Digital Communication 6th Sem B.E.

June 2012 Solved

PART - A
1. a.
ANS-1.a.

Figure 1: Basic Digital Communication System

The Block Diagram Shown above consists of 3 main Blocks:

i) Transmitter
ii)Communication Channel
iii) Receiver
Digital Information source:
The source of information is assumed to be digital i.e. symbols, letters etc.
If I/p is analog signal, then it is converted into digital form by using sampler & quantizer.
The Sources of information are human voice, Television picture, Teletype data, atmospheric
temperature & pressure etc.
Source Encoder & Decoder:
Digital information coming out of source consists of lots of redundancy which when transmitted as it is
results in Improper utilization of Bandwidth. Hence results in poor efficiency. The objective of source
encoder is to eliminate or reduce redundancy.
Source Decoder:
Source Decoder at the receiver behaves exactly in a reverse way to source encoder. Decoder converts the
codes back to symbols i.e. converts digital information to discrete symbols.
Channel Encoder/Decoder:
Channel Encoder & decoder are used to reduce the channel Noise Effect.

Digital Communication 6th Sem B.E. June 2012 Solved

Channel coding is the process of adding controlled redundancy to the data to be transmitted, to

detect and/or correct the errors caused by the channel noise at the receiver.
Addition of redundancy increases bit rate & hence increases bandwidth.
Decoder detects the error in the received data & corrects the error.
E.g.:- Error correcting codes like linear block codes, cyclic codes & convolution codes.
Modulator & Demodulator:
Modulator converts the bitstream into a waveform suitable for transmission over the

communication channels. E.g.:- ASK, FSK, PSK, QPSK etc.

Demodulator converts the waveform into digital data. {optimum detectors are used to minimise

the probability of error.}

Communication Channel:
It is the media through which signal can be transmitted.
E.g.:- Free Space, Twisted wire, Co-axial cable, Waveguide, OFC etc.
1. b.
ANS- 1.b.

g ( t )=10 cos 20 t . cos 200 t

A=200, B=20
W.K.T CosA.CosB= [Cos (A-B) + Cos (A+B)]

g ( t )=

10
[ cos ( 20020 ) t+cos ( 200+20 ) t ]
2

g ( t )=5 [ cos 180 t+ cos 220 t ]

g ( t )=5 cos 180 t + 5 cos 220 t
From eqn2,

W1=180
2

f1= 180

W2=220
2

f1= 90Hz
f2=110Hz
fm=max(f1,f2)=f2=110Hz

f2=220
fm=110Hz

g ( t )=5 cos 2 (90)t+5 cos 2 (110) t

Taking FT on both sides of eqn3,we get

G(f) =

5
5
( f 90 )+ ( f + 90 ) ] + [ ( f 110 )+ ( f + 110 ) ]
[
2
2

G(f) =

2.5 [ ( f 90 ) + ( f + 90 ) ]+ 2.5 [ ( f 110 ) + ( f +110 ) ]

The Spectrum of the signal g(t) is drawn using eqn4 as shown in figure below:

Digital Communication 6th Sem B.E. June 2012 Solved

i) WKT

G (f) = fs

G ( f nfs )

Given: fs=250Hz

n=

(f) = [ ( f 250 n90 )+ (f 250 n+90)]+2.5 fs

n=

2.5 fs

n=

ii)

Digital Communication 6th Sem B.E. June 2012 Solved

The cut-off frequencies of the ideal LPF should be more than 110Hz & less than 140Hz for recovering g(t)
from g

(t).

iii) Nyquist rate for g(t):

Fs=2fm = 2* 110Hz
Fs=220Hz
1. c.
ANS- 1.c. Flap-Top Sampling
As the name itself indicates after sampling, the pulses will have Flat Top. It is very easy to

obtain flat of samples compared to Natural samples.

The top of the samples remains constant &equal to instantaneous value of the base band signal g(t) at the
start of sampling.

Digital Communication 6th Sem B.E. June 2012 Solved

It is observed from the figure that only starting edge of the pulse represents the instantaneous value of base
band signal g(t).

(t) is given by
g (t) = g(t). S (t)

The Sampled signal g

(t) = g(t)

( tnTs )

n=

(t) =

g ( nTs ) (t nTs)

n=

(t) with the pulse h(t), we get

S (t) = g (t)*h (t)
2

Convolving g

S (t) =

g ( t ) . h ( t ) d

Substituting eqn1 in eqn2, we get

S (t) =

g ( nTs ) ( nTs ) h ( t ) d

n=

{From Shifting Property

x ( t ) . ( t ) dt=x ()

h ( t ) ( nTs ) d=h ( tnTs )

}
Applying Shifting property in eqn4, we get

S (t) =

g ( nT s ) h( tnTs)

n=

Eqn5 represents the value of S(t) in terms of sampled value g(nTs) & function h(t-nTs) for flat top sampled
signal.
WKT
S (t) =

(t) * h (t)

Taking F.T on both sides of above equation, we get

S (f) = G
Substituting G

(f) . H (f)

(f) in above equation, we get

S (f) = fs

G ( f nfs ) . H ( f )

n=

Eqn6 represents the spectrum of flat Top Sampled Signal.

Digital Communication 6th Sem B.E. June 2012 Solved

2. a.
ANS- 2. a.
Let the random variable Q denotes the quantization error & q its sample value.
Let us assume that the quantization error Q is uniformly distributed over a single quantizer
interval

Hence, probability density function (PDF) of Quantization error Q is then

=0

The mean quantization error

The variance of Quantization error is

Digital Communication 6th Sem B.E. June 2012 Solved

/2

( q)2 .

/ 2

fQ (q). dq

( q )2 . 1 . dq

Eqn1 is known as Mean Squared Quantization Error or Normalized Noise Power or

Quantization error in terms of power.

Let us Consider N- bits to represent L quantized level, then

L=2 N

Digital Communication 6th Sem B.E. June 2012 Solved

Let p denotes the average power of the message signal x(t), then the o/p SNR of a
uniform quantizer is

2. b.
ANS -2. b.
Soln:-

Given:

=100, (SNR)dB = 45dB, L=?

(SNR)dB = 45dB

Digital Communication 6th Sem B.E. June 2012 Solved

10 log10 (SNR) =45 dB
(SNR) =

45
log 1
10
( 10

(SNR)o =31622.7766

2. c.
ANS -2. c.
An important feature of TDM is conservation of time i.e. different time intervals (periods) are allocated for
different message signals, so that a common channel is utilized for transmission of these signals without
any interference.

Digital Communication 6th Sem B.E. June 2012 Solved

The concept of TDM is illustrated in the block diagram.

The Low Pass pre-alias filters are used to remove high frequency components which may be present in the

message signal.
The o/p of the pre-alias filters are then fed to a commutator, which is usually implemented using electronic

switching circuitry.
The function of Commutator is 2 fold:
1) To take a Narrow Sample of each of the N I/p signals at a rate fs

2W, where W is the cut-off

frequency of pre-alias filter.

2) To sequentially interleave these N samples inside a sampling interval Ts= 1/fs .
This interleaving is nothing but multiplexing.
The multiplexed signal is applied to a pulse amplitude modulator whose purpose is to transform the

multiplexed signal into a form suitable for transmission over a common channel.
At the receiving end, the pulse amplitude demodulator performs the reverse operation of PAM & the

decommutator distributes the signals to the appropriate low pass reconstruction filters.
The decommutator operates in synchronization with the commutator.

Suppose that the N message signals to be multiplexed (Txed) have the same spectral properties (BW). Then

the Sampling rate for each message signal is determined in accordance with the Sampling theorem.
Let Ts denotes the Sampling period.
Let Tx denote the time spacing between adjacent samples in the TDM signal.

Digital Communication 6th Sem B.E. June 2012 Solved

i.e. Tx= Ts

N as shown in fig2.

Number of pulses per second = 1/ Tx = 1/(Ts/N) = N/ Ts

Number of pulses per second is also called as Signalling rate r
i.e. r=Nfs
Since fs
r

2fm

N2fm

Transmission Bandwidth =

Signalling rate
2

3. a.
ANS -3. a.
Delta Modulation transmits only one bit per sample i.e. the present sample value is compared with the
previous sample value & the indication, whether the amplitude is increased or decreased is sent.
The i/p signal x(t) is approximated to step signal by the delta modulator. The difference between I/p signal
x(t) & staircase approximated signal is quantized into only two levels i.e. +

or .

If the difference is +ve, then approximated signal is increased by one step i.e. +

bit 1 is

transmitted.

If the difference is ve, then approximated signal is reduced by one step i.e.

Thus for each sample only one-bit is transmitted.

DM Transmitter:-

& bit 0 is transmitted.

Digital Communication 6th Sem B.E. June 2012 Solved

The error between the sampled value x(nT s) & last approximated sample is given by
e(nTs) = x(nTs) -

^x (nTs)

Let u(nTs) be the present sample approximation of Staircase o/p.

From above Fig:

^x (nTs) = u(n-1)Ts
^x (nTs) = u(nTs-Ts)

Substituting eqn2 in eqn1, we get

e(nTs) = x(nTs)- u(nTs-Ts)

The binary quantity b(nTs) is the algebraic sign of the error e(nTs), except for the scaling factor
i.e. b(nTs) =

sgn[e(nTs)]

b(nTs) depends on the sign of error e(nTs), the sign of step-size

i.e.

b(nTs) = +
b(nTs) = -

If b(nTs) = +

, if x(nTs)
, if x(nTs)

will be decided

^x (nTs)
^x (nTs)

, then binary 1 is transmitted, & if b(nTs) = -

, then binary 0 is transmitted.

Hence, u(nTs) =u[nTs-Ts] + b(nTs)

The previous Sample approximation u[nTs-Ts] is restored by delaying one sample period T s.

DM Receiver:-

Fig above shows the block diagram of DM Receiver.

Digital Communication 6th Sem B.E. June 2012 Solved

The accumulator generates the Staircase approximated signal O/p & is delayed by one Sampling
period Ts. It is then added to the I/p Signal.

If I/p is binary 1 then it adds +

If I/p is binary 0 then one step

The LPF is used to remove Step variation & to get smooth reconstructed message signal x(t).

''

step to the previous o/p.

is subtracted from the delayed Signal.

DM Systems are subjected to two types of Quantization error

1) Slope-Overload distortion
2) Granular Noise

1) Slope overload distortion:

Slope overload distortion arises because of the large dynamic range of the I/p Signal.
In Fig above, it can be seen, the rate of rise of I/p signal x(t) is so high that the Staircase Signal cannot
approximate it, the Step Size

becomes too small for Staircase Signal x(t) to follow the Step

Segment of x(t). Thus large error between the Staircase approximated Signal & the original I/p Signal x(t).

This error is called Slope overload distortion.

To reduce this error, the Step-Size should be increased when slope of the Signal x(t) is high.
i.e. Slope of the Staircase u(t)

Granular Noise:-

Slope of the message Signal.

Digital Communication 6th Sem B.E. June 2012 Solved

This noise occurs when the Step Size is too large compared to small variations in the I/p Signal i.e. for very
Small variations in the I/p Signal, the Staircase Signal is changed by large amount because of large Step
Size

In Fig above, I/p Signal is almost flat, the Staircase Signal u(t) keeps on oscillating by +

Signal.
The error between the I/p & approximated Signal is called Granular Noise. The Solution of this problem is
to make Step Size small.

3. b.
ANS -3. b.

Polar NRZ Format:-

around the

Digital Communication 6th Sem B.E. June 2012 Solved

Digital Communication 6th Sem B.E. June 2012 Solved

3. c.

Digital Communication 6th Sem B.E. June 2012 Solved

ANS -3. c.
Sr.
No

Parameter

PCM

Differential Pulse Code

Modulation (DPCM)

.
1

Number of Bits

It can use 4, 8 or 16 bits per sample.

Bits can be more than one but

Levels, step size

The number of levels depend on

are less than PCM.

Fixed number of levels are

Quantization error and

number of bits. Level size is fixed

Quantization error depends on number

used.
Slope overload distortion and

Distortion
Bandwidth of transmission

of levels used.
Highest bandwidth is required since

quantization noise is present.

Bandwidth required is lower

channel
Feedback

number of bits are high.

There is no feedback in transmitter or

than PCM.
Feedback exists.

6
7
8

System Complexity
SNR
Applications

receiver.
System is complex.
Good.
Audio & Video Telephony.

Simple.
Fair.
Speech and Video.

4. a.
ANS -4. a.
Nyquist Pulse Shaping Criterion:In detection process received pulse stream is detected by sampling at intervals KT b, and then in detection
process we will get desired output. This demands sample of i th transmitted pulse in pulse stream at K th sampling
interval should be

------------- (1)
If received pulse P(t) satisfy this condition in time domain, then

y(ti) =

ai

Let us look at this condition by transform eqn (1) into frequency domain.
Consider sequence of samples {P(nTb)} where n=0,1. . . . . . . by sampling in time domain, we write in frequency
domain

1
P(f) = T b

p(f

n=

n
)
Tb

Where P (f) is Fourier transform of an infinite period sequence of delta functions of period T b but P(f) can
be obtained from its weighted sampled P(nTb) in time domain.

P(f) =

m=

p (mTb ) (t-mTb ) e j 2 ft dt = p(t). (t)

Where m = i-k, then i=k, m=0; so

Digital Communication 6th Sem B.E. June 2012 Solved

p ( 0 ) ( t ) e j 2 ft

P(f) =

dt

Using property of delta function

i.e.

(t)

dt =1

Therefore P(f) =P(0) =1

Hence, P(f) =1
3
P(0) =1 ,i.e. pulse is normalized (total area in frequency domain is unity)
Comparing (3) and (2)

f n
p
1

/ Tb ) =1

n=

Tb
OR

f n
p

/ Tb ) = Tb =1

n=

Rb
Where Rb = Bit Rate
Is desired condition for zero ISI and it is termed Nyquists first criterion for distortion less base band
transmission. It suggests the method for constructing band limited function to overcome effect of ISI.
4. b.
ANS -4. b.
EYE Pattern: The Eye Pattern is used to study the effect of ISI in a PCM or data transmission System.
Eye Pattern can be obtained by applying the received wave to the vertical deflection plates of an
oscilloscope & to apply a sawtooth wave to the horizontal deflection plates at a transmitted Symbol rate

R= 1/T.
The waveforms in Successive Symbol intervals are thereby translated into one interval on the oscilloscope
display as shown in fig 1- a & b.

Digital Communication 6th Sem B.E. June 2012 Solved

The resulting display is called Eye Pattern because of its resemblance to the human eye for binary waves.

The interior region of the Eye Pattern is called the Eye Opening.
An Eye Pattern provides information about the performance of the System, as described in Fig.2.

1) The width of the Eye Opening defines the time interval over which the received wave can be sampled
without error from ISI.

Digital Communication 6th Sem B.E. June 2012 Solved

2) The Sensitivity of the System to timing error is determined by the rate of closure of the Eye as the
Sampling time is varied.
3) The height of the Eye Opening, at a specified Sampling time defines the margin over Noise.
4) Any non linear transmission distortion would reveal itself in an asymmetric or squinted eye. When the
effected of ISI is excessive, traces from the upper portion of the eye pattern cross traces from lower portion
with the result that the eye is completely closed.
Example of eye pattern:
Binary-PAM Perfect channel (no noise and no ISI)

4. c.
ANS -4. c.
Adaptive Equalization: The transmission characteristics of the Channel keep on changing. To compensate this, adaptive

equalization is used.
In Adaptive Equalization, the filters adapt themselves to the dispersive effects of the channel. The coefficients of the filters are changed continuously according to the received data in such a way that the

distortion in the data is reduced.

There are two types of equalization:

Digital Communication 6th Sem B.E. June 2012 Solved

1) Pre-Channel Equalization
2) Post-Channel Equalization.
Pre- Channel equalization is done at the transmitting side. It requires feedback to know the amount of

distortion in the received data.

In post-channel equalization, feedback is not required. The equalizer is placed after the receiving filter in
the receiver.

Fig Above shows the block diagram of an adaptive equalizing filter. It is adaptive in nature because it is
capable of adjusting its co-efficients w0, w1, .. , wm-1 by operating on the channel o/p in accordance with

some algorithm.
The adaptive equalizing filter consists of delay elements & adjustable filter co-efficients (Taps).
The Sequence x(nT) is applied to the I/p of the adaptive filter . The O/p y(nT) of the adaptive filter will be:
M

y(nT) =

w
i=0

x(nT-iT)

A known sequence {d(nT)} is transmitted 1st . This Sequence is known to the receiver.
An error Sequence is calculated i.e.
e(nT) = d(nT) y(nT)
If there is no distortion in the channel, then d(nT) & y(nT)will be exactly same producing zero error

Sequence.
If there is distortion in the Channel, then e(nT) exists. The weights of the filter i.e. w i are changed

recursively such that error e(nT) is minimized.

The algorithm used to change the weights of the adaptive filter is Least Mean Square Algorithm (LMS).
The tap weights are adapted by this algorithm as follows:

w
^
Where,

w
^
w
^

(nT + T) =

w
^

(nT) +

e(nT) x(nT- iT)

(nT) is the present estimate for Tap i at time nT

(nT + T) is the updated estimate for tap i at time nT

Digital Communication 6th Sem B.E. June 2012 Solved

is the adaptation constant
x(nT iT) is the filter I/p &
e(nT) is the error Signal.

PART - B
5. a.
ANS -5. a.
Requirements of Passband Transmission Scheme:Any passband transmission scheme should satisfy following requirements:1. Maximum data transmission rate.
2. Minimum probability of symbol error.
3. Minimum transmitted power.
4. Minimum channel bandwidth.
5. Maximum resistance to interfering signals.
6. Minimum circuit complexity.
5. b.
ANS -5. b.
Power Spectral Density of a BPSK Signal:Step 1: Fourier transform of basic NRZ pulse.
We know that the waveform b (t) is NRZ bipolar waveform. In this waveform there are rectangular pulses of
amplitude Vb. If we say that each pulse is (T b

2) around its center as shown in Fig.1, then it becomes easy to

find Fourier Transform of such pulse.

Fig.1 NRZ pulse

Digital Communication 6th Sem B.E. June 2012 Solved

The Fourier Transform of this type of pulse is given as,

Tb) By Standard relations:

sin

X(f)= Vb Tb

( f

Tb)

Step 2: PSD of NRZ pulse.

For large number of such positive and negative pulses the power spectral density S(f) is given as
S(f) =

X ( f )

Ts

Here

X (f ) denotes average value of X(f) due to all the pulses in b (t). And T s is symbol duration. Putting value

of X(f) from equation 1 in equation 2 we get,

Step 3: PSD of baseband signal b(t).

For BPSK since only one bit is transmitted at a time, symbol and bit durations are same i.e. Tb = T s. Then above
equation becomes,

3
The above equation gives the power spectral density of baseband signal b (t).
Step 4: PSD of BPSK signal.
The BPSK signal is generated by modulating a carrier by the baseband signal b (t). Because of modulation of the
carrier of frequency f0, the spectral components are translated from f to (f 0 + f) and (f0 f). The magnitude of those
components is divided by half.
Therefore from equation 3 we can write the power spectral density of BPSK signal as,

The above equation is composed of two half magnitude spectral components of same frequency 'f above and below
f0 Let us say that the value of Vb =

. That is the NRZ signal is having amplitudes of +

and -

. Then above equation becomes,

4
The above equation gives power
spectral density of BPSK signal

Digital Communication 6th Sem B.E. June 2012 Solved

for modulating signal b(t) having amplitudes of

2 P

f0t) and s2(t) = cos

s(t) =

If b(t) =

2 P

2 P

f0t)
cos

. We know that modulated signal is given by s1(t) =

f0t) as,
cos
Since, A=

2 P

, then carrier becomes,

(t) =

f0t)
cos

Plot of PSD:Equation 3 gives power spectral density of the NRZ waveform. For one rectangular pulse, the shape of S (f) will be
a sinc pulse as given by equation 3. Fig. 2 shows the plot of magnitude of S (f). Below figure shows that the main
lobe ranges from -fb to + fb Here fb = (1/Tb). Since we have taken Vb =

in equation 3, the peak value of

the main lobe is P Tb,

Fig. 2 Plot of power spectral density of NRZ baseband signal

Now let us consider the power spectral density of BPSK signal given by equation 4. Fig. 3 shows the plot of this
equation. The figure thus clearly shows that there are two lobes; one at f 0 and other at -f0. The same spectrum of Fig.
2 is placed at + f0 and - f0 But the amplitudes of main lobes are (PTb/2) in Fig. 3.

Digital Communication 6th Sem B.E. June 2012 Solved

Fig. 3 Plot of power spectral density of BPSK signal

Thus they are reduced to half. The spectrums of S (f) as well as SBPSK (f) extends over all the frequencies.
5. c.
ANS -5. c.
Advantages of MSK as Compared to QPSK:1. The MSK baseband waveforms are smoother compared to QPSK.
2. MSK signal have continuous phase in all the cases, whereas QPSK has abrupt phase shift of

2 or

.
3. MSK waveform does not have amplitude variations, whereas QPSK signal have abrupt amplitude
variations.
4. The main lobe of MSK is wider than that of QPSK. Main lobe of MSK contains around 99% of signal
energy whereas QPSK main lobe contains around 90% signal energy.
5. Side lobes of MSK are smaller compared to that of QPSK. Hence interchannel interference is
significantly large in QPSK.
6. To avoid interchannel interference due to sidelobes, QPSK needs bandpass filtering, where as it is not
required in MSK.
7. Bandpass filtering changes the amplitude waveform of QPSK because of abrupt changes in phase. This
problem does not exist in MSK.
The distance between signal points is same in QPSK as well as MSK. Hence the probability of error is also same.
6. a.
ANS -6. a.
Geometric interpretation of signal:Using N orthonormal basis functions we can represent M signals as
n

Si (t) =

j=1

ij

j (t) 0 t T

i = 1, 2 ,.., M

Coefficients are given by

T

Sij =

S
0

(t)

(t) dt

i = 1, 2,.., M
j = 1, 2,..., N

Digital Communication 6th Sem B.E. June 2012 Solved

Given the set of coefficients {sij}, j= 1, 2, .N operating as input we may use the scheme as shown in fig(a) to
generate the signal si(t) i = 1 to M. It consists of a bank of N multipliers, with each multiplier supplied with its own
basic function, followed by a summer.

Conversely given a set of signals s i(t) i = 1 to M operating as input we may use the scheme shown in fig below to
calculate the set of coefficients {sij}, j= 1, 2, .N.

Digital Communication 6th Sem B.E. June 2012 Solved

The

vector

si

is

called

signal

vector

We may visualize signal vectors as a set of M points in an N dimensional

Euclidean space, which is also called signal space.
The squared-length of any vector si is given by inner product or the dot
product of si.

Where sij are the elements of si.

Two vectors are orthogonal if their inner product is zero.
The energy of the signal is given by
T

Ei =

S
0

2
i

(t) dt

Substituting the value si(t) from

We get

Digital Communication 6th Sem B.E. June 2012 Solved

Interchanging the order of summation and integration

Since

(t) forms an orthonormal set, the above equation reduce to

Ei =

S
j=1

ij

This shows that the energy of the signal si(t) is equal to the squared-length of the signal vector si.
The Euclidean distance between the points represented by the signal vectors si and sk is

6. b.
ANS -6. b.

Let rearrange the above Figure as shown below.

We first observe that s1(t), s2(t) and s3(t) are not linearly independent because s3(t) = s2(t) + s1(t)

Digital Communication 6th Sem B.E. June 2012 Solved

The energy of s1(t) is
2

E1 =

32
0

dt = 9 * 2= 18

1(t) =

The first basis function is therefore

2

Define: s21 =

s 1(t)
E 1

3
18

3
=

3 2

.=

1
0 t 2
2 . For

s
0

0 12

1(t) dt =

(t)

dt =0

The Energy of the signal s2(t) is

4

E2 =

32
2

dt = 9 * 2 = 18

The second basis function

2(t) is therefore

30
180

3
=

3 2

1
2

For

2 t 4

Coefficient s31: s31 =

s
0

(t)

1(t) dt =

3
2 = 3 2
2
4

Coefficient s32:

s32 =

s
2

(t)

2(t) dt = 3 2

Intermediate function

g3(t) = -3 for

0 t 4

The third basis function

3
=

9 dt

= -3/6 =

1
2

For

The corresponding orthonormal functions are shown in the figure below:

0 t 4

dt
31
2
0

Digital Communication 6th Sem B.E. June 2012 Solved

Representation of the signals

s1(t) =

3 2

1(t)

s2(t) =

3 2

2(t)

s3(t) =

3 2

1(t) + 3 2

2(t) + 6 3(t)

The signal constellation diagram is shown below:

7. a.
ANS -7. a.

Digital Communication 6th Sem B.E. June 2012 Solved

Without loss of Generality, assume that the signal s(t) =a 0p(t) is normalized so that Ep =

|ao|

and thus Es = E[

]. If a0 is a binary random variable with values a 0

random variable with mean either A or B and variance

is:

a^o = A iff

Assuming B

a^o = A iff

, else set

( A)2 <(B)2 or

<

A+ B
2

A, the probability of a decision error given a0 = A is

where the complementary error function erfc(x) is defined as

erfc (x) =

2
e d

z
2

erfc (-x) = 1-erfc(x) =

= N0/2, i.e.,

Substituting the conditional pdfs yields:

Decide

2
e d .

The probability of a decision error given a0 = B is computed similarly as

=1

{A, B}, then b0 is a Guassian

The decision rule for a maximum likelihood (ML) receiver upon receiving b0 =
Decide

|p()2|

a^o = B.

Digital Communication 6th Sem B.E. June 2012 Solved

The probability of a symbol error for an AWGN channel with matched filter receiver an ML decision rule and a 0

{A, B} is therefore

,
i.e., it depends only on the distance
spectral density

|B A|

between the possible values of a0 and the (2-sided) noise power

No/ 2 . If antipodal signaling is used (e.g., for BPSK or QPSK modulation of a carrier) then B =

-A which implies

|B A|=2| A|=2 Eb

where

Eb= A 2 P ( ao= A ) +B 2 P(ao=B) ,

That is Eb is the bit energy.

Uncoded antipodal (a0

{ A ,+ A } ) signaling. The probability of bit error on an AWGN channel with SNR

Eb/No and a matched filter receiver with ML decision rule is

,
Where Eb =
Coded antipodal (a0

A 2 is the energy per bit.

{ A ,+ A } ) signaling. The probability of error between two binary codewords

Hamming distanced apart on an AWGN channel with SNR Ec/No per code bit and a matched filter receiver with
ML decision rule is
,
Where Ec =

A 2 is the energy per coded bit.

7. b.
ANS -7. b.
For an AWGN channel and for the case when the transmitted signals are equally likely,
the optimum receiver consists of two subsystems as shown in figure below.

Digital Communication 6th Sem B.E. June 2012 Solved

1.

(a) Detector
The detector part of the receiver is as shown in fig (a). It consists of a bank of M product-integrator or
correlators supplied with a set of orthonormal basis function 1(t) ,2(t) .M(t) that are generated
locally.
This bank of correlator operate on the received signal x(t) to produce observation vector x.

2.

(b) Vector Receiver

The 2nd part of the receiver, namely the vector receiver is as shown in fig (b).
The vector x is used to produce an estimate

m
^ of the transmitted symbol mi, where i= 1,2, . , M to

minimize the average probability of symbol error.

The N elements of the observation vector x are first multiplied by the corresponding N elements of each of
the M signal vectors s1, s2 sM , and the resulting products are successively summed in accumulator to
form the corresponding set of Inner products {(x, s k)} k= 1, 2 ..M. The inner products are corrected for the
fact that the transmitted signal energies may be unequal.
Finally, the largest in the resulting set of numbers is selected and a corresponding decision on the

transmitted message made.

The optimum receiver is commonly referred as a correlation receiver.

Digital Communication 6th Sem B.E. June 2012 Solved

8. a.
ANS 8. a.
The definition of SS (Spread Spectrum) may be stated in two parts:
1.

It is a means of transmission in which the transmitted data sequence occupies a larger bandwidth then the

2.

minimum bandwidth necessary to send the data.

Spreading of data is done before transmission through the channel using a code which is independent of
data sequence. The same code is used at the receiving end to despread the received signal so that original
data may be recovered.

Direct Sequence Spread Spectrum with coherent binary Phase shift Keying:The transmitter involves two stages of modulation

In 1st stage, the data sequence b(t) is modulated with the code sequence c(t). So the Spread signal is
m(t) =b(t). c(t)
In 2nd stage, the Spread signal m(t) is modulated with the binary PSK modulator.

When the polarity of b(t) & c(t) are same, the product b(t). c(t) = 1, hence the phase of the BPSK signal is
(2

fct) radians.

Similarly when b(t) & c(t) are of different polarities, the product b(t). c(t) = -1, hence the phase of the
BPSK signal is (2

fct + ) radians.

Digital Communication 6th Sem B.E. June 2012 Solved

Figure.1: Direct-sequence spread coherent phase-shift keying. (a) Transmitter. (b) Receiver.

Figure.2: (a) Product signal m(t) = c(t). b(t). (b) Sinusoidal carrier. (c) DS/BPSK

Digital Communication 6th Sem B.E. June 2012 Solved

The receiver consists of two stages of demodulation.

In 1st stage demodulation, the received signal y(t) & a locally generated replica of the PN sequence are

applied to a multiplier.
In 2nd stage, m(t) is despread by multiplying it by c(t) i.e. it consists of a coherent detector, the o/p which
provides an estimate of the original data sequence.

8. b.
ANS 8. b.
Properties of PN Sequence
Randomness of PN sequence is tested by following properties
1. Balance property
2. Run length property
3. Autocorrelation property
1. Balance property
In each Period of the ML (Maximum Length) - sequence, the number of 1s is always one more than the number of
0s (i.e. number of 1s exceeds the number of 0s by one).

2. Run length property

A run is defined as a subsequence of identical symbols within the ML-Sequence. The length of the Subsequence is
known as the run-length.

The total number of runs =

( N +1)
2

Among the runs of ones and zeros in each period, it is desirable that about one half the runs of each type are of
length 1, one- fourth are of length 2 and one-eighth are of length 3 and so-on.

Digital Communication 6th Sem B.E. June 2012 Solved

3. Auto correlation property

Auto correlation function of a maximal length sequence is periodic and binary valued.
Autocorrelation sequence of binary sequence in polar format is given by

where,
N is the length or period of the PN sequence &
K is the lag of the autocorrelation sequence &

8. c.
ANS -8. c.
Sl.No
1)
2)
3)
4)
5)
6)
7)
8)

Slow FH SS
Multiple Symbols are transmitted in one frequency hop
Symbol rate = Chip rate (Rs = Rc)
Hop rate is lower than Symbol rate (Rh Rs)

Fast FH SS
Multiple hops are taken to transmit one symbol
Hop rate = Chip rate (Rh = Rc)
Hop rate is higher than Symbol rate (R h

One or more Symbols are transmitted over the same carrier

Rs)
One symbol is transmitted over multiple

frequency
Less secure than fast FH-SS

carriers in different hops

More secure than slow FH-SS

PG =

2K

PG =

2K

Several modulation symbols per hop

Shortest uninterrupted waveform in the system is that of

Several frequency hops per modulation

Shortest uninterrupted waveform in the system

data symbol

is that of hop