Page 10: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 10 25/08/2006 BMGC

( ) dBBR

BRNEb 6.169.0

169.01min0 −==

−+≈

---------------------------------------- ---------------------------------------------------- To avoid calculating logs to the base 2,

+≈

+=

NSB

NSBC 1log32.31log

2log1

101010

This means that if S/N >>1, C ≈ 0.332 times B times the SNR in dB.

Hence if B=20 MHz and SNR = 20 dB, max bit rate = 0.332 x 20 x106 x 20 = 132.8 Mb/s ----------------------------------------- ------------------------------------- 2(b) Because of the channel’s finite bandwidth, the response to any symbol will not be time-limited. It will continue ringing forever, though dying away quite rapidly in amplitude. Inter-symbol interference (ISI) can occur due to the ringing of one symbol into the next. [2] The zero-forcing equaliser is as follows:

Delay T s

Delay T s x(t)

C0 C1 C2 y(t)

When the input is a single signalling pulse of the appropriate shape but distorted by the channel’s frequency response, and it is centred on t=0 as detected by the timing circuitry, the output must be forced to be zero at t=0 and t=2T. The output at t=T is arbitrary and may be taken to be 1. The output at t= 0, T and 2T is as follows: 0 = y(0) = x(0)C0 + x(-T)C1 + x(-2T) C2 1 = y(T) = x(T)C0 + x(0)C1 + x(-T) C2 0 = y(2T) = x(2T)C0 + x(T)C1 + x(0) C2 In matrix form:

−−

−=

2

1

0

11.01.01.011.01.1.01

010

CCC

b = A c (det A = 1.03) Therefore c = A-1 b and it follows that C0 = -0.0874 ; C1 = 0.961 ; C2 = -0.1068; H((f)) = C0 + C1exp(-2πjfT) + C2exp(-4πjfT) This will make the channel noise no longer white, introduces correlation in the noise from sampling point to sampling point, increases the noise at the sampling points and thus affects the assumptions used to design the matches filter. This is a disadvantage of this simple equalisation technique.

Page 11: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 11 25/08/2006 BMGC Question 3 (a)

.dB SQNR acceptable power with signalMin

overload avoidingpower signalMax log10D 10Y

= [1]

With uniform quantisation (with step-size ∆), the power of the noise created by quantisation-error remains constant (at ∆2/12) as signal power varies. [1] To make signal-to-quantisation-noise ratio (SQNR) acceptable for the lowest signal level within a specified dynamic range, ∆ must be chosen such that the SQNR is unnecessarily high for higher signal levels. [1] The ideal would be to have the SQNR independent of signal power. [1] To achieve this the step-size ∆ would have to vary with signal level. The smaller the voltage the smaller the value of ∆ as illustrated below: [1]

x(t)

t 001 111

Implemented by passing the samples of the input signal x(t) through a logarithmic “compressor” circuit to produce samples of a signal y(t) which are then quantised uniformly and transmitted. At the receiver, the quantised samples of y(t) are passed through an “expander” which reverses the effect of the compressor to produce output samples which are close to the original samples of x(t). [1] An ideal logarithmic compressor would require an infinitesimally small value of ∆ and hence an infinite number of quantisation levels to span the dynamic range. A compromise is to use logarithmic spacing between quantisation levels for voltages higher than a certain threshold and uniform (constant) spacing for voltages below this threshold. [1] 3(b) Sampling rate = 40kHz. Let number of bits/sample be b. Then bit-rate = 40000 b bits/seconds. Symbol time T = 1 / (40000 b) seconds [1] Channel bandwidth = 450 kHz and QPSK used with pulse shaping is 50% RRC Bandwidth of each symbol at base-band would be: (1 + 50/100)/(2T) = 20000 b (1.5) = 30,000 b Hz [1] This bandwidth is doubled by dsb modulation (multiplication by carrier) but with QPSK, we have in-phase and quadrature components each carrying data. [1] Therefore QPSK bandwidth = 30,000 b Hz. [1] (Bandwidth doubles but bit-rate doubles as well, so we get same bandwidth efficiency as at baseband). Bandwidth available = 450 kHz, therefore b = 450k / 30k = 15 bits/sample [1] Therefore max SQNR = 6 b + 1.8 dB = 91.8 dB. [1] Bandwidth efficiency = 15*40k / 450k b/s / Hz = 1.33 b/s per Hz [1] Dynamic range = 91.8 - 40 = 51.8 dB [1] 3(c) Explain coherent detection and its adv/disadv wrt non-coherent detection. Coherent detection requires a locally generated clock waveform at the receiver which is exactly synchronised to the received carrier. This allows a sine and cosine signal of the same frequency to carry separate data and thus increases the capacity of the channel. This is not possible with non-coherent

Page 12: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 12 25/08/2006 BMGC methods. The disadvantage is the additional complexity over non-coherent methods (which do not require the generation of a local carrier). [2] Show how a QPSK transmission may be generated at the transmitter using a vector modulator and detected coherently at the receiver.

[2]

Transmit

QPSK

CosΩct

SinΩct

±V (Bit 1)

±V (Bit 2)

q(t)

i(t)

QPSK Detector: Coherent detection is achieved using a vector-demodulator fed with locally generated

versions of cos(2πfCt) and sin (2πfCt).

±V/2 (Bit 1)

±V/2 (Bit 2)

Ωc=2πƒc

Input)

CosΩct

SinΩct

LPF (MF)

LPF (MF)

Local carrier generator

Local carrier generator

[2]

Page 13: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 13 25/08/2006 BMGC Question 4(a) Solution: The matched filter is optimally tuned to the shape of the transmitted signal to minimise the effect of added noise on the channel. With binary signaling with input signals s1(t) and s2 (t) , corrupted by AWGN, a matched filter H((ƒ)) has the property that the ratio |a2(T)-a1(T)|/(2σ0 ) is maximised when ai(t) is the filter’s response to si(t), σ0 is the standard deviation of n0(t), the filter’s response to n(t), and the symbol rate is 1/T. Since the error rate is Q( |a2(T)-a1(T)| /(2 σ0 ) ) , maximising |a2(T)-a1(T)| / (2σ0 ) minimses the bit error rate. If white Gaussian noise (WGN) with 2-sided PSD N0 /2 Watts/Hz over -B to B Hz is applied to the filter with frequency response H((f)) , the power of the output n0(t) (also Gaussian noise) is:

when the bandwidth B of the noise is much wider than the bandwidth of the filter H((f)). ∫∞

∞−≈∴ dffHN 2

02

0 ))((0.5 σ

Let the noise variance be σ2 before the filter and σ02 after it.

It may be assumed that σ02 is equal to the power of n0(t) . [2]

If symbol si(t) with FT Si((f)) is passed through a filter with frequency-response H((f)), output waveform produced has FT spectrum S((f))H((f)) and by the inverse FT is as follows:

∫∞

∞−= dfefSfHta ftj

iiπ2))(())(()(

We take a threshold γ = (a1(T) + a2(T))/2 halfway between a1(T) and a2(T). Then the error probability PB = Q( z) with z = |a2(T)-a1(T)| /(2 σ0 ). [3] We would like to make |a2(T)-a1(T)| /(2 σ0 ) as large as possible. Let F = |a2(T)-a1(T)| 2 /( σ0 2) =

∫∫

∞−

∞−−

dffHN

dfefSfSfH jft

20

22

21

))((5.0

)))(())(())((( π

Schwartz’s inequality for complex valued functions x(u) and y(u) states that:

Equality applies if x(u) = k y*(u) ∫∫∫ ≤ dududuuyux 22 2

y(u) x(u) )()(

Taking u as f, x(u) as H((f)) and y(u) as (S1((f))-S2((f)) )e2πjft this inequality gives:

∫∫ ∫

∞−

∞−

∞−

≤dffHN

dfefSfSdffH jft

20

-

2221

2

))((5.0

)))(())((( ))(( F

π

∫∞

∞−−≤∴ dfefSfS

NF jft 22

210

)))(())(((5.01 π ∫

∞−−= dffSfS

N2

210

))(())((5.01

For equality, i.e. to maximise F and hence minimise Q( |a1(T)-a2(T)| / (2σ0)) we must have H((f)) = k (S1*((f)) - S2*((f)))e-2πjfT for some k. We call this the “matched filter”.

Page 14: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 14 25/08/2006 BMGC H((f)) is the FT of h(t), the filter’s impulse response. If the FT of s(t) is S((f)), then the FT of s(-t) is S*((f)). Also, if the FT of s(t) is S((f)), the Fourier transform of s(t-T) is S((f))e-2jπfT . Therefore h(t) is equal to s1(t) -s2(t) modified in the following 3 ways:

(i) reversed in time, (ii) delayed by T seconds, and (iii) multiplied by any constant k. [3] Substituting H((f)) = k [S1*((f)) - S2*((f))]e-2πjfT in the formulae for a1(t) & a2(T) with t=T, we obtain:

∫∞

∞−−=− dffSfSkTaTa 2

2121 |))(())((||)()(|

By Parseval’s theorem, this also means that when we have a matched filter:

))()(( scaledkE ))()((|)()(| 21 d2

2121 tstsofenergydttstskTaTa −==−=− ∫∞

∞−

For convenience take k = 1.

Therefore |a1(T)-a2(T)| / (2σ0 ) = Ed / (2σ0 ) and since |H((f))|2 = |S1((f))-S2((f))|2 for a matched filter, it follows that:

Therefore |a1(T)-a2(T)| / (2σ0 ) = Ed / 2√(0.5N0Ed) Therefore, error probability = Q(|a1(T)-a2(T)| / (2σ0) ) = Q(√(Ed/(2N0) ) )as required.

dENdffHN 02

02

0 5.0))((0.5 =≈ ∫∞

∞−σ

----------------------------- (b) If s1(t) is 2 volts for T seconds and s2(t) is -2 volts for T seconds, the energy of |s1(t) - s2(t)| is 4 2 T = 16 T. Given that N0 = 2 x 10-3 , Q(√(Ed/(2N0) ) ) = Q ( √(16T/(4x10-3) = Q( √(4 x 103 T ) ). if Q(z) < 0.001, from the graph z > 3.1. Therefore 4 x 103 T > 9.61 1/T < (4/9.61) x 103 Hz = 0.416 kHz This is the minimum symbol rate required.

Page 15: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 15 25/08/2006 BMGC Question 5 (a) solution: Binary frequency Shift Keying (BFSK) can be a very straightforward form of digital modulation, simple to generate and detect and, being of constant amplitude throughout a transmission, insensitive to fluctuations of the channel attenuation. It is effectively frequency modulation, but uses a set of two distinct frequencies to represent the required symbols. The principle is to transmit a constant amplitude sine wave whose frequency varies between the frequencies assigned to each symbol. For binary signalling there would be two frequencies, ƒ0 and ƒ1 say. The simple generation methods may be referred to illustrate the answer.

FSK

1 0

Minimum shift keying (MSK) is a form of FSK where the difference between ƒ0 and ƒ1 is not 1/T but 1/(2T) Hz. This narrow spacing makes MSK very efficient in its spectral utilisation, but the price to be paid is increased complexity in the generation and detection process. The spectrum of FSK can be reduced by a pulse-shaping filter which may be a 100r % root raised cosine frequency response filter. This is placed just before the FSK modulator and therefore controls how the frequency changes from ƒ0 to ƒ1 and vice-versa. In GSM based digital cellular mobile phone systems the shaping is not 100r% root raised cosine but something similar called a root-Gaussian filter. The latter has a root-Gaussian shaped gain response . GSM phones use Gaussian shaping in conjunction with MSK. i.e. “Gaussian MSK”.

fil

Convert to ±δ(t) impulse

Gaussian

FM Modulator(VCO)

Advantages of GMSK:

1. Constant envelope hence not too sensitive to varying

of RF amplifiers.

2. Detection based on frequency changes, therefore not

Doppler shifts in mobile systems, etc.

Disadvantages of FSK:

1. Less bandwidth efficient than QPSK

2. Bit error rate performance in AWGN a bit worse tha

Question 5(b): Gray coding makes the bit error rate equal tois high enough to cause jumps to non-adjacent symbols . With Gray coding there is only one bit difference between a '8_APK': Average power = (1/8)(4A2/2 + 4 (4A2)/2) = 5A2

Symbol rate = 1 k baud therefore bit rate is 3 kb/s.

0 1 0

attenuation on the channel and non-linearity

very sensitive to frequency shifts of channel,

n QPSK.

the symbol error rate except when the noise

djacent M-ary levels.

/4

Page 16: CS3282 Digital Communications Examination June'05 \(complete\)

CS3282 Exam June 2005 16 25/08/2006 BMGC EB = (5A2/4) / 3k = 5A2/12 x10-3 Joule '8-PSK': Average power = R2/2 Bit rate as before. EB = R2/6 x 10-3 Joule If EB (8-APK) = EB (8-PSK) 5A2/12 = R2 /6 Therefore R2 = 2.5 A2 and R ≈ 1.58 A For 8-APK, min distance = A For 8-PSK min distance ≈ 2πR/8 ≈ (2πR/8) x 1.58 A = 1.24 A Therefore 8-PSK is better for equal EB/N0. Question 5(b) concluded: Waveform for given '8-APK' scheme coding 000 110 011 101

2A

-A

A

V

3 ms2 ms1 ms

t

Waveform for given '8-PSK' scheme coding 000 110 011 101 :-

3 ms2 ms1 ms

-R

R

V

t


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