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NH-7 A.K.SAMUTHIRAM, NAMAKKAL

TWO MARK Q&A

Sem/Branch : VI/ECE-A&B

Subject : Digital communication

Prepared by : R.Shankar, Lecturer/ECE

Let the signal be band limited to ‘w’ Hz. Then nyquist rate is given as, Nyquist=2w samples/sec.

Aliasing will not take place if sampling rate is greater than nyquist rate.

Aliasing effect take place when sampling frequency is less than nyquist rate under such condition,

the spectrum of the sampled signal overlaps with itself. Hence higher frequency components are called

aliasing effect.

3. Define PWM.

PWM is basically width of the pulse changes according to amplitude of the modulating signal. It

is also referred to as pulse duration modulation or PDM.

Sampling theorem states that a band limited signal of finite energy, which has no frequency

components higher than W Hz, is completely described by specifying the values of the signal at the

instants of time separated by 1/2w seconds.

i. Band width requirement of DPCM is less compared to PCM.

ii. Quantization error is reduced because of prediction filter.

iii. Numbers of bits used to represent one sample value are also reduced compared to PCM.

Dm encodes the input sample by only one bit. It sends the information about +Ѕ or -Ѕ i.e. step rise

or fall. DPCM can have more than one bit for encoding the sample. It sends the information about

difference between actual sample value and predicated sample value.

The message can be recovered from PAM by passing signal through reconstruction filter. The

reconstruction filter integrates amplitudes of PAM pulses. Amplitude smoothing of reconstructed

signal is done to remove amplitude discontinues due to pulses.

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DIGITAL COMMUNICATION

8. Write an expression for band width of binary PCM with N messages with maximum frequency of

FmHz ?

If ‘v’ number bits are used to code each input sample then bandwidth of PCM is given as

BT >=N.v.Fm .Here v.fm is the bandwidth required by one message.

The PDM signal is given as clock signal to monostable multivibrator. The multivibrator triggers on

the falling edge .hence the PPM pulse width is produced after falling edge of PDM pulse. PDM signal

represent the input signal amplitude in the form of width of pulse. A PPM pulse after this width of

PDM pulse.

10. Mention the use of adaptive quantizer in adaptive digital wave form coding scheme.

Adaptive quantizer changes its step size according to variance of input signal. Hence quantization

error is reduced. ADPCM uses adaptive quantization. The bit rate of such schemes reduced due to

adaptive quantization.

In adaptive coding quantization step size and prediction filter co-efficient are changed as per

properties of input signals. This quantization error and number of bits used to represent the sample

value. Adaptive coding is used at low bit rates.

Delta modulation uses one bit to encode one sample. Hence bit rate of delta modulation is low

compared to PCM.

13. Define Dirac comb or ideal sampling function. What is its Fourier Transform?

Dirac comb is nothing but a periodic impulse train in which the impulses

are spaced by a time interval of Ts seconds. The equation for the function is given

By

14. Give the interpolation formula for the reconstruction of the original signal g(t)

From the sequence of sample values {g(n/2W)}.

N is the number of samples.

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DIGITAL COMMUNICATION

If a finite –energy signal g(t) contains no frequencies higher than W hertz, it is completely

determined by specifying its co=ordinates at a sequenc of points spaced 1/2W seconds apart.

If a finite energy signal g(t) contains no frequencies higher than W hertz, it may be completely

recovered from its co=ordinates at a sequence of points spaced 1/2W seconds apart.

Quadrature sampling is used for uniform sampling of band pass signals

Consider

The in-phase component gI(t) and the quadrature component gQ(t) may be

respectively and then suppressing the sum-frequency components by means of ppropriate low pass filter.

Under the assumption that fc>W, we find that gI(t)&gQ(t) are both low-pass signals limited

to -W<f<W. Accordingly each component may be sampled at the rate of 2W samples per

second. This type of sampling is called quadrature sampling.

17. Give the expression for aliasing error and the bound for aliasing error.

Where,

E is the aliasing error.

|G(f)| is the amplitude spectrum of the signal g(t)

Where,

E is the bound for aliasing error.

Pulse code modulation (PCM) is a method of signal coding in which the message signal is

sampled, the amplitude of each sample is rounded off to the nearest one of a finite set of discrete

levels and encoded so that both time and amplitude are represented in discrete form. This allows the

message to be transmitted by means of a digital waveform.

The conversion of analog sample of the signal into digital form is called Quantizing

process.

1. The peak-to-peak range of input sample values subdivided into a finite set of decision

levels or decision thresholds.

2. The output is assigned a discrete value selected from a finite set of representation levels

are reconstruction values that are aligned with the treads of the staircase.

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DIGITAL COMMUNICATION

Idle channel noise is the coding noise measured at the receiver output with Zero transmitters

input.

The difference between the actual samples of the process at the time of interest and the

predictor output are called a prediction error.

Delta modulation is the one-bit version of differential pulse code modulation.

The performance of a delta modulator can be improved significantly by making the step size of

the modulator assume a time- varying form. In particular, During a steep segment of the input signal

the step size is increased. Conversely, When the input signal is varying slowly, the step is reduced , In

this way, the step size is adapting to the level of the signal. The resulting method is called adaptive

delta modulation (ADM).

1. Mid tread type quantizer.

2. Mid riser type quantizer.

Origin of the signal lies in the middle of a tread of the staircase.

Origin of the signal lies in the middle of a riser of the staircase

Quantization error is the difference between the output and input values of quantizer.

Step size is not uniform. Non-uniform quantizer is characterized by a step size that increases as

the separation from the origin of the transfer characteristics is increased. Non-uniform quantization is

otherwise called as robust quantization.

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DIGITAL COMMUNICATION

30. Draw the quantization error for the mid tread and mid-rise type of quantizer.

31. What is the disadvantage of uniform quantization over the non-uniform quantization?

SNR decreases with decrease in input power level at the uniform quantizer but non uniform

quantization maintains a constant SNR for wide range of input power levels. This type of

quantization is called as robust quantization.

The signal is compressed at the transmitter and expanded at the receiver. This is called as

companding. The combination of a compressor and expander is called a compander.

33. Draw the block diagram of compander? Mention the types of companding?

Block diagram:

PAM is the pulse amplitude modulation. In pulse amplitude modulation, the amplitude of a

carrier consisting of a periodic train of rectangular pulses is varied in proportion to sample values of a

message signal.

35. What is the need for speech coding at low bit rates?

The use of PCM at the standard rate of 64 Kbps demands a high channel bandwidth for its

transmission, so for certain applications, bandwidth is at premium, in which case there is a definite

need for speech coding at low bit rates, while maintaining acceptable fidelity or quality of

reproduction.

It means adaptive differential pulse code modulation, a combination of adaptive quantization

and adaptive prediction. Adaptive quantization refers to a quantizer that operates with a time varying

step size. The autocorrelation function and power spectral density of speech signals are time varying

functions of the respective variables. Predictors for such input should be time varying. So adaptive

predictors are used.

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DIGITAL COMMUNICATION

Sampler, Quantizer, Encoder, Modulator, Decoder, Channel, Demodulator, Reconstruction

Filter are the components of a Digital Communication System.

Ruggedness to channel noise and other interferences. Flexible implementation of digital

hardware system .Coding of digital signal to yield extremely low error rate and high fidelity. Security

of information.

39. What are the different types of PTM systems?

There are two kinds of PTM schemes they are: Pulse duration or pulse width or pulse length

modulation (PDM (Or) PWM (Or) (PLM) Pulse position modulation

The method in which the samples of the message signal are used to vary the duration width

of the individual pulses. This is referred to as pulse duration modulation.

In PPM, the position of a pulse relative to its unmodulated time of occurrence is varied in

accordance with the message signal.

It is the modulation in which the message and also the carrier are in discrete form. These

are classified as

Pulse code modulation, Delta modulation

Ruggedness to transmission noise and interference

Efficient regeneration of the coded signal along the transmission path.

The possibility of a uniform format for the different kinds of baseband signals.

The performance of a PCM system is influenced by two major sources of noise.

Transmission Noise: Which is introduced anywhere transmitter output and the receiver input. It is

also named as channel Noise.

Quantizing noise: This is introduced in the transmitter and is carried along to the receiver output.

45. What delta modulation and give the comparison between DM and DPCM?

Delta modulation is one bit (or)two level version of DPCM.

They are similar except for two important difference namely, the use of one bit quantizer in

delta modulator and the prediction filter is replaced by a single delay element.

In delta modulation we observe quantization noise .there are two major sources of quantizing error

in DM systems. they are

Slope over load distortion , Granular noise

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DIGITAL COMMUNICATION

Multiplexing may be defined as a technique which allows many user to share a

common communication channel simultaneously. There are two major types of multiplexing

techniques; Frequency division multiplexing(FDM),Time division multiplexing(TDM).

There are basically two types of multiplexer

Low speed multiplexer

High speed multiplexer

The band pass signal x(t) whose maximum band width is 2w can be completely represented

into and recovered from its samples. if it is sampled at the minimum rate of twice the bandwidth.

1. slope overload distortion.

It occurs due to limited step size and fast variation in the signal.

2. Granular noise

It occurs due to too large step size and very small amplitude variations in the input signal.

1. with linear quantization, the signal to quantization noise ratio reduces at low signal levels.

2.compression uses nonlinear quantization. It improves the signal to quantization noise ratio at low

level signals.

52. A band pass signal has the spectral has the spectral range that extends from 20khz to

82khz. Find the acceptable range of sampling frequency fs?

Here bandwidth,

Bw = 82khz-20khz

= 62 khz

fs= 2×bw

= 2×62khz = 124khz

53. what is the SNR of PCM system if number of quantization levels is 28?

Quantization levels= 2υ =28

υ=8

(s/n)db= 4.8+6υdb

=4.8+6×8

=52.8db

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DIGITAL COMMUNICATION

54.What is correlative coding?

Correlative level coding is used to transmit a base band signal with the signaling rate of 2Bo the

Chanel of bandwidth Bo. This made physically possible by allowing ISI transmitted controlled manner.

This ISI is known to the receiver. The correlative coding is implemented by duo binary signally and

modified duo binary signaling.

55. What is an intersymbol interference in base band binary PAM system?

In base band binary PAM symbol are transmitted one after another. These symbols are separated

by sufficient time duration. The transmitter channel and receiver acts as filter to this base band data.

Because of the filtering characteristics transmitted PAM pulses are spread in time. Let the transmitted

wave form be represented as ,

∞

X(t)= ∑ AK g(t-kTb)

k=-∞

56. Define duobinary base band PAM system.

Duo binary encoding reduces the maximum frequency of the base band signal. The word ‘duo’

means to double the transmission capacity of the binary system. Let the PAM signal ak represents kth

bit. Then the encoder generates the new wave form as

Ck=ak+ ak-1

Thus two successive bits are added to get encoded value of kth bit. Hence Ck becomes a

correlated signal even though ak is not correlated. This introduces intersysmbol in the controlled manner

to reduced the band width .

The output of duo binary encoder is given as

Ck=ak+ ak-1

Let ak represents an estimate of ak at the decoder. Then above equation becomes Ck=ak+ ak-

1.Hence ak can be obtain as ak=ck+ ak-1 .This shows that if c k is received with error then ak will have

error. This error will propagate the output sequence. This is the main draw back of duo binary encoding.

Eye pattern is used to study the effect of ISI base band transmission

i. Width of eye opening defines the interval over which the received wave can be sampled without

error from ISI.

ii. The sensitivity of the system to timing error is determined by the rate of closer of the eye has the

sampling time is varied.

iii. Height of the eye opening at sampling time is called margin over noise.

59. What is the value of maximum signal to noise ratio of the matched filter? When it become maximum.

Maximum signal to noise ratio of the matched filter is the ratio of energy of the signal to

psd of white noise

Pmax=E/No/2

This maximum value of occurs at the end of bit duration Tb.

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DIGITAL COMMUNICATION

AQF: Adaptive quantization with forward estimation. Unquantized samples of the input

signal are used to derive the forward estimates.

AQB: Adaptive quantization with backward estimation. Samples of the quantizer outputs

are used to derive the backward estimates.

APF: Adaptive prediction with forward estimation, in which unquantized samples of the

input signal are used to derive the forward estimates of the predictor coefficients.

APB: Adaptive prediction with backward estimation, in which Samples of the quantizer

output and the prediction error are used to derive estimates of the predictor coefficients.

61. What are the limitations of forward estimation with backward estimation?

Side information

Buffering

Delay

For the adaptation of the predictor coefficients the least mean square (LMS) algorithm is used.

It is a frequency domain coder, in which the speech signal is divided in to number of sub bands

and each one is coded separately. It uses non masking phenomenon in perception for a better speech

quality. The noise shaping is done by the adaptive bit assignment.

In the context of speech production the formant frequencies are the resonant frequencies of the

vocal tract tube. The formants depend on the shape and dimensions of the vocal tract.

Nfs= (MN) (fs/M)

Nfs->bit rate

M->number of sub bands of equal bandwidths

N->average number of bits

fs/M->sampling rate for each sub band

It is a nonlinear estimator that provides an estimate of some desired response without

requiring knowledge of correlation functions, where the filter coefficients are data dependent. A

popular filtering algorithm is the LMS algorithm.

Data signalling rate is defined as the rate measured in terms bits per second (b/s) at which

data are transmitted.

Data signaling rate Rb=I/Tb Where Tb=bit duration.

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DIGITAL COMMUNICATION

It is defined as the rate at which signal level is changed depending on the nature of the format

used to represent the digital data. It is measured in bauds or symbols per second.

In this format binary 0 is represent by no pulse and binary 1 is represented by the positive pulse.

Binary 1 is represented by a positive pulse and binary 0 is represented by a negative pulse.

Binary 0 is represented by no pulse and binary one is represented by the alternative positive and

negative pulse.

Binary the first half bit duration negative pulse and the second half bit duration positive

pulse.

Binary first half bit duration positive pulse and the second half bit duration negative pulse.

It defines the time interval over which the received waveform can be sampled without error

from intersymbol interference.

The sensitivity of the system to timing error is determined by the rate of closure of the eye as the

sampling time is varied.

The height of the eye opening at a specified sampling time defines the margin over noise.

The transmitted signal will undergo dispersion and gets broadened during its transmissio

through the channel. So they happen to collide or overlap with the adjacent symbols in the

transmission. This overlapping is called Inter Symbol Interference.

The eye pattern is obtained by applying the received wave to the vertical deflection plates of

an oscilloscope and to apply a saw tooth wave at the transmitted symbol rate to the horizontal

deflection plate.

78. Mention the need of optimum transmitting and receiving filter base band data transmission.

When binary data is transmitted over base band channel noise interference with it. Because of

this noise interference error introduced in signal detection. Optimum filter perform to functions while

receiving the noisy signal.

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DIGITAL COMMUNICATION

i. Optimum filter integrate a signal during the bit interval and check the output at the time

instant where signal to noise ratio is maximum.

ii. Transfer function of the optimum filter is selected so as to maximize signal to noise ratio.

iii. Optimum filter minimizes the probability of error.

iv.

79. How is eye pattern obtained on the CRO?

Eye pattern can be obtained on the CRO by applying the signal tonone of the input channels

giving an external trigger of (1/Tb)hz.this makes one sweep of beam equal to ‘Tb seconds.

Zero ISI can be obtained if the transmitted pulse satisfies the following condition:

Time domain: Þ[(i-k)]= {1 for i=k

0 for i≠k

Frequency domain: ∞

= ∑ p (f - nfb) =Tb

k=-∞

81. From the eye pattern, how is the best time for sampling determined?

It is preferable to sample the instant at which eye is open widest. At this instant chances of error are

minimum.

Eye pattern can be used for:

1. To determined an interval over which the received wave can be sampled without error due to ISI.

2. To determine the sensitivity of the system to timing error.

3. The margin over the noise is determined from the eye.

In a switched telephone network the distortion deponds upon

1. Transmission characteristics of individual links.

2. Number of links in connection.

Hence fixed pair of transmit and receive filters will not serve the equalization problem. The

transmission characteristics keep on changing. The adaptive equalization is used.

P(t)=sin(2πB ot)/2πBot

P(f)={1/2B0 for –b0<f<b0

0 for elsewhere

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DIGITAL COMMUNICATION

85. Define ASK.

In ASK carrier is switched on when binary ‘1’ is to be transmitted and it is switched off when

binary ‘0’ transmitted ASK is also called on-off keying.

In DPSK input sequence is modified. Let input sequence be d(t) and output sequence b(t). How b(t)

changes level at the beginning of each interval in which d(t)=1 and it does not changes level when

d(t)=0.

When b(t) changes level phase of the carrier is changed. And as started above b(t) changes it level

only if d(t)=1. Hence the technique is called differential PSK.

In coherent detection the local carrier generated at the receiver phase locked with carrier at the

transmitter. The detection is done by correlating received noisy signal and locally generated carrier.

The coherent detection is a synchronous detection.

In PSK phase of the carrier is switched according to input bit sequence .In FSK frequency of the

carrier is switched according to input bit sequence.FSK needs double of the bandwidth of PSK .

In coherent ASK correlation is used to detect the signal. Locally generated `carrier is correlated

with incoming ASK signal. The locally generated carrier is in exact phase with the transmitted carrier.

Coherent ASK is also called synchronous ASK.

90. What is the major advantage of coherent PSK over coherent ASK?

ASK is on off signaling where as the modulated carrier is continuously transmitted in PSK. Hence

peak power requirement is more in ASK whereas it is reduced in case of PSK.

The signal to noise ratio of the matched filter depends only upon the the ratio of the signal

energy to the psd of white noise at the filter input 1) The output signal of a matched filter is

proportional to a shifted version of the autocorrelation function of the input signal to which the filter

is matched.

Consider a message signal m. The task of transforming an incoming message mi=1, 2,…..M, into

a modulated wave si(t) may be divided into separate discrete time & continuous time operations. The

justification for this separation lies in the Gram-Schmidt orthogonalization procedure which permits

the representation of any set of M energy signals, {si(t)}, as linear combinations of N orthonormal

basis functions, where N ≤M.

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DIGITAL COMMUNICATION

A filter whose impulse response is a time reversed & delayed version of some signal πj (t) then

it is said to be matched to πj (t) correspondingly, the optimum receiver based on the detector is referred

to as the matched filter receiver.

Maximum likelihood detector computes the metric for each transmitted message compares

them and then decides in favor of maximum. The device for implementing the decision rule i.e.; set ^m

= mi if In [ fx(x/mk)] is maximum for k=i is called maximum –likelihood detector and the decision rule

is called maximum likelihood.

A pair of sinusoidal signals that differ only in a phase shift of 180 degrees are referred to as

antipodal signals.

96. Explain how QPSK differs from PSK in term of transmission bandwidth and bit Information it

carries?

For a given bit rate 1/Tb, a QPSK wave requires half the transmission bandwidth of the

corresponding binary PSK wave. Equivalently for a given transmission bandwidth, a QPSK wave carries

twice as many bits of information as the corresponding binary PSK wave.

97. Give the equation for average probability of symbol error for coherent binary

PSK.

Average probability of signal error,

Pe = 1 / 2 erfc√Eb / No

QPSK is Quadriphase –shift keying. In QPSK the phase of the carrier takes on one of the four

equally spaced values Such as π/4 , 3π/4, 5π/4 and 7π/4.

A unique pair of bits is called a dibit. Gray encoded set of dibits 10, 00, 01 & 11

Si(t) = {√2Eb/Tb Cos(2πf i t) , 0 ≤ t ≤Tb

O, elsewhere

fi = nc+ i/ Tb

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1. Differential encoding of the input binary wave

2. Phase –shift keying hence, the name differential phase shift keying.

The parameter h is defined by

h= Tb(f1-f2)

h is deviation ratio , measured with respect to bit rate 1/Tb.

S(t) = √2Eb/Tb Cos [ 2πf 1t + θ(0)] for symbol 1

√2Eb/Tb Cos [ 2πf 2 t + θ(0)] for symbol 0

Nominal carrier frequency is the arithmetic mean of the two frequencies f1 and f2 and

it is given as fc = ½ (f1 + f2) Where f1 is the frequency for symbol –1 f2 is the frequency for

symbol – 0

The carrier synchronization is required in coherent detection methods to generate a

coherent reference at the receiver. In this method the data bearing signal is modulated on the carrier in

such a way that the power spectrum of the modulated carrier signal contains a discrete component

at the carrier frequency.

1. Carrier synchronization

2. Symbol & Bit synchronization

3. Frame synchronization.

Carrier synchronization using Mth Power loop

Costas loop for carrier synchronization

In a matched filter or correlation receiver, the incoming signal is sampled at the end of

one bit or symbol duration. Therefore the receiver has to know the instants of time at which a

symbol or bit is transmitted. That is the instants at which a particular bit or symbol status and

when it is ended. The estimation of these times of bit or symbol is called symbol or bit

synchronization.

110. What are the two methods of bit and symbol synchronization?

Closed loop bit synchronization

Early late gate synchronizer

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If there is a long string of 1’s and o’s then y(t) has no zero crossings and

synchronization may be lost. If zero crossing of y(t) are not placed at integer multiples of Tb, the

Synchronization suffers from timing Jitter.

112. In minimum shift keying what is the relation between the signal frequencies and

bit rate?

Let the bit rate be fband frequency of carrier be f0.the higher and lower MSK signal frequncies

are given as,

fH= f0+(fb/4)

fL=f0+(fb/4)

113. Write the expression for bit error rate for coherent binary FSK?

The bit error rate of cocherent binary FSK is given as

Pe=1/2erfc√(0.6/n0)

Error probability of MSK:

P e=1/2erfc√(eb/n0)

Error probability of DPSK:

P e= 1/2e(-eb/n0)

115. Highlight the major difference between a QPSK signal ana a MSK signal.

MSK signal have continious phase in all the cases, whereas QPSK signal has abrupt

phase shift of π/2 or π.

BPSK:

Pe= 1/2 erfc√(eb/n0)

BFSK:

Pe=1/2 erfc √(0.6e b/4n 0)

For the fixed value of eb/n 0, error probability of BPSK is less than BFSK.

For the given probability of error, the eb/n 0 of BPSK is 3db less compared to that

BFSK.

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117. What are the error detection and correction capabilities of hamming codes?

The minimum distance (dmin) of hamming codes is ‘3’. Hence it can be used to detect double errors

or correct signal error. Hamming codes are basically linear block codes with dmin=3.

118. How syndrome is calculated in hamming codes and cyclic codes?

In hamming codes the syndrome is calculated as,

S=YHT

Here Y is the received and HT is the transpose of parity check matrix. In cyclic codes the syndrome

vector polynomial is given as,

S(p)=rem[Y(p)/G(p)]

Here Y(p) is received vector polynomial and G(p) is generated polynomial .

BCH codes are most extensive and powerfull error correcting cyclic codes. The decoding of BCH codes

is comparatively smaller. For any positive inter ‘m’ and ‘t’ (where t<2m-1) there exists a BCH code with

following parameter:

Block length:N=2m-1

Number of parity bits: n-k<= mt.

Minimum distance:

dmin>=2t+1.

These are non binary BCH codes. The encoder for RS codes operates on multiple bits simultaneously.

The (n,k) RS code take the group of m-bit symbols of the incoming binary data stream. It takes such ‘k’

number of symbols in one block. Then the encoder adds (n-k) redundant symbols to form the code word

of ‘n’ symbols.

Rs codes have:

Block length: N=2m-1 symbols

Message size: K symbols

Parity checks size: N-k=2t symbols

Minimum distance: d min=2t+1 symbols.

121. What is the difference between block code and convolution code?

Block code take ‘k’ number of message bit simultaneously and from n bit code vector. This code

vector is also called block. Convolutional code takes one message bit at a time and generates two or

more encoded bit. Thus convolutional code generates a string of encoded bit for input message string.

Free distance is the minimum distance between code vectors. It is also equal to minimum weight of code

vector. Coding gain is used as bases of comparison for different coding method. To achieve the same bit

error rate the coding gain is defined as

A=(Eb /No) encoded/ (Eb /No) coded

For convolutional coding gain is given as A=rdf/2

A code is linear if the sum of any two code vectors produces another code vector.

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DIGITAL COMMUNICATION

Code rate is the ratio of message bits (k) and the encoder output bits (n). It is defined by r (i.e.) r= k/N

125. Define code efficiency.

It is the ratio of message bits in a block to the transmitted bits for that block by the encoder i.e.

The hamming distance between two code vectors is equal to the number of elements in which they

differ. For example let the two code vectors be X= (101) and Y= (110).These two code vectors differ in

second and third bits. Therefore the hamming distance between x and Y is two.

In a systematic block code, message bit appear first and then check bits. In the non-systematic code,

message and check bits cannot be identified in the code vector.

128. What are the conditions to satisfy the hamming code?

1) No. of Check bits q ≥3

2) Block length n = 2 q –1

3) No of message bits K = n-q

4) Minimum distance d min =3

129. Define code word & block length.

The encoded block of ‘n’ bits is called code word. The no. of bits ‘n’ after coding is called block length.

( n,k) Linear block code for which the minimum distance equals n – k + 1 is called maximum distance

separable codes. For RS code minimum distance equals n – k + 1 so it is called as maximum distance

separable code

Golay code is the (23, 12) cyclic code whose generating polynomial is,

G(p) =P11+P9+p7+P6+p5+p+1

This code has a minimum distance of dmin=7. This code can correct up to 3 errors. It is perfect code.

1. Encoders and decoders for cyclic codes are simple

2. Cyclic codes also detect error burst that span many successive bits.

The non-zero output of the produce Yh t is called syndrome and it is used to detect the errors in y.

syndrome is denoted by’s’ and it is given as,

S=YHt

134. What is convolutional code?

Fixed number of inputs bits are stored in the shift register and they are combined with the help of

mod-2 adders.this operation is equivalent to binary convolution and hence it is called convolution

coding.

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DIGITAL COMMUNICATION

135. What is pseudo noise sequence?

Pseudo noise sequence is a noise like high frequency signals. The sequence is not completely

random but it is generated by well defined logic. hence it is called Pseudo noise sequence. Pseudo noise

sequence is used in spread spectrum communication for spreading message signal.

Processing gain =bandwidth of spreaded signal\ bandwidth of unspreaded signal

For DS-SS processing gain is given as PG=Tb\Tc

Tb=bit period of data sequence

Tc=bit period of PN sequence

And for FH-SS processing gain is given as,

PG=2t Here ‘t’ is the number of bit in PN sequence.

A pseudo-noise sequence is defined as a coded sequence of 1s and Os with certain

autocorrelation properties. It is used in spread Spectrum communications. It is periodic in that a

sequence of 1s and 0s repeats itself exactly with a known period.

‘000’ is not a state of the shift register sequence in PN sequence generator, since this results in a

catastrophic cyclic code i.e. once the 000 state is entered, the shift register sequence cannot leave this

state.

A random binary sequence is a sequence in which the presence of a binary symbol 1 or 0 is

equally probable.

In each period of a maximum length sequence, the number of 1s is always one more than the

number of 0s. This property is called the balance property.

Among the runs of 1s and 0s in each period of a maximum length sequence, one half the runs of

each kind are of length one, one fourth are of length two, one eighth are of length three, and so or as

long as these function represent meaningful numbers of runs. This property is called the run property.

The autocorrelation function of a maximum length sequence is periodic and binary valued. This

property is called the correlation property.

An important attribute of spread-spectrum modulation is that it can provide protection against

externally generated interfering (jamming) signals with finite power. The jamming signal may consist of

a fairly powerful broadband noise or multitone waveform that is directed at the receiver for the purpose

18

DIGITAL COMMUNICATION

of disrupting communications. Protection against jamming waveforms is provided by purposely making the

information bearing signal occupy a bandwidth far in excess of minimum bandwidth necessary to transmit it.

Processing Gain (PG) is defined as the ratio of the bandwidth of spread message signal to the

bandwidth of unspreaded data signal i.e.

In the frequency band of the interest, somebody else transmits the signals intentionally since

these signals the in the frequency band of transmission, they interface the required signal. Hence it

becomes difficult to detect the required signals. This is called jamming effect.

With the help of spread spectrum method, the transmitted signals are spread over the mid

frequency band. Hence these signals appear as noise. Then it becomes difficult for the jammers to send

jamming signals. This is called anti jamming.

147. What are the three codes used for the anti jamming application?

1. Golay code (24, 12)

2. Expurgated Golay (24, 11)

3.maxmum length shift register code.

In frequency hop spread spectrum, the frequency of the carrier hops randomly from one

frequency to another frequency.

If the symbol rate of MFSK is an integer multiple of hop rate (multiple symbols per hop) then it

is called slow frequency hopping

If the hop rate is an integer multiple of symbol rate (multiple hops per symbol) then it is called

fast frequency hopping.

1. Spread Jammer over the entire measure of the spectrum of Txed signal.

2. Retuning the Jamming signal over the frequency band of Txed signal.

1. It does not require external synchronization networks.

2. CDMA offers gradual degradation in performance when the no. of users is increased But it is

easy to add new user to the system.

3. If offers an external interference rejection capability.

19

DIGITAL COMMUNICATION

The interference caused by the interfacing of the signal form the indirect path with the signal of direct

path is called multipath interference.

The main advantage of spread spectrum technique is its ability to reject interference whether it

be the unintentional interference of another user simultaneously attempting to transmit through the

channel (or) the intentional interference of a hostile transmitter to jam the transmission.

The encoded block of “n” bits is known as code word. It consists of message bits and redundant

bits.

The ratio of message bits(K) and the encoder output bits(n) is known as code rate.Usually code rate is

denoted by r I.e

r=k\n , we find that 0<r<1.

157. Define channel data rate.

Channel data rate is the bit rate at the output of encoder. If the bit rate at the input of encoder is Rs,

then channel data rate be,

Channel data rate (R0)=n\k Rs

20

DIGITAL COMMUNICATION

PART- B QUESTIONS

2. Describe the following systems by presenting appropriate diagrams.

(i) Time Division multiplexing

(ii) delta modulation

3. Draw block diagram of differential PCM and explain the function performed by each block.

4. Explain the process of quantization, encoding and decoding in PCM? In what way differential PCM is

better than PCM?

5. Discuss the basic issues involved in the design of a regenerative repeater for pulse code modulation.

6. With neat sketches, explain the duo binary signaling scheme.

7. with neat sketches, explain the modified duo binary signaling scheme.

8. Write briefly about eye pattern.

9. Derive the Nyquist criterion for distortionless transmission.

10. Discuss on the following:

(i)Baseband M-array PAM transmission(ii) Adaptive equalization

11. Draw the block diagram of QPSK transmitter and coherent QPSK receiver and explain their operation.

12. Draw the block diagram of MSK transmitter and explain the function of each block.

13. With necessary equations and signal space diagram, obtain the probability of error for coherent binary

FSK systems.

14. Explain BPSK signal transmission and coherent BPSK reception with suitable diagrams. Derive an

expression for the probability of symbol error for the scheme.

15. With necessary equations and signal space diagram, explain briefly about FSK system.

16. Obtain probability of error in terms of Eb/No for QPSK.

17. Describe a decoding procedure for linear block code.

18. Write the generator matrix and parity check matrix of (7,4) hamming code.

19. Briefly explain the viterbi decoding algorithm.

20. State and prove the properties of syndrome decoding.

21. Explain the features of RS code.

22. State and explain the properties of maximal length sequence.

23. Draw the block diagram of DS-SS transmitter and receiver and explain the function performed by each

block in brief.

24. Explain the principle of operation of frequency hopped M-ary FSK spread spectrum system.

25. Derive the expression for gain in the SNR obtained by the use of spread spectrum starting from the set

of orthonormal basis function.

26. Derive and plot the power spectra of NRZ uni polar and bipolar format signals.

27. For a (2,1,3) convolution code with g1 = ( 1 0 1 1) and g2 = ( 1 1 1 1), design the encoder and find the

following.

(i) Generator matrix

(ii) Transfer function matrix Compute the coded output using both the methods assuming the input u =

(101101).

28. Derive an expression for Jamming margin for direct sequence spread spectrum system with BPSK

modulation. ii) An PN sequence is generated using a feed back register of length m = 4. The chip rate is

107 chips per sec. Find the length, chip duration and the period of the PN sequence.

21

DIGITAL COMMUNICATION

29. (i) With neat sketches ,explain the frequency hop spread spectrum techniques.

(ii) Discuss the antijam charecteristics.

30. With necessary sketches and expressions, briefly explain about the flattop sampling.(ii) With supportive

derivation prove that if a signal contains no frequency higher than W hertz it may be reconstructed from

its samples at a sequence of points spaced 1/2 W seconds apart.

22

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