Digital Communication Assignment Department of Electronics and Telecommunication Third Year Undergraduates Bengal Engineering and Science University,

Shibpur

List of Problems with the corresponding Class Roll Numbers: 3.18 - 2 3.19 - 3, 4 3.25 - 5, 6 3.28 - 7, 8 3.29 - 9, 10 4.2 - 11, 12 4.3 - 13, 14 4.4 - 15, 16 4.7 - 17, 18 4.8 - 19, 20 4.9 - 21, 22 6.3 - 23, 24 6.4 - 25, 26 6.5 - 27, 28 6.6 - 29, 30 6.23 - 31, 32 6.24 - 33, 34 7.2 - 35, 36 7.5 - 37, 38 7.10 - 39, 40 (Refer: Communication Systems by Simon Haykin)

3.18) A PCM system uses a uniform quantizer followed by a 7-bit binary encoder. The bit rate of the system is equal to 50 106 b/s. (a) What is the maximum message bandwidth for which the system operates satisfactorily ? (b) Determine the output signal-to-(quantization) noise ratio when a full-load sinusoidal modulating Wave of frequency 1 MHz is applied to the input. Ans:- (a) Let the message bandwidth be W. Then , sampling the message signal at its Nyquist rate , and using an R-bit code to represent each sample of the message signal, we find that the bit duration is
Tb = =2

The bit rate is

1 50 106 27

= 2WR

The maximum value of message bandwidth is therefore Wmax = = 3.57 106 Hz

(b) The output signal-to-quantization noise ratio is given by 10 log10 (SNR)o = 1.8 + 6R = 1.8 + 6 7 = 43.8 dB

3.19: ans Let a signal amplitude lying in the range Xi 1/2i x xi+1/2i be represented by the quantized amplitude xi . The instantaneous square value of the error is (x-xi)2. Let the probability density function of the input signal be fX(x). If the step-size I is small in relation to the input signal excursion, then fX(x) varies a little within the quantum step and may be approximated by fX(xi). Then the mean-square value of the error due to signals falling within this quantum is E[Qi ]= 1/2 = 1/2
+1/2 +1/2

( )2() ( )2() ( )2 1/2

1/2

=fX(xi) dx 1/2

+1/2

=1/12i3 fX(xi) (1)

The probability that the input signal amplitude lies within its interval is

PI = 1/2

+1/2

()
dx

= fX(xi) 1/2 = fX(xi) i

+1/2

..(2)

Therefore , eliminating fX(xi) between eq. (1) and eq.(2), we get

E[Qi2]=1/12 Pi i2
The total mean-square value of the quantizing error is the sum of that contributed by each of the several quanta. Hence, (solved) [2 ] = 1/12 2

Problem 3.25 The test signal m(t) = A tanh (t) To avoid slope overload, we require, 2 /T(S) max|A sech (t)| ------------------------------------------------------------------------------------------(1) From m(t),
dm(t) dt

=A sech2 (t)

----------------------------------------------------------------------2

------(2) Using equation 2 ,

/T(s) max|A sech (t)|

1 cosh(t) 2

sech(t) = then

sech(t) = t+t

max of sech(t)=1 so that , /T(s) max|A| AT(s)

Problem 3.28
For the sinusoidal signal m(t)=Amsin(2fmt)
dm(t) =2fmAmcos(2fmt) dt dm(t) or,| dt |max=|2fmAm|max

or,

similarly,
Ts

|2fmAm|max
fs

|Am|max=Ts2fm=2fm hence, =
2fmA fs fs 2

The average signal power=2 /2=0.5(2fm) Average quantization noise power is

2 3

The waveform of the reconstruction error is a pattern of bipolar binary pulses characterized by duration=Ts = 1/fs and average power=/3. Hence,the autocorrelation function of the quantization noise is triangular in shape with a peak value of 2 /3 and base 2Ts

Area of the triangle =

Ts/3

From Random Process theory, Sq(f) |f=0=

+

Rq()d

which yields SQ(0)=2 Ts/3 Typically in Delta Modulation,the sampling rate fs is very large compared to the highest frequency component of the original message signal. We may therefore approximate the power spectral density of granular quantization noise as SQ(f)= 2 /3fs 0. -w f w otherwise

where w= bandwidth of reconstruction filter at the demodulation output Hence the average quantization noise power, w N= SQ(f)df=22 w/3fs w Substituting the values in equation, N=2[2 fmA/fs]2w/fs =8fm2A2w/3fs3 b) Correspondingly Output SNR is SNR =[()A2]/[(8fm2A2w)/3fs3] =3fs3/(162fm2w)

3.29: (a) Say the modulating signal is m(t)=A

mCos(2fmt).

The slope of m(t) is= =-2fmAM Sin(2fmt). So the maximum slope can be =2fmAm. The maximum average slope of the overloading signal m a(t) produced by the DM is fs where is the step-size and fs is the sampling frequency.The limiting value of amplitude is therefore givenby2fmAmfs (2fmAm)/fs So the step-size (2103)/(50103) = 0.126 volt (b)Nq=

n2fN(n)dn

=(1/2)(23/3) =2/3 SN(f)=Nq/2fs=2/6fs from the previous value of , SN(f)=(42A2fm2Ts2)/6fs noise,N=w SN(f)df=(42A2fm2Ts2)*2W/6fs SNR=S/N=((A2/2)/N) =(3fs3/82fm2W) =(3/82)(fs3/fm2w) here, (3/82)(50*103)3/(103)2*5*103 =950 10 log10(SNRo)= 10log10(950)=30dB
+w

4.2(a) s1( t)= {+A/2, 0tT/2

-A/2, T/2tT } Therefore the matched filter for s1(t) is h1(t)=s1(T-t )

Therefore the matched filter response for s1(t) is h1(t)=s1(T-t ) which is as follows:
T

y11 =

s1 ()h1 (t )d

S2(t)={ -A/2, 0tT/4 +A/2, T/4t3T/4 -A/2, 3T/4tT }

Therefore the matched filter response for s2(t) is h2(t)=s2(T-t ) which is as follows:

s2(t)=

h2(t)=

b) (i) When the pulse s1(t) is applied to the two-dimensional filter, the response of the lower
matched filter is s1(t)* h2(t)=y12 = 0
T

s1 ()h2 (t )d, which is obtained as follows:

The response of the lower matched filter is zero in this case as the area under the curve obtained 0tT is equal and opposite in sign to that obtained Tt2T (i.e. the area above the x-axis is equal to that obtained below the x-axis) and hence the net area under the curve is zero.

(ii) When the pulse s2(t) is applied to the two-dimensional filter, the response of the upper matched filter is s2(t)* h1(t)=y21 = 0
T

s2 ()h1 (t )d, which is obtained as follows:

Again the response of the upper matched filter is zero when pulse s2(t) is applied to the 2-d filter because the net area under the curve obtained above is zero.

Generalization : From the above results we conclude that when s1(t) is applied to the input, the output of the filter matched to s1(t) yields a result while the output of the filter matched to s2(t) is zero. Again, when s2(t) is applied to the input, the output of the filter matched to s2(t) yields a result while the output of the filter matched to s1(t) is zero. Generalizing,if we consider that there are n pulses s1(t),s2(t),....sn(t) with corresponding matched filters h1(t), h2(t),.... hn(t),then if an input (for example s1(t)) is applied to the input then the output of the corresponding matched filter (h1(t)) will have a value while the outputs of the filters corresponding to the other inputs will be zero.

Problem 4.3

Problem 4.6

4.8.(a) The average probability of error is Pe =1/2erfc(Eb/No)

NoTb.

where Eb=A2 Tb
1

We may rewrite the formula as Pe=2erfc(A/) where A is the pulse amplitude at = Let u=Eb/No=A/ We are given that 2 =10^-2 V^2
2 8

=.1 Volt

Pe=10

8 2

Since Pe is quite small we can approximate it as erfc(u)=exp(-u

)/u

Hence exp(-u )u/2=10 Solving this eqn we get u=3.97 The corresponding value of pulse amplitude is A=u = 0.13.97=0.397 b. Let 2 T = + 2 i where 2 due to noise and 2 i due to interference. The new value of average probability of error is 10^-6 So,106=1/2erfc(A/T ) = erfc(uT ) where uT =A/T The above equation may be approximated as exp(-u2 T )/2uT =10^-6 Solving for uT ,we get uT =3.37 The corresponding 2 T =(A/uT )^2=0.0138 volt^2 The variance in interference is therefore
2 i =2 T -2
2

=0.0138-0.01 =0.0038 volts2

Problem 4.9 Consider the performance of a binary PCM system in the presence of channel noise. We evaluate the average probability of error for such a system. We consider two different types of signals separately. We begin by considering the first kind of error that occurs when symbol 0 is sent and the receiver chooses symbol 1. In this case, the probability of error is just the probability that the correlator output will exceed the threshold l owing to the presence of noise, so the transmitted symbol is mistaken for symbol 1. The probs for 0 and 1 are equal. l = A2Tb/2 [ Tb is bit duration]

let y denote the correlator output:

Under hypothesis H0 :

[ conditional variance of the correlator output ] Now proceeding to find the auto-correlation function and integrating,

is the

where subscript in f0(y) signifies the condition that symbol 0 was sent. Now the conditional probability of error , given that symbol 0 was sent is

Let

Hence rearranging the terms :

Now complementary error function

We redefine the conditional probability of error as

Similarly we can find p01. Hence average probability of error is Pe = p0p10 + p1p01 Therefore

(Proved)

(Kuntal Roy and Narendra Prasad) Problem 6.3: Question: Consider a phase-locked loop consisting of a multiplier, loop filter, and voltage-controlled oscillator (VCO). Let the signal applied to the multiplier input be a PSK signal defined by: () = cos[2 + ()] Where kp is the phase sensitivity, and the data signal m(t) takes on the value +1 for binary symbol 1 and -1 for binary symbol 0. The VCO output is () = sin[2 + ()] (a) Evaluate the loop filter output, assuming that this filter removes only modulated components with carrier frequency 2f. (b) Show that this output is proportional to the data signal m(t) when the loop is phase locked, that is (t)=0. Solution: (a) The noiseless PSK signal is given by () = cos[2 + ()] = cos(2 ) cos ( ()) sin(2 ) sin ( ()) Since m(t)=1, it follows that cos ( ()) = cos( ) = cos( ) sin ( ()) = sin( ) = sin( ) Therefore, () = cos(2 ) cos( ) sin(2 ) sin( ) The VCO output is () = sin[2 + ()] The multiplier output is therefore 1 ()() = cos sin () 2 1 + sin(4 + ()) () sin cos () + cos(4 + ()) 2 The loop filter removes the double-frequency components, producing the output 1 1 () = cos sin () () sin cos () 2 2 Note that if kp=/2, (i.e., the carrier is fully deviated), there would be no carrier component for the PLL to track. (b) Since the error signal tends to drive the loop into lock (i.e., (t) approaches zero), the loop filter output reduces to 1 () = sin( )() 2 which is proportional to the desired data signal m(t). Hence, the phase-locked loop may be used to recover m(t).

Problem 6.4 :

(a) The Signal Space Diagram is : This differs from conventional PSK in that it is 2-dimensional, if k is reduced to zero it reduces to the original form. (b)

The signal at the dicision input is

proceeding similarly as in the case of aconventional PSK system, we see that the probability of error is

where

( c ) For the case where

and

we get,

where

using the approximation ,

we get,

solving we get u=2.64.

Hence, Expressed in decibel this value corresponds to 8.9db. ( d ) For a conventional PSK system, we have,

In this case , Expressed in decibels, this value corresponds to 8.4db. Thus the conventional system requires 0.5db less than the modified schme for the same probability of error.

( Solved )

6.5)

6.6) Let PeI = average probability of symbol error in to the in-phase channel PeQ= average probability of symbol error in to the quadrature channel Since the individual outputs of the in-phase and quadrature channels are statistically independent , the overall average probability of correct reception is; Pc= ( 1 - PeI ).( 1 - PeQ ) = 1 PeI PeQ + PeQ. PeI The overall average probability of error is therefore; Pe = 1 Pc = PeI + PeQ - PeQ. PeI

6.23 a) Since two oscillators used to represent symbols 1 and 0 are independent , we may view the resulting binary FSK wave as the sum of two on-off keying (00k) signals. One 00k signal operates with the oscillator of frequency f1 . The second 00k signal operates withthe oscillator of frequency f2 . The power spectral density of a random binary wave x1(t) is represented by A volts and symbol 0 by 0 volt is given by , , in which symbol 1

sX1(f) = A2 (f)/2 + A2Tbsinc2(fTb)/4

where Tb is the bit duration . When this binary wave is multiplied by a sinusoidal wave of unit amplitude and frequency

fc+f/2 we gte the first 00K signal with ,

A=2EB/TB
The power spectral density of this 00k signal equals,

S1(f)= 4[ Sx1 (f - fc -fc/2) + Sx1 (f - fc - fc/2) ]

The power spectral density of random binary wave x2(t)=x1(t)

, in which symbol 1 is

represented by 0 volt and symbol 0 is represented by A volts , is given by ,

sx2(f) = sx1(f)
When x2(t) is multiplied by second sinusoidal wave of unit amplitude and frequency ,fc

f/2 we get the second 00k signal whose power spectral density equals S2(f)= 4[ Sx2(f - fc + f/2 )+ Sx2(f + fc - f/2) ]
1

the power spectral density of the FSK signal equals ,

SFSK(f)=S1(f) + S2(f) =EB/8TB [ (f - fc - f/2) + (f + fc + f/2) +(f fc +f/2) + (f + fc - f/2) ] + EB/8TB {sinc2[TB(f - fc -f/2)] + sinc2[TB(f + fc + f/2)] + sinc2[TB(f - fc + f/2)] + sinc2[TB( f + fc - f/2)] }
This result shows that the power spectrum of this BFSK wave contains delta funtioc at

= fc f/2.
b) From the property of the sinc function we know that at high values of x , sinc(x) falls off proportionally with 1/x.
Here we can see that power spectral density of the BFSK signal varies with the square of the sinc function at higher values of f compared to that of carrier frequency. so , from this we can conclude that power spectral density of BFSK signal varies with 1/f at high values of f (compared to fc ). 2

6.24 :

1 = E 10 = E

2 = E 5 = 7 = sin(2fct)

4 = 6 = Tb sin(Tb) 9 = cos(2fct)

8 = Tb cos(Tb)

7.2. The sequence of outputs present by the shift register at each clock are given 1000,0100,0010,1001,1100,0110,1011,0101,1010,1101,1110,1111,0111,0011,0001,1000 the output sequence is 0001001101011110 solved

Problem 7.5: Madhuja Chattopadhyay and Swarnava Majumdar Table 1(as given in text-book) Evolution of the maximal-length sequence generated by the feedback shift register State of shift Register Feedback Symbol 1 0 1 0 1 0 0 1 0 0 0 1 0 1 1 1 1 1 0 1 1 0 0 1 1 1 0 0 0 0 1 1 1 0 1 0 1 0 0 1 0 0 0 1 0 1 1 1 1 1 0 1 1 0 0 1 1 1 0 0 0 0 1 0 1 1 0 1 0 1 0 0 1 0 0 0 1 0 1 1 1 1 1 0 1 1 0 0 1 1 1 0 0 0 0 0 0 1 1 0 1 0 1 0 0 1 0 0 0 1 0 1 1 1 1 1 0 1 1 0 0 1 1 1 0 0 0 0 0 0 1 1 0 1 0 1 0 0 1 0 0 0 1 0 1 1 1 1 1 0 1 1 0 0 1 1 1 0 0 0 0 0 0 1 1 0 1 0 1 0 0 1 0 0 0 1 0 1 1 1 1 1 0 1 1 0 0 1 1 1 0 Output Symbol 0 0 0 0 1 1 0 1 0 1 0 0 1 0 0 0 1 0 1 1 1 1 1 0 1 1 0 0 1 1 1

Now, as per the problem: Table 2 State of feedback-shift register Initial State 1 0 0 1 0 0 0 0 0 0 1 1 1 0 1 0 0 1 0 0 1 1 1 0 0 1 1 1 1 0 1 0 1 0 1 0 0 1 1 1 0 1 0 0 0 0 1 1 0 1 0 0 1 1 0 1 1 1 0 1 1 1 1 0 Output Symbol 0 0 1 0 0 1 0 1 0 1 1 0 0 0 0 1 1 1 0 0 1 1 0 1 1 1 1 1 0 1 0 0 0 0 0 1 0 0 1 0 1 0 1 1 0 0 0 0 1 1 1 0 0 1 1 0 1 1 1 1 1 0 1 0 0 0 0 0 1 1 1 0 0 1 1 0 1 1 1 1 1 0 1 0 0 0 1 0 0 1 0 1 0 1 1 0 0 0 0 0 1 1 1 0 0 1 1 0 1 1 1 1 1 0 1 0 0 0 1 0 0 1 0 1 0 1 1

The 31-element code generated by the scheme as shown in the above table is exactly the same as that described in Table 1. However, the code described in Table 1 appears in reversed order to that described the above Table 2; this reversal is clearly of a trivial nature.

7.10:

Question: In a DS/BPSK system,the feedback shift register used to generate the

PN sequence has length m =19.The system is required to have an average probability of symbol error due to externally generated interfering signals that does not exceed 10-5 .Calculate the following system parameters in decibels: (a) Processing gain. (b) Antijam margin. Answer: (a) Processing gain = 10log10(2m-1) = 10 log10(219-1) =57dB (b) Antijam margin =(Processing gain) 10log10(Eb/NO) The probability of error is Pe =
1 2

erfc (Eb/No) (is case of BPSK)

With Pe =10-5 , and from the erfc table we have Eb/NO =9. Antijam margin = 57 -10log 109 = 57-9.5 = 47.5 dB