Digital Comunication 2 Marks

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DIGITAL COMMUNICATION 1 GNANAMANI COLLEGE OF TECHNOLOGY NH-7 A.K.SAMUTHIRAM, NAMAKKAL DEPARTMENT OF ELECTRONICS&COMMUNICATION ENGG, TWO MARK Q&A Sem/Branch : VI/ECE-A&B Subject : Digital communication Prepared by : R.Shankar, Lecturer/ECE UNIT -1 SAMPLING AND WAVEFORM CODING 1. Define Nyquist rate. 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. 2. What is meant by aliasing effect? 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. 4. State sampling theorem. 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. 5. Mention two merits of DPCM. 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. 6. What is the main difference in DPCM and DM? 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. 7. How the message can be recovered from PAM? 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.

Transcript of Digital Comunication 2 Marks

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GNANAMANI COLLEGE OF TECHNOLOGY NH-7 A.K.SAMUTHIRAM, NAMAKKAL

DEPARTMENT OF ELECTRONICS&COMMUNICATION ENGG,

TWO MARK Q&A Sem/Branch : VI/ECE-A&B Subject : Digital communication Prepared by : R.Shankar, Lecturer/ECE

UNIT -1 SAMPLING AND WAVEFORM CODING

1. Define Nyquist rate. 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. 2. What is meant by aliasing effect? 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. 4. State sampling theorem. 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. 5. Mention two merits of DPCM.

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.

6. What is the main difference in DPCM and DM? 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. 7. How the message can be recovered from PAM?

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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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. 9. How is PDM wave converted into PPM system?

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. 11. What do you understand from adaptive coding? 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. 12. What is the advantage of delta modulation over PCM? 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

The Fourier transform of 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)}.

Where , 2W is the bandwidth

N is the number of samples.

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15. State sampling theorem. 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. 16. Define Quadrature sampling.

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.

18. What is meant by PCM? 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.

19. Define quantizing process.

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

20. What are the two fold effects of 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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21. What is meant by idle channel noise? Idle channel noise is the coding noise measured at the receiver output with Zero transmitters

input.

22. What is meant by prediction error? The difference between the actual samples of the process at the time of interest and the

predictor output are called a prediction error.

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

24. Define adaptive delta 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).

25. Name the types of uniform quantizer.

1. Mid tread type quantizer. 2. Mid riser type quantizer.

26. Define mid tread quantizer. Origin of the signal lies in the middle of a tread of the staircase.

27. Define mid-riser quantizer. Origin of the signal lies in the middle of a riser of the staircase

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

29. What do you mean by non-uniform quantization? 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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30. Draw the quantization error for the mid tread and mid-rise type of quantizer.

For mid tread type:

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.

32. What do you mean by companding? Define compander. 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:

34. What is PAM? 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.

36. Define ADPCM.

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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37. What are the components of a Digital Communication System?

Sampler, Quantizer, Encoder, Modulator, Decoder, Channel, Demodulator, Reconstruction Filter are the components of a Digital Communication System.

38. What are the advantages of Digital Communication?

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

40. What is pulse duration modulation (PDM)? 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.

41. What is pulse position modulation? In PPM, the position of a pulse relative to its unmodulated time of occurrence is varied in

accordance with the message signal.

42. What is pulse digital modulation scheme? 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

43. What are the advantages of digital representation of analog signals? 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.

44. Discuss Noise effect in PCM. 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.

46. Discuss the noise effect in delta modulation? 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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47. Define multiplexing. 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).

48. Give the different types of multiplexer.

There are basically two types of multiplexer Low speed multiplexer High speed multiplexer

49. State band pass sampling theorem? 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.

50. What are the two limitation of delta modulation? 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.

51. Why compressor are used in PCM? 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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UNIT-2 BANDLIMITED SIGNALLING

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 .

57. What is the main disadvantage of duo binary scheme? 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 ck 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.

58. What are eye patterns?

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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60. What is meant by forward and backward estimation?

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

62. How are the predictor coefficients determined?

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

63. Define adaptive sub band coding. 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.

64. What are formant frequencies?

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.

65. What is the bit rate in ASBC?

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

66. Define Adaptive filter.

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.

67. Define data signaling Rate. 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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68. Define modulation rate.

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.

69. State NRZ unipolar format.

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

70. State NRZ polar format. Binary 1 is represented by a positive pulse and binary 0 is represented by a negative pulse.

71. State NRZ bipolar format.

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

72. State Manchester format. 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.

73. What is the width of the eye?

It defines the time interval over which the received waveform can be sampled without error from intersymbol interference.

74. What is sensitivity of an eye?

The sensitivity of the system to timing error is determined by the rate of closure of the eye as the sampling time is varied.

75. What is margin over noise?

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

76. What is Inter symbol interference? 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.

77. How eye pattern is obtained?

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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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.

80. What is the condition for zero inter symbol interference?

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.

82. What is the purpose of using an eye pattern? 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.

83. Why do you need adadtive equalization in a switched telephone network?

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.

84. What is an ideal nyquist channel? P(t)=sin(2πBot)/2πBot Such pulse have the spectrum of, P(f)={1/2B0 for –b0<f<b0 0 for elsewhere

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UNIT-3 PASS BAND DATA TRANSMISSION

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.

86. What is meant by DPSK? 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.

87. Explain coherent detection. 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.

88. What is the difference between PSK&FSK?

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 .

89. What is mean by coherent ASK?

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. 91. What are the properties of matched filter?

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.

92. Why do we go for Gram-Schmidt Orthogonalization procedure?

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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93. What is matched filter receiver?

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.

94. What is maximum likelihood detector?

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.

95. Define antipodal signals.

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

98. Give the signal space characterization of QPSK.

99. Define QPSK. 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.

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

101. Give the transmitted signal of Non-coherent binary FSK. Si(t) = {√2Eb/Tb Cos(2πf i t) , 0 ≤ t ≤Tb O, elsewhere fi = nc+ i/ Tb

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102. Give the two basic operation of DPSK transmitter.

1. Differential encoding of the input binary wave 2. Phase –shift keying hence, the name differential phase shift keying.

103. Define deviation ratio in MSK?

The parameter h is defined by h= Tb(f1-f2)

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

104. Define MSK signal in interval 0 t Tb. S(t) = √2Eb/Tb Cos [ 2πf 1t + θ(0)] for symbol 1 √2Eb/Tb Cos [ 2πf 2 t + θ(0)] for symbol 0

105. What is nominal carrier frequency in MSK? 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

106. What is carrier synchronization ?

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.

107. What are the three broad types of synchronization?

1. Carrier synchronization 2. Symbol & Bit synchronization 3. Frame synchronization.

108. What are the two methods for carrier synchronization? Carrier synchronization using Mth Power loop Costas loop for carrier synchronization

109. What is called symbol or bit 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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111. What are the disadvantages of closed loop bit synchronization. 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)

114. What is the error probability of MSK and DPSK? Error probability of MSK: Pe=1/2erfc√(eb/n0) Error probability of DPSK: Pe= 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.

116. Compare the probability of error of PSK with that of FSk?

BPSK: Pe= 1/2 erfc√(eb/n0) BFSK: Pe=1/2 erfc √(0.6eb/4n0)

For the fixed value of eb/n0, error probability of BPSK is less than BFSK. For the given probability of error, the eb/n0 of BPSK is 3db less compared to that

BFSK.

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UNIT-4 ERROR CONTROL CODING

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 . 119. What is BCH code? 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. 120. What is RS code? 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: dmin=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. 122. Define free distance and coding gain? 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/2 123. What is linear code? A code is linear if the sum of any two code vectors produces another code vector.

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124. What is code rate? 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.

126. What is hamming distance? 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. 127. What is meant by systematic & non-systematic code? 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 = 2q –1 3) No of message bits K = n-q 4) Minimum distance dmin =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. 130. Why RS codes are called maximum distance separable codes ? ( 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 131. What are Golay codes? 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. 132. What are the advantages of cyclic codes? 1. Encoders and decoders for cyclic codes are simple 2. Cyclic codes also detect error burst that span many successive bits. 133. What is meant by syndrome of linear block code? The non-zero output of the produce Yht 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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UNIT-5 SPREAD SPECTRUM SYSTEMS

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. 136. What is processing gain? 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. 137. Define pseudo-noise (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. 138. What does the term catastrophic cyclic code represent? ‘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. 139. Define a random binary sequence. A random binary sequence is a sequence in which the presence of a binary symbol 1 or 0 is equally probable. 140. State the balance property of random binary sequence. 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. 141. Mention about the run 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. 142. Give the correlation property of random binary sequence. The autocorrelation function of a maximum length sequence is periodic and binary valued. This property is called the correlation property. 143. Mention the significance of spread spectrum modulation. 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

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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. 144. What is called processing gain? Processing Gain (PG) is defined as the ratio of the bandwidth of spread message signal to the bandwidth of unspreaded data signal i.e.

145. What is called jamming effect? 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. 146. What is Anti jamming? 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. 148. What is called frequency hop spread spectrum? In frequency hop spread spectrum, the frequency of the carrier hops randomly from one frequency to another frequency. 149. What is slow frequency hopping? If the symbol rate of MFSK is an integer multiple of hop rate (multiple symbols per hop) then it is called slow frequency hopping 150. What is fast frequency hopping? If the hop rate is an integer multiple of symbol rate (multiple hops per symbol) then it is called fast frequency hopping. 151. What are the two function of 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. 152. What are the features of code Division multiple Accesses? 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.

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153. What is called multipath Interference? The interference caused by the interfacing of the signal form the indirect path with the signal of direct path is called multipath interference. 154. What is the advantage of a spread spectrum technique? 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. 155. Define code word The encoded block of “n” bits is known as code word. It consists of message bits and redundant bits. 156. Define code rate.

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

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PART- B QUESTIONS

1. Drive the expression for the sampling process in time domain. 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.

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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.