DC 2 Marks Q&A

Dr.N.G.P.
Institute of Technology / ECE Question Bank - 2 Marks with Answers
Dr. N.G.P. INSTITUTE OF TECHNOLOGY

Coimbatore-641048
DEPARTMENT OF
ELECTRONICS AND COMMUNICATION ENGINEERING
TWO MARKS WITH ANSWERS
EC6501 / DIGITAL COMMUNICATION

REGULATION: 2013
Prepared by
Ms.C.Karthika, AP/ECE
1 EC6501 – Digital Communication

Dr.N.G.P. Institute of Technology / ECE Question Bank - 2 Marks with Answers
UNIT – I
SAMPLING & QUANTIZATION
1. State sampling theorem for band-limited signals and the filter to avoid aliasing. [Nov 2015, Apr
2018]
The sampling theorem for band-limited signals of finite energy can be stated in two parts:
a) If a finite-energy signal g(t) contains no frequencies higher than W hertz, it is completely
determined by specifying its ordinates at a sequence of points spaced 1/2W seconds apart.
b) If a finite energy signal g(t) contains no frequencies higher than W hertz, it may be completely
recovered from its ordinates at a sequence of points spaced 1/2W seconds apart.
The aliasing can be avoided in two steps:

a) A pre-alias filter (a low pass filter) is used before sampling to avoid the high frequency
components of the signal which are mostly considered as non informative signals.
b) The filtered signal is then sampled at a frequency rate slightly higher than Nyquist rate 2W,
where W is the cutoff frequency of the pre-alias filter.
2. Write the two fold effects of Quantization Process. [Nov 2015]

The two fold effect of quantization process are
a) The peak to peak range of input sample values is subdivided into a finite set of decision levels or
decision threshold called ‘risers’.
b) The range of sample rate is divided into a finite set of representation levels or reconstruction
values called ‘treads’.
3. What is aliasing? [May 2016, Nov 2016]

The phenomenon of high frequency in the spectrum of the original signal g(t) seemingly taking
on the identity of a lower frequency in the spectrum of sampled signal g(t) is called as aliasing or
foldover. The high frequencies in the frequency shifted replicas of G(f) are interfering into the low
frequencies in Gδ(f).
4. What is companding? Sketch the input-output characteristics of a compressor of an expander.

[May 2016, Nov 2016, Apr 2017]
The practical speech signal have the ratio of peaks of loud talk to weak talk as 1000:1, which is difficult
to quantize with an uniform quantizer, here the non-uniform quantizer is used.
The non-uniform quantization is achieved by compressor at the transmitter and expander at the receiver.
This combined process of compressing – uniform quantization – expanding is called as companding.

5. A certain lowpass bandlimited signal x(t) is sampled and the spectrum of the sampled version has
the first guard bandfrom 1500 Hz to 1900 Hz. What is the sampling frequency? What is the
maximum frequency of the signal?
The spectrum of the sampled version has the guard band from 1500 Hz to 1900 Hz.
Hence, maximum frequency of signal is fm = 1500 Hz
The sampling frequency is fs = (fm + guard band) *2
= (1500 + 400)*2
fs = 3800 Hz.
6. Derive the expression for quantization noise of a PCM system. [Nov 2017]
In source encoding, the difference between the actual analog value and quantized digital value is
called quantization error. This error is either due to rounding off function.
The quantization noise is considered as a additive independent source of noise which is uniformly
distributed throughout the signal. The quantization noise is considered as a random variable Q with a
sample value q
The mean of quantization noise is µ=0
The variance of quantization noise is σQ2 = ∆2 / 12
7. In a PCM system the output of the transmitting quantizer is digital. Then why is it further
encoded? [Nov 2017, Apr 2018]
In combining the process of sampling and quantizing, the specification of a continuous baseband
signal becomes limited to a discrete set of values, but not in the form best suited for transmission over a
channel. This requires the process of encoding to represent the discrete set of sample value to a more
appropriate form of signal.
Hence, each of the discrete value is represented using binary codes or ternary code, by which
each symbol will have its unique value.
8. What is the need for non-uniform quantization? [May 2014]
 The Uniform quantization is not efficient for all types of signals. Especially in speech
communication, smaller amplitudes are more dominant than the larger amplitudes.
 Thus the usage of uniform step size in inefficient for speech communication, since most of the
quantization levels are rarely used.
 It is desirable to use smaller step size at lower levels and larger step size at higher levels, This
type of quantization is called non-uniform quantization.
 The effect of non-uniform quantization is indirectly achieved by compressing the signal first and
applying the uniform quantization.
9. State any two non-uniform quantization rules. [May 2013]

There are two widely used solutions for the non-uniform quantization.
a) µ-law companding: In the µ-law companding, the compressor characteristics c(x) is continuous. µ
takes up values from 0 to 255, where the practical value is considered to be 255. The µ-law is
used for PCM telephone systems in the United States, Canada and Japan.
b) A-law companding: In the A-law companding, the compressor characteristics c(x) is piecewise,
linear segment for low-level inputs and a logarithmic segment for high-level inputs. A takes
values from 1 , where 87.56 is the practical value. A-law is used for PCM telephone systems in
Europe and India.
µ-law companding
A-law companding
10. What is natural sampling? [May 2013]

In this scheme of sampling, instead of using
impulse train, a rectangular pulse c(t) of
duration T and amplitude A is used.

11. Write the A law of compression. [Nov 2013]

A-law companding: In the A-law companding, the compressor characteristics c(x) is piecewise,
linear segment for low-level inputs and a logarithmic segment for high-level inputs. A takes values from 1
, where 87.56 is the practical value. A-law is used for PCM telephone systems in Europe and India.
12. An analog waveform with maximum frequency content of 3 KHz is to be transmitted over an
M-ary PCM system, where M=16. What is the minimum number of bits/sample that should be used
is digitizing the analog waveform? (the quantization error is specified not to exceed ±1% of the
peak-to-peak analog signal) [Dec 2012]
Here M = 16
m = log2 M = log2 16
m=4 bits / sample.
13. What is the difference between natural and flat top sampling? [Nov 2014]
Natural Sampling: In this scheme of sampling, instead of using impulse train, a rectangular pulse c(t) of
duration T and amplitude A is used.
Flat top sampling: In this scheme, the analog signal g(t) is sampled instantaneously at the rate fs and the
duration of each sample is lengthened to T of a rectangular pulse h(t)

Natural sampling Flat top sampling
14. Mention the application of pulse communication systems. (or PCM) [Nov 2014]
a) High noise immunity
b) Repeaters place will give good output power
c) Can store PCM due to digital nature
d) Various encoding techniques can be used since that it is digital
15. Define Nyquist Interval.
Consider an input analog signal x(t) with maximum frequency content W Hz. As per the Nyquist
theorem, the instantaneous sample values should be uniformly spaced in time with a period
Ts = 1 / 2W.
16. List out the advantages and disadvantages of Digital communication over analog
communication.
Advantages
 Enable privacy and security through the use of encryption
 More robust to transmission noise and interference
 Possibility of efficient regeneration
 Enables the compression and error correction facilities.
Disadvantages
 Quantization error
 High power consumption
 Infinite bandwidth requirement
 Regenerative repeaters will increase the cost of transmission
17. What is the main role of commutator in TDM.
The commutator is implemented in TDM as a switching circuitry, which has the following functions,
 To take a narrow sample of each of the N input messages at a rate fs that is slightly higher than
2W, where W is the cutoff frequency of the pre-alias filter
 To sequentially interleave these N samples inside a sampling interval Ts = 1/fs
18. What should be the minimum bandwidth required to transmit a PCM channel?
[Dec 2011]
The minimum bandwidth required for the transmission of PCM signal is
BW >= nfm
Where, n = no. of bits per sample
fm = max frequency
19. Give the expression of Signal to Noise Ratio in PCM. [Dec 2010]
The Signal to Noise Ratio is defined as the ratio between the variance of the input signal to the variance
of the quantization noise.
𝜎𝑥2
𝑆𝑁𝑅 =
𝜎𝑄2
Substituting σQ2 = ∆2 / 12
𝜎𝑥2
𝑆𝑁𝑅 =
∆2 / 12
20. What is the function of commutator in TDM?
The commutator is implemented using electronic switching circuitry. The function of commutator is
a) To take a narrow sample of each of the N input messages at a rate f s that is slightly higher than
2W, where W is the cut off frequency of the pre-alias filter
b) To sequentially interleave these N samples inside a sampling interval Ts.
21. What is TDM?

An important feature of pulse-amplitude modulation is conservation of time. That is, for a given
message signal, transmission of the associated PAM wave engages the communication channel for only a
fraction of the sampling interval on a periodic basis.
Hence, some of the time interval between adjacent pulses of the PAM wave is cleared for use by
other independent message signals on a time-shared basis which is called as Time division Multiplexing.
22. What do you mean by the term ‘crosstalk’?

The TDM is immune to amplitude nonlinearities in the channel as a source of crosstalk, because
the different message signals are not simultaneously impressed on the channel. The term ‘crosstalk’ refers
to the signal from an adjacent channel spilling over into a desired time slot.
23. Define Aperture Effect.
This results from the use of flat-top pulses to perform the sampling operation. The distortion due
to the aperture effect is compensated for by cascading an amplitude equalizer with the reconstruction
filter in the receiver.
UNIT II
WAVEFORM CODING
1. What is meant by Granular noise and how to overcome it? [Apr 2017]

 Granular noise occurs which the step size is too large comparing to the slope characteristics of the
input signal x(t).
 Hence there is fluctuating of the a approximation u(t) between + δ and - δ for a flat segment of the
input signal.
 The granular noise can be overcome by decreases the step size.
2. What is Linear Predictor? On what basis are the predictor co-efficients determined.[May 2016,
Nov 2016, Apr 2017]
 The Prediction is a special form of estimation. The requirement is to use a finite set of present and
past samples of a signals to predict the future values. The prediction is linear if it is a linear
combination of the given sample of the signals. The filter designed to perform this kind of a
prediction is called a Linear Predictor.
 The predictor co-efficient h01 is chosen such that
h01 = RX (k)
RX (k-1)
Where,
RX (1) = auto correlation function of kth tapping of the filter and the k-1 tapping
RX (0) = auto correlation function of k-1th tapping of the filter and the k-2 tapping
3. Why Delta Modulation is superior to DPCM?[Nov 2017]

 DPCM encodes the difference between adjacent samples, DPCM can have more than one bit of
encoding the sample whereas DM encodes the input sample by one bit per sample value by
increasing or decreasing step size by + δ or –δ which eliminates the need for word framing.
 The bit rate of DM is less compared to DPCM.
 The bandwidth of DM is much reduced than DPCM.
 DM can be used for coding of over sampled values more effectively than DPCM
 DM has simplicity of design of transmitter and receiver.
4. What is meant by Delta Modulation? [Apr 2018]

 The oversampled signals exhibit the increase correlation between the adjacent samples. This
increases the need for the Delta Modulation.
 DM is a one-bit quantizer or a two level quantizer.
 The input signal is approximated by the uniform staircase approximation of step-size + δ or –δ
 The difference between the input and the approximation is quantized into only two levels.
 The staircase increases by + δ if the approximation is less than the sample value of the input
signal and decreases by –δ if the approximation is greater than the sample value of the input
signal.

5. What are the advantages of Delta Modulator. [May 2016]

The Delta Modulation offers two unique features
 Delta modulation transmits only one bit for one sample. Thus the signaling rate and transmission
channel bandwidth is quite small for delta modulation.
 The transmitter and receiver implementation is very much simple for delta modulation. There is
no analog to digital converter involved in delta modulation.
6. Define AQF and AQB.

Both these are the techniques used for the Adaptive quantization in ADPCM
a) AQF – Adaptive Quantization with Forward Estimation
The unquantized samples of the input signal are used to derive forward estimates of σX(nTs)
Advantages are
 The step size prediction is obtained from unquantized samples hence it is more reliable than that
from AQB
Disadvantages are
 A processing delay (of 16ms for speech) in the encoding operation results from the used of AQF,
which is unacceptable in some critical applications.
 The level information of the step size (5 to 6 bits per step size) is to be sent to the receiver as
additional side information which increases the bandwidth.
 The buffering of the input signal is also an addition drawback.
b) AQB – Adaptive Quantization with Backward Estimation
Samples of the quantizer output are used to derive backward estimates of σX(nTs)
Advantages are
 No processing delay in the encoder
 No level information to be carried to the receiver as there is a local level estimator in the receiver
 No buffer required
Disadvantages are
 The step size prediction is obtained from quantized samples hence it is not as reliable as that from
AQF.
7. Define APF and APB. [Nov 2015]

Both these are the techniques used for the Adaptive prediction in ADPCM

a) APF – Adaptive Prediction with Forward Estimation

The unquantized samples of the input signal are used to derive estimates of the predictor coefficients.
Advantages are
 The prediction value is obtained from unquantized samples hence it is more reliable than that
from APB
Disadvantages are
 A processing delay (of 16ms for speech) in the encoding operation results from the used of AQF,
which is unacceptable in some critical applications.
 The level information of the predictor co-efficients is to be sent to the receiver as additional side
information which increases the bandwidth.
 The buffering of the input signal is also an addition drawback.
b) AQB – Adaptive Quantization with Backward Estimation
The samples of the quantizer output and the prediction error are used to derive estimates of the
predictor coefficients.
Advantages are
 No processing delay in the encoder
 No level information to be carried to the receiver as there is logic for adaptive prediction in the
receiver for estimation of predictor co-efficients.
 No buffer required
Disadvantages are
 The step size prediction is obtained from quantized samples hence it is not as reliable as that from
APF.
8. What are the limitations of Delta Modulation. [Nov 2015]

Delta Modulation are subject to two types of quantization errors
a) Slope of overload distortion.
The step size is too small for the staircase approximation u(t) to follow a steep segment of the
input waveform x(t) , with the result that u(t) falls behind x(t). This is called as slop-overload distortion.
This can be overcome by increasing the step size.
b) Granular noise.
Granular noise occurs which the step size is too large comparing to the slope characteristics of the
input signal x(t). Hence there is fluctuating of the a approximation u(t) between + δ and - δ for a flat
segment of the input signal. The granular noise can be overcome by decreases the step size.
9. What is slope overload distortion in Delta Modulation? How to overcome it?[Nov 2016, Nov
2017]

 The step size is too small for the staircase approximation u(t) to follow a steep segment of the
input waveform x(t) , with the result that u(t) falls behind x(t).
 This is called as slop-overload distortion.
 This can be overcome by increasing the step size.
10. List any four speech encoding procedures.[Apr 2014]

Source produces speech signal that vary with time. The encoder captures the temporal characteristics of
the source waveforms. The encoder digitize sometime varying parameter of the time waveform is known
as speech coding. Types are
a) PCM – Pulse Code Modulation
b) APCM – Adaptive Pulse Code Modulation
c) DPCM – Differential PCM
d) ADPCM – Adaptive Differential PCM
e) DM – Delta Modulation
f) ADM – Adaptive DM
11. List down the applications of waveform coders.

There are two related applications
 Hierarchy of digital multiplexers, whereby digitized voice and video signals as well as digital
data are combined into one final data stream
 T1 System
 M12 Multiplexer
 Light wave transmission link that is well-suited for use in a long-haul telecommunication net-
work
12. Design requirements for low bit rate speech coders.

For coding speech at low bit rates, a waveform coder of prescribed configuration is optimized by
exploiting both statistical characterization of speech waveforms and properties of hearing.
In particular, the design has two constraints
 To remove redundancies from the speech signal as far as possible
 To assign the available bits to code the non redundant parts of the speech signal in a perceptually
efficient
A bit rate from 64kb/s to 32kb/s, 16kb/s, 8kb/s, 4kb/s can be achieved by proper removal of redundancies
and bit assignment.
13. Mention the merits of DPCM compared to PCM.

PCM – involved coding of each of the sampled values of the input signal.
DPCM – is coding of the difference between the adjacent samples of the input signal.
1 Bandwidth requirement of DPCM is less compared to PCM.

2. Quantization error is reduced because of prediction filter.
3. Numbers of bits used one sample value are also reduced compared to PCM.
4. Redundancies are reduced in DPCM.
14. What is DPCM?

 When a voice or video signal is sample at a rate slightly higher than the Nyquist rate, then the
resulting sampled signal exhibit a high degree of correlation between adjacent samples.
 The adjacent samples are dependant or is similar to each other.
 When these highly correlated samples are encoded, the resulting encoded signal contains
redundant information.
 By removing the redundancy, before encoding, we obtain a more efficient coded signal. This type
of encoding is called as DPCM.
15. What is meant by adaptive delta modulation? What is its operating principle?
In adaptive delta modulation, the step size is adjusted as per the slope of the input signal. Step
size is made high if slope of the input signal is high. This avoids slope overload distortion.
Principle of ADM
a) If successive errors (difference between adjacent samples) are of opposite polarity, then the delta
modulation is operating in its granular mode. The step size is reduced.
b) If however, successive errors (difference between adjacent samples) are of the same polarity then
the delta modulator is operating in its slope overload mode. The step size is increased.
Thus by varying the step size in accordance with the changes in the input signal is called ADM.
process is modeled as linear system. If the number of speech parameter is sufficiently small, then large
compression ratio and reduction in bandwidth are achieved.
16. What are the advantages and disadvantages of ADM?

Advantages of ADM
a) One bit per sample is taken
b) Because of the variable sep size, the dynamic range is wide
c) Utilization of the bandwidth is better than DM
d) SNR is better than DM
Disadvantage of ADM
a) Quantization noise-granular noise present
b) Adaptive algorithm uses complex circuits for modulation and demodulation.
17. How bandwidth required for the DM signal is less than that of PCM?
In PCM, we have to transmit N bits per quantized sample but in DM, we have to transmit only 1 bit per
sample. This reduces the bandwidth.
BW (PCM) = nfs
Where, fs - sampling frequency
n – no. of bits per sample (2n = no. of level of quantizer, then n = no. of bits)
BW (DPCM) = nfs = fs (since n=1)
n – no. of bits per sample also remains 1 irrespective to the no. of levels in the quantizer.
18. What are the two types of adaptive quantizers used in ADPCM?
a) Adaptive Quantization with Forward Estimation (AQF)

b) Adaptive Quantization with Backward Estimation (AQB)
19. What is the principle of Linear Predictor coder or vocoder (LPC) or

 Linear prediction provides the basis of an important source coding technique for the digitization
of speech signals.
 LPC is used for encoding the speech on a short time basis say 20ms voiced speech is found to be
periodic with a fundamental frequency of a speaker.
 This voiced speech is simulated by exciting the filter Hz with a periodic impulse train having its
frequency equal to the pitch of the speech.
20. Mention the use of adaptive quantizer in adaptive digital waveform coding schemes.
 Adaptive quantizer changes its step size according to the variance of .the .input signal. Hence
quantization error is significantly reduced due to the adaptive quantization.
 ADPCM uses adaptive quantization.
 The bit rate of such schemes is reduced due to adaptive quantization.
There are two methods
a) Adaptive Quantization with Forward Estimation (AQF)
b) Adaptive Quantization with Backward Estimation (AQB)
21. Difference between DM and ADM.

DM
DM is a one-bit quantizer or a two level quantizer.
The input signal is approximated by the uniform staircase approximation of step-size + δ or –δ
There are two noises 1) slope overload distortion 2) granular noise
ADM
In adaptive delta modulation, the step size is adjusted as per the slope of the input signal. Step size is
made high if slope of the input signal is high. This avoids slope overload distortion.
The bandwidth, encoding and decoding procedures, SNR remains same for DM and ADM.
UNIT III
BASEBAND TRANSMISSION
1. State the desirable properties of line codes. (Or) What are the requirements of a line codes. [Apr
2017, Nov 2012]
a) Transmission Bandwidth : The line codes should use less transmission bandwidth, since the
bandwidth is a scarce and costly resource
b) Transmission Power Efficiency: The baseband transmission techniques or line codes are frequently
used for short range communication. For example, signal transmission from one chip to another chip of a
computer. Since it deals with low power IC’s and chips, the power required for data transmission should
be as minimum as possible.
c) Low Probability of Error or Noise Immunity: For some transmitted energy, some codes produce lesser
bit detection errors than others. While reconstruction at the receiver, it has to give low probability of
error.
d) Self Synchronization: Any digital communication system requires bit synchronization. Additionally
coherent detectors require carriers synchronization. Some line codes have inherent synchronization
features without requiring extra overhead. There reconstruction at the receiver, it has to give low
probability of error.
d) Error Detection and Correction Capability: Some codes, such as duobinary, provide the means of
detecting data errors without introducing additional error-detection bits into the data sequence. Line
coding should comfortably work with channel encoders and decoders.
e) Suitable Spectrum for Band limited Channel: The symbol bandwidth should be less than channel
bandwidth to avoid ISI.
f) DC Components: The dc components (zero frequency component) is usually not desirable because
some of the components of communication system like transformer will not pass dc components. It will
create unwanted energy loss and distortion.
g) Transparency: A line code should be so designed that the receiver does not go out of synchronization
for any long sequence of data symbols. If code is not transparent if for some sequence of symbols, the
clock is lost.
2. What is ISI and what are the causes of ISI?[May 2016, Apr 2014, Nov 2014]
The digital baseband channels are dispersive and hence the received pulses are not confined to
their respective time slots when they arrived at the receiver. Each pulse is influenced by its adjacent
pulses causing Inter Symbol Interference. ISI can lead to wrong decision by the decision making section
in making decision of what symbol was transmitted during each time slot.
Here the first term is the desired output term and the second term represents the residual effect of all other
transmitted bits o the decoding of the ith bit, this is called as ISI.
Causes:
One of the causes of intersymbol interference is multipath propagation in which a wireless signal from a
transmitter reaches the receiver via multiple paths.
Another cause of intersymbol interference is the transmission of a signal through a bandlimited channel,
i.e., one where the frequency response is zero above a certain frequency (the cutoff frequency).
3. What is an eye diagram?[Apr 2017]

The waveforms in the successive symbol intervals are hereby translated into one interval on the
oscilloscope display, for the case of a binary wave for which T=T b. The resulting display is called an eye

pattern because of its resemblance to the human eye for binary waves. The eye pattern provides a great
deal of information about the performance of the pertinent system.
Eye pattern is used to study the effect of ISI in baseband transmission.
1) Width of eye opening defines the .interval over .which the .received wave can be sampled without error
from ISI.
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.) Height of the eye opening at sampling time is called margin over noise
4. What are line codes? Name some popular line codes.[May 2016, Nov 2013]
Line codes are Electrical representation of binary codes. Any binary data stream is a random
sequence of binary digits 0 and 1. A method to electrically represent these binary digits is devised to
transmit the code words over the channel. This method of representation is called as line coding.
Some popular line codes are
a) Unipolar NRZ & RZ
b) Polar NRZ & RZ
c) Bipolar NRZ & RZ
d) Manchester
5. Define correlative level coding.[Nov 2016]

ISI is considered as an undesirable phenomenon that produces degradation in system performance
causing a nuisance effect. Neverthless, by adding intersymbol interference to the transmitted signal in a
controlled manner, it is possible to achieve a bit rate of 2B0 bit per second in a channel of bandwidth B0
hertz. Such a scheme are called correlative4 coding or partial-response signaling schemes.
The correlative coding is implemented by duo binary signaling and modified duo binary
signaling.
6. For the binary data 01101001, draw the Unipolar and RZ signal.[Nov2016]
- Refer notes to solve
7. Why do you need adaptive equalization in a switched telephone network?

In switched telephone network the distortion depends 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. Therefore adaptive equalization is used.
8. State Nyquist condition for pulse shaping.

The necessary condition for the pulse P(t) to satisfy in time domain is
In frequency domain,
∞
𝑚
𝑃(𝑓) = ∑ 𝑃 (𝑓 + ) = 𝑇𝑠
𝑇𝑠
𝑚=−∞
9. Define transparency of a line code. Give two example of line codes which are not
transparent.[Apr 2013]
Transparency: A line code should be so designed that the receiver does not go out of synchronization for
any long sequence of data symbols. If code is not transparent if for some sequence of symbols, the clock
is lost.
Unipolar NRZ is not transparent – Long string of zeros causes loss of synchronization.
Polar NRZ is not transparent.
10. What is Data Signalling rate.

The rate at which data is transmitted, measured in bits per second. It is also called as bit rate.
11. What is Modulation rate.

The rate at which signal level is changed is called modulation rate. It depends on the nature of the line
code format used. It is measure in symbols per second.
12. State any two application of eye pattern.[Nov 2012]

from ISI.
13. What is the necessity of equalization?[Nov 2014]

When the signal is passed through the channel distortion is introduced in terms of 1) amplitude 2) delay
this distortion creates problem of ISI. The detection of the signal also become difficult this distraction can
be compensated with the help of equalizer.
14. What is the use of eye pattern?[Nov 2013]

from ISI.
15. What is Manchester coding? What are its advantages?[Nov 2014]
Advantages
a) No DC component.
b) Does not suffer from signal droop (suitable for transmission over AC coupled lines)
c) Easy to synchronise
d) Is Transparent
16. How can ISI be overcome by Ideal Solution?

Nyquist criteria for distortionless transmission is
If received pulse P(t) satisfy this condition in time domain, then y(ti) = μai

17. What is a “Raised Cosine Spectrum”? Discuss how does it help to avoid ISI?
The frequency characteristic consists of a flat amplitude portion and a roll-off portion that has a
sinusoidal form. The pulse spectrum p(f) is specified in terms of a roll off factor α as follows:
The frequency response of α at 0, 0.5 and 1 are shown in graph below. We observed that α at 1 and 0.5,
the function P(f) cutoff gradually as compared with the ideal Nyquist channel and is therefore easier to
implement in practice.

UNIT – IV
DIGITAL MODULATION SCHEMES
1. Draw a block diagram of a coherent BFSK receiver.[Nov 2015, Nov 2016]
2. Distinguish between coherent and non-coherent reception. [Apr 2016, Nov 2016, Nov 2017, May
2013]
Coherent: The local carrier generated at the receiver is phase locked with the transmitted carrier is
known as coherent or synchronous detection.
Non-coherent: There is no synchronization between transmitter carrier and receiver is knows as non-
coherent detection, it is simple but error probability increases.
Coherent detection Non-coherent detection

When receiver exploits knowledge of the carrier’s The receiver does not utilize such phase reference
phase to detect the signal information
The receiver is said to be phase locked. During Employs demodulation that operate without
demodulation the receiver correlates the incoming knowledge of value of incoming signal phase.
signal.
Phase estimator is required Phase estimator is not required
3. What is QPSK? Write the expression for signal set of QPSK? [Apr 2016, Apr 2017]
The information carrier by the transmitted wave is contained in the phase. Phase of the carrier
takes on one of four values π/4, 3π/4, 5π/4, 7π/4.
By using trigonometric formulas the above expression can be written as
4. What do you understand by non-coherent detection? [Apr 2017, Apr 2018]

Non-coherent: There is no synchronization between transmitter carrier and receiver is knows as non-
coherent detection, it is simple but error probability increases.
5. What are the advantages of QPSK over BPSK? [May 2014]
The QPSK and PSK have the same BER.
But the QPSK is bandwidth efficient than BPSK. In QPSK, two bits can be transmitted in one symbol
whereas in BPSK only one bit can be transmitted in one symbol.

6. Draw the PSK and QPSK waveform of the bit stream 0 1 1 0 1 0 0 0 .[Nov 2017, Apr 2018]
7. Draw the signal space diagram for QAM signal for M=8.[May 2014]
8. What is QAM? List the application of QAM.[May 2013]

In M-QAM scheme, the I and Q components are permitted to be independent. In this schemes, the
carrier experiences amplitude as well as phase modulation. This leads to a square constellation.
The general form of M-ary QAM is given as
where ai and bi are
Applications:
 Stereo broad casting of AM signals
 Encoding colour signal in analog TV broad casting system
 Used in data modems

 Used in digital communication system
9. What is meant by memoryless modulation? [Dec 2012]

If the analog waveform {gm(t)} is mapped from the digital sequence {an} without any constraint
imposed by the previously transmitted symbols, the modulated signal is called memoryless modulation.
10. Draw the constellation diagram of QAM.[Dec 2014]
11. How can BER of a system be improved?

 By increasing the transmission signal power.
 By using proper equalization.
 By using Gray coding
 By using Optimum threshold at the receiver.
12. Why synchronization is necessary in a digital communication system?

In digital communication, it is necessary to detect the transmitted data from the noisy corrupted
Received signal. The repeaters should regenerate the signal from the noisy received signal. To do these
operations, the timing information is necessary at the receiver or repeater. Transmitter and receiver clock
timing signals will use the same frequency. For the coherent detection and phase of the carrier.
The estimation of carrier phase and frequency is called carrier synchronization. The receiver
should also know the time instants at which the transmitted symbols start and end. The estimation of these
time instants is called clock recovery or symbols synchronization.
13. What are the types of synchronization?
 Carrier Synchronization
 Symbol & bit synchronization
 Fame synchronization
14. Signal space diagram for coherent BPSK and BFSK.
BPSK BFSK
15. Signal space diagram for coherent QPSK and PSK(M=8)
QPSK PSK (M=8)

16. Differentiate QAM and QPSK.
17. Differentiate BPSK, BFSK and M-ary FSK

18. What is signal constellation diagram.

The diagram which defines the collection of M message points in N dimensional Euclidean space
is called signal constellation diagram. It helps to find the probability of error.
19. Define BPSK, BFSK, MSK, DPSK and give its signal constellation.
BPSK: Phase of the carrier is shifted between two values according to input bit sequence (1,0) bandwidth
= 2fb (1 symbol = 1 bit).
BFSK: Frequency of the carrier is switch according to the input bit sequence. Bandwidth = 4 fb
MSK: CPFSK signal with a deviation ratio of one half is referred to as (MSK) minimum shift keying.
DPSK: Differential phase shift keying uses differential encoding. Phase shift keying signal is modulated
at the transmitted side.
20. Mention the applications of digital modulation technique.
 Voice grade modems uses 8 phase DPSK technique
 Digital radio uses 16-ary QAM
 Satellite communication uses BPSK, QPSK technique
 Voice grade telephone channel uses FSK
21. Define Modulation rate.

The modulation rate 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 is bands or symbols per seconds.
22. What is data rate?
The data rate is defined as the rate at which data is transmitted. It is measured in bits per seconds (b/s).
Data rate or signaling rate or bit rate is denoted by
Rb = 1 / Tb
Tb = Bit duration
23. Give the error probability for the different digital modulation schemes.

UNIT – V
ERROR CONTROL CODES
1. State channel coding theorem [Nov 2015, Nov 2016, Apr 2017.
Definition
The channel capacity theorem for a discrete memoryless channel, represents the maximum amount of
information transferred per channel use. The channel coding theorem states that if a code generated by a
source has a code rate r which is less than the channel capacity C, then it is possible to find a code that
achieves error-free transmission over the channel.
The channel coding theorem thus specified the channel capacity C as a fundamental limit on the
rate at which the transmission of reliable (error-free) message can take place.
Draw backs:
a) The channel coding theorem is non constructive in nature
b) Theorem does not tell the way to find the good coding technique.
2. List the properties of cyclic codes. [Nov 2015]

Linearity property:
The sum of two code word polynomials is also another code word polynomial.
Cyclic property:
Any cyclic shift of a code word polynomial is also a code word polynomial.
3. List the properties of the generator polynomial.
a) The generator polynomial of an (n,k) cyclic code is unique in that it is the only code word
polynomial of minimum degree of (n-k)
b) Any multiple of the generator polynomial g(D) is a code word polynomial, as shown by
x(D) = a(D)g(D)mod(Dn-1)
where a(D) is a polynomial in D
c) The generator polynomial g(D) and the parity-check polynomial h(D) are factors of the
polynomial 1+Dn, as shown by
h(D)g(D) = 1+Dn
4. List the properties of syndrome.

a) The syndrome depends only on the error pattern, and not on the transmitted code word
b) All error patterns that differ at most by a code word have the same syndrome
c) The syndrome s is the sum of those columns of the matrix H corresponding to the error locations
d) With syndrome decoding, an (n,k) linear block code can correct up to t errors per code word,
provided that n and k satisfy the Hamming bound.
5. What is a linear code? [May 2016]

In block coding, the encoder accepts a k-bit message block and generates an n-bit codeword.
Thus, the code words are produces on a block-by-block basis.
When a block of data bits are added with parity bits to yield code vector, it given block codes.
When any two block code words in the valid codes are added (modulo 2 addition), it will lead to another
code word. These types of codes are Linear Block Codes.

6. What is a Convolutional Coding?

A Convolutional encoder operates on the incoming message sequence continuously in a serial
manner. The encoder of a binary convolutional code may be viewed as a finite-state machine that consists
of converts serial data into codeword in a serial manner. In applications where the message bits come in
serially rather than in large blocks, in which case the use of buffer may de undesirable. In such situations,
the use of Convolution coding may be preferred.
7. Define hamming code.

Consider a (n,k) linear block codes with the following parameters:
Number of parity bits (m) = n – k
Block length (n) = 2m – 1
Number of message bits (k) = 2m – 1 – m
where m ≥ 3
These codes are called as Hamming codes.
8. What is Viterbi Decoding scheme?
Viterbi decoding performs maximum likelihood decoding and it reduces the computational load by taking
advantages in code trellis. Decoding is done with algorithm
Metric:- It is the discreprepancy between the received signal and decoded signal at particular node.
Survivor path:- This is the path of decoded signal with minimum metric.
9. What is meant by constraint length of a convolution encoder? [May 2016]
The Constraint Length of a convolution code, expressed in terms of message bits, is defined as
the number of shits over which a single message bit can influence the encoder output. In an encoder with
an M-state register, the memory of the encoder equals M message bits, and K = M + 1 shifts are required
for a message bit to enter the shift register and finally come out. Hence the constraint length of the
encoder is K.
10. Generate the syndrome calculator for (n,k) cyclic code. [Nov 2016]
11. What are the different methods of describing the structure of a convolutional code? [Apr 2017]
The structural properties of a convolutional encoder are represented in graphical form by using
the following equivalent diagrams:
a) Code Tree – Each branch represents an input symbols, with the corresponding pair of output
binary symbols indicated on the branch.
b) Trellis – It is a tree like structure with remerging branches
c) State Diagram – The repeating portion of the trellis can be emerged into a state diagram
12. What is meant by syndrome of linear block code? [Nov 2017, Apr 2018]
The receiver has the task of decoding the code vector x from the received vector y. The algorithm
commonly used to perform this decoding operation starts with the computation of a 1-by-(n-k) vector

called the error-syndrome vector or simply the syndrome. The importance of a syndrome lies in the fact
that it depends only upon the error pattern.
Syndrome is formulated as s = yHT
13. Write the various techniques / algorithms used in encoding and decoding of convolution code.
[Apr 2018]
Encoding Techniques
Time-domain Approach
The time-domain of a binary convolutional encoder may be defined in terms of a set of n impulse
responses in time domain. The code word is obtained by convoluting the generator sequences and the
input message.
Transform-domain Approach
The convolution integral which describes the linear filtering operation in the time domain, is
replaced by the multiplication of Fourier transforms in the frequency domain. The code word is obtained
by multiplying the generator polynomial and the message polynomial.
Decoding Techniques
Veterbi Decoding
The equivalence between maximum likelihood decoding and minimum distance decoding for a
binary sysmmetric channel implies that we may decode a convolutional code by choosing a path in the
code tree whose coded sequence differs from the received sequence in the fewest number of places.
14. Define hamming distance and hamming weight. [May 2014]
Hamming Distance:
The hamming distance between any two code vectors is the number of locations (bits) in which
their elements differ.
Example:
Code word 1 (C1) = 1 0 1
Code word 2 (C2) = 1 1 0
These two code words differ in the 2nd and 3rd bits hence Hamming distance is 2
Hamming Weight:
The hamming weight of a code vector is defined as the number of nonzero elements in the code
vector. Equivalently, it can be stated that hamming weight of a code vector is the distance between the
code vector and an all-zero code vector.
Example:
Code word = 1 0 1 0 1 1 1
Hamming weight is 5
15. State the significance of minimum distance of a block code. [May 2013]
Hamming distance measures the number of bit errors it takes to transform one code word to
another. It tells the error detection and correction capability of a code.
Number of error detection capability s ≤ dmin -1
Number of error detection and correction capability t ≤ [dmin -1 / 2]
where dmin is the minimum distance
16. Find the Hamming distance between 101010 and 010101. If the minimum Hamming distance of
(n.k) linear block code is 3, what is its minimum hamming weight? [Dec 2012]
C1 = 1 0 1 0 1 0
C2 = 0 1 0 1 0 1
The above two codeword differ in all the six bits, hence
Hamming distance (d) = 6
Minimum Hamming weight dmin = 3

17. Define code rate of a block code. [Nov 2013]
Code rate is defined as the ratio between the number of message bits (k) to the number of code
word bits (n)
r = The number of message bits (k)
__________________________ 0<r<1
The number of codeword bits (n)
18. What is the need for error control codes? [Nov 2014]
During transmission of digital data there is attenuation present in the pulse transmission which
will influence the decision device to choose the wrong symbol at the receiver. The error control control
codes add parity bits along with the message in the transmitter which is helpful in the receiver to decode
the bits and detection and correct the errors.
The error detection will improve the accuracy of the received message.
19. List out the application of error correction codes.
a) Internet:
Each Ethernet frame carries a Cyclic Redundancy Check (CRC) – 32 checksum. Here
frames received with incorrect checksums are discarded by the receiver.
b) Satellite Communication:
The transponder in Satellite Communication usually words with solar power. The power
required for transmission has to be reduced without compromising the quality. Here they
use FEC to achieve the above demand.
c) Data Storage:
Error detection and correction codes are often used to improve the reliability of data
storage. Read Solomon Codes are used in Compact Discs and Extended Binary Hamming
codes are used in computer memories.
d) Deep Space Telecommunications:
The signal power gets extremely low in case of interplanetary distance in deep space
communication. The aboard space probes have limited power. To consume less power
and to offer high QoS, FEC codes like convolutional codes, Read-Muller Codes are used.
20. List the advantages and disadvantages of Cyclic codes

Advantages:
a) The error detection and decoding methods of cyclic codes are simpler and easy to implement
b) The encoders and decoders are simpler than non-cyclic code.
c) Cyclic codes have well defined mathematical structure. Hence it is an efficient and power code to
defect burst errors.
Disadvantages:
a) The error detection is simple but error correction is little complicated
b) The decoders used are complex circuit.
21. What are the advantages and disadvantages of convolutional codes?

Advantages:
1 .The decoding delay is small in convolutional codes since they operate o smaller blocks of data.
2. .The storage hardware required by convolutional decoder is less since the block sizes are
smaller.

Disadvantages:
1 .Convolutional codes are difficult to analyze since their analysis is complex.
2. .Convolutional codes are not developed much as compared to block codes.
22. Differentiate Block code and Convolution code.

Block code Convolutional Code
In block code the encoder accepts a k-bit In convolutional code the encoder accepts
message block and generates a n-bit word the message bits come in serially rather than
in block and generates n-bit codeword for
every message
These codes process the input message block These codes process the input message
by block serially
23. Compare between Code Tree and Trellis Diagram?
Code Tree Trellis Diagram
Code tree indicates flow of the coded signal Trellis diagram indicates transitions from
along the nodes of the tree. current to next states.
Code tree is lengthy way of representing Code trellis diagram is shorter or compact
coding process. way of representing coding process.

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Dr.N.G.P.

Institute of Technology / ECE Question Bank - 2 Marks with Answers

Dr. N.G.P. INSTITUTE OF TECHNOLOGY

TWO MARKS WITH ANSWERS

EC6501 / DIGITAL COMMUNICATION

1 EC6501 – Digital Communication

The aliasing can be avoided in two steps:

2. Write the two fold effects of Quantization Process. [Nov 2015]

3. What is aliasing? [May 2016, Nov 2016]

4. What is companding? Sketch the input-output characteristics of a compressor of an expander.

3 EC6501 – Digital Communication

9. State any two non-uniform quantization rules. [May 2013]

10. What is natural sampling? [May 2013]

5 EC6501 – Digital Communication

11. Write the A law of compression. [Nov 2013]

6 EC6501 – Digital Communication

Natural sampling Flat top sampling

21. What is TDM?

22. What do you mean by the term ‘crosstalk’?

8 EC6501 – Digital Communication

3. Why Delta Modulation is superior to DPCM?[Nov 2017]

4. What is meant by Delta Modulation? [Apr 2018]

9 EC6501 – Digital Communication

5. What are the advantages of Delta Modulator. [May 2016]

6. Define AQF and AQB.

7. Define APF and APB. [Nov 2015]

10 EC6501 – Digital Communication

a) APF – Adaptive Prediction with Forward Estimation

8. What are the limitations of Delta Modulation. [Nov 2015]

11 EC6501 – Digital Communication

10. List any four speech encoding procedures.[Apr 2014]

11. List down the applications of waveform coders.

12. Design requirements for low bit rate speech coders.

13. Mention the merits of DPCM compared to PCM.

1 Bandwidth requirement of DPCM is less compared to PCM.

14. What is DPCM?

16. What are the advantages and disadvantages of ADM?

13 EC6501 – Digital Communication

b) Adaptive Quantization with Backward Estimation (AQB)

19. What is the principle of Linear Predictor coder or vocoder (LPC) or

21. Difference between DM and ADM.

3. What is an eye diagram?[Apr 2017]

15 EC6501 – Digital Communication

5. Define correlative level coding.[Nov 2016]

7. Why do you need adaptive equalization in a switched telephone network?

2) Number of links in connection.

8. State Nyquist condition for pulse shaping.

10. What is Data Signalling rate.

11. What is Modulation rate.

12. State any two application of eye pattern.[Nov 2012]

13. What is the necessity of equalization?[Nov 2014]

17 EC6501 – Digital Communication

15. What is Manchester coding? What are its advantages?[Nov 2014]

16. How can ISI be overcome by Ideal Solution?

18 EC6501 – Digital Communication

19 EC6501 – Digital Communication

Coherent detection Non-coherent detection

By using trigonometric formulas the above expression can be written as

4. What do you understand by non-coherent detection? [Apr 2017, Apr 2018]

20 EC6501 – Digital Communication

8. What is QAM? List the application of QAM.[May 2013]

where ai and bi are

 Used in data modems

DC 2 Marks Q&A

DC 2 Marks Q&A