DIGITAL COMMUNICATION AND STOCHASTIC

PROCESSES
BY SOURAV CHAKRABARTI

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 PREVIOUS YEARS’ QUESTIONS

Q. What is Sampling Theorem Statement and Its

Applications.
A continuous signal or an analog signal can be
represented in the digital version in the form of samples.
Here, these samples are also called as discrete points. In
sampling theorem, the input signal is in an analog form
of signal and the second input signal is a sampling signal,
which is a pulse train signal and each pulse is equidistance
with a period of “Ts”. This sampling signal frequency
should be more than twice of the input analog signal
frequency. If this condition satisfies, analog signal
perfectly represented in discrete form else analog signal
may be losing its amplitude values for certain time
intervals. How many times the sampling frequency is
more than the input analog signal frequency, in the same
way, the sampled signal is going to be a perfect discrete
form of signal. And these types of discrete signals are well
performed in the reconstruction process for recovering
the original signal.
The sampling theorem can be defined as the conversion
of an analog signal into a discrete form by taking the
sampling frequency as twice the input analog signal
frequency. Input signal frequency denoted by Fm and
sampling signal frequency denoted by Fs.

The output sample signal is represented by the samples.

These samples are maintained with a gap, these gaps are
termed as sample period or sampling interval (Ts). And
the reciprocal of the sampling period is known as
“sampling frequency” or “sampling rate”. The number of
samples is represented in the sampled signal is indicated
by the sampling rate.
Sampling frequency Fs=1/Ts
Sampling theorem states that “continues form of a time-
variant signal can be represented in the discrete form of
a signal with help of samples and the sampled (discrete)
signal can be recovered to original form when the
sampling signal frequency Fs having the greater
frequency value than or equal to the input signal
frequency Fm. Fs ≥ 2Fm

If the sampling frequency (Fs) equals twice the input

signal frequency (Fm), then such a condition is called the
Nyquist Criteria for sampling. When sampling frequency
equals twice the input signal frequency is known as
“Nyquist rate”. Fs=2Fm

If the sampling frequency (Fs) is less than twice the input

signal frequency, such criteria called an Aliasing effect.
Fs<2Fm

So, there are three conditions that are possible from the
sampling frequency criteria. They are sampling, Nyquist
and aliasing states. Now we will see the Nyquist sampling
theorem.
Nyquist Sampling Theorem

In the sampling process, while converting the analog

signal to a discrete version, the chosen sampling signal is
the most important factor. And what are the reasons to
get distortions in the sampling output while conversion
of analog to discrete? These types of questions can be
answered by the “Nyquist sampling theorem”.

Nyquist sampling theorem states that the sampling signal

frequency should be double the input signal’s highest
frequency component to get distortion less output signal.
As per the scientist’s name, Harry Nyquist this is named
as Nyquist sampling theorem. Fs=2Fm
Sampling Output Waveforms

The sampling process requires two input signals. The first

input signal is an analog signal and another input is
sampling pulse or equidistance pulse train signal. And the
output which is then sampled signal comes from the
multiplier block. The sampling process output waveforms
are shown below.

Shannon Sampling Theorem

The sampling theorem is one of the efficient techniques
in the communication concepts for converting the analog
signal into discrete and digital form. Later the advances in
digital computers Claude Shannon, an American
mathematician implemented this sampling concept
in digital communications for converting the analog to
digital form. The sampling theorem is a very important
concept in communications and this technique should
follow the Nyquist criteria for avoiding the aliasing effect.
Applications
There are few applications of sampling theorem are
listed below. They are
To maintain sound quality in music recordings.
Sampling process applicable in the conversion of analog
to discrete form.
Speech recognition systems and pattern recognition
systems.
Modulation and demodulation systems
In sensor data evaluation systems
Radar and radio navigation system sampling is
applicable.
Digital watermarking and biometric identification
systems, surveillance systems.

Sampling Theorem for Low Pass Signals

The low pass signals having the low range frequency and
whenever this type of low-frequency signals need to
convert to discrete then the sampling frequency.
should be double than these low-frequency signals to
avoid the distortion in the output discrete signal. By
following this condition, the sampling signal does not
overlap and this sampled signal can be reconstructed to
its original form.

Bandlimited signal xa(t)

Fourier signal representation of xa(t) for reconstruction
Xa(F)

Proof of Sampling Theorem

The sampling theorem states that the representation of
an analog signal in a discrete version can be possible with
the help of samples. The input signals which are
participating in this process are analog signal and sample
pulse train sequence.

Input analog signal is s(t) 1

The sample pulse train is

The spectrum of an input analog signal is,

Fourier series representation of the sample pulse train is

The spectrum of the sample output signal is,

When these pulse train sequences are multiples with the
analog signal we will get the sampled output signal
which is indicated by here as g(t).

When the signal related to equation 3 passes from the

LPF, only Fm to –Fm signal only passed to the output
side and the remaining signal is going to be eliminated.
Because LPF is assigned to the cut off frequency which is
equal to the input analog signal frequency value. In this
way at one side analog signal going to converted to
discrete and recovered to its original position simply
passing from a low pass filter.

Q. What is BPSK?
Binary Phase Shift Keying (BPSK) is a modulation
technique employed in communication systems to
transmit information via a communication channel.
In BPSK the carrier signal is modified by altering its phase
by 180 degrees, for each symbol. A phase shift of 180
degrees denotes a binary 0 while no phase shift
represents a binary 1. The BPSKs modulation process is
straightforward and efficient making it suitable for
scenarios where the communication channel suffers from
noise and interference.

Importance of BPSK
BPSK holds significance in communication systems
like Wi-Fi, Bluetooth and satellite communication. Its
simplicity and robustness make it an excellent choice for
applications where the quality of the communication
channel isn't optimal. Using a basic phase shift to convey
symbols BPSK can reliably transmit data, over channels
guaranteeing dependable communication.
Implementation details of BPSK
The implementation details of BPSK carries basic
functions in digital modulation techniques, choosing
appropriate basic functions and modulation process in
BPSK.

Basis Functions in Digital Modulation Techniques

In modulation techniques, we select a group of functions
to represent the modulation scheme. These functions are
usually orthogonal, to each other. Can be derived
through the Gram-Schmidt orthogonalization
procedure. By choosing functions we can express any
vector in the signal space as a linear combination of
them.
Choosing the Basic Function for BPSK
In BPSK the modulation process involves using a sinusoid
as the basis function. By adjusting the phase of this
sinusoid based on the message bits we can achieve
modulation. When transmitting a 1 there is no phase
shift, in the carrier signal. However, when transmitting a
0 there is a phase shift of 180 degrees in the carrier
signal. This straightforward modulation scheme enables
the transmission of data.
Modulation Process in BPSK
In BPSK we represent symbols as phase shifts in the
carrier signal during the modulation process. A binary 1
is sent without any change in phase while a binary 0 is
sent with a phase shift of 180 degrees. This information
about phase shifts is encoded into the carrier signal to
facilitate data transmission. The constellation diagram,
for BPSK, displays two constellation points positioned
along the x in phase). There are no points projected onto
the y-axis (quadrature) as BPSK relies on one basis
function. The phase of the carrier wave carries all the
information being transmitted.
Q. Explain Advantages and Disadvantages of BPSK.
Advantages of BPSK
There are multiple advantages of Binary Phase Shift
Keying process including all signals form, such as:

Simplicity: BPSK, which stands for Binary Phase Shift

Keying is a modulation scheme. It simplifies
implementation, in hardware and software by utilizing
two phase states; 0 degrees and 180 degrees.

Effective Operation with Reliability: It has ability to

operate effectively in the presence of noise or
interference from signals ensuring reliable performance.

Less Power Consumption: BPSK

consumes power compared to alternative methods
making it advantageous for battery powered devices.

Easy Detection: Receivers find it easy to comprehend

BPSK accurately determining the frequency and phase of
the transmitted signal.

Compatible: BPSK serves as a building block for

complex modulation schemes like QPSK (Quadrature
Phase Shift Keying) and higher order Quadrature
Amplitude Modulation (QAM).
Disadvantages of BPSK
Some of the disadvantages of Binary Phase Shift Keying
process includes:

Low Data Sending Rate: However when it comes to

data transmission speed, BPSK has limitations as it can
only send one piece of data at a time.

Less Efficient: It inefficiently uses the signal space, same

like using a whole road for just one small car.

Can be Tricky: In situations such as communication,

signal bouncing can pose challenges, for BPSK as it
weakens the signal strength and leads to potential issues.

Limited Error Correction: BPSK does not provide as

much inherent error correction capability as more
complex modulation schemes, therefore, error correction
needs to be added separately, which can increase system
complexity.
Not for Huge Data: If you need to send a lot of data fast
then BPSK might not be the best choice.

Q. What is Delta Modulation ? Show Block Diagram of

Delta Modulator.
Nyquist rate or frequency is defined as the minimum rate
at which a finite bandwidth signal needs to be sampled to
retain all of the information. In order to get a
comparatively better sampling rate in the differential
pulse code modulation process, the signal’s sampling rate
is maintained higher than the Nyquist rate.
In the Differential Pulse Code Modulation (DPCM)
process, when the sampling interval is reduced, the
sample-to-sample amplitude difference becomes small,
like the difference is of 1-bit quantization. Hence the step-
size will be very small.
Delta modulation is a process mainly used in the
transmission of voice information. It is a technique where
analog-to-digital and digital-to-analog signal conversion
are seen. Delta modulation (DM) is an easy way of DPCM.
In this technique, the difference between consecutive
signal samples is encoded into n-bit data streams. In DM,
the data which is to be transmitted is minimized to a 1-
bit data stream.

Block Diagram of Delta Modulator

The sampling rate is comparatively very high in the delta
modulation technique. The value of the step size after
quantization is smaller. In the delta modulation process,
the quantization design is easy and simple, and it gives
the user the option to design the bit rate.
The delta modulator includes a 1-bit quantizer as shown
in the figure above and a delay circuit along with two
summer circuits. The output of the delta modulator will
be a stair-case approximated waveform. The step size of
this waveform is the delta (Δ). The output quality of the
waveform is moderate. In order to obtain a high ratio
signal-to-noise, DM must adapt oversampling
techniques. In oversampling techniques, the analog signal
is sampled many times higher than the Nyquist rate.
The bandwidth in bits/second is needed for the
transmission of a delta-modulated signal. This signal is
equal to the sampling frequency. We can find the
bandwidth to transmit the modulated signal using the
below formula.
Bandwidth required to transmit the modulated signal =
ffs samples/second X 1 bit/sample
= ffs bits/second
Where,
fs is the signal’s sampling frequency
Delta Demodulator
Refer to the below figure for the delta demodulator. The
delta demodulator includes a delay circuit, a low pass
filter, and a summer connected as per the image below.
Since the predictor circuit is removed, there is no
assumed input given to the demodulator.
A low pass filter is included in the circuit for noise
elimination and to obtain better out-of-band signals.
Granular noise is eliminated at the transmitter, and
granular noise is referred to as the step-size error. When
zero noise is seen, then the output of the modulator is
equal to the demodulator input.
Q. Explain Advantages AND Disadvantages of Delta
Modulation.
Advantages of Delta Modulation

Design is easy and simple.

It is a 1-bit quantizer.

Modulator & demodulator can be designed easily.

In delta modulation, the quantisation design is very

simple.

The bit rate can be designed by the user.

Disadvantages of Delta Modulation

When the value of the delta is small, slope overload

distortion is seen, which is a type of noise.
When the value of delta is large, granular noise is seen,
which is a type of noise.

Q. Explain Advantages & Disadvantages of Adaptive

Delta Modulation.

Advantages of Adaptive Delta Modulation

Some of the benefits of the adaptive modulation process

are as follows:

The slope error in delta modulation is reduced through

adaptive delta modulation.

It employs a low pass filter to reduce quantized noise

during demodulation.

This modulation solves the slope overload problem and

granular error in delta modulation. As a result, the signal-
to-noise ratio of this modulation is higher than that of
delta modulation.

This modulation enables resilient performance in the

presence of bit mistakes. In radio design, it decreases the
requirement for error detection and correction circuits.

Adaptive delta modulation has a wide dynamic range

since the changing step size covers a wide range of
values.
Disadvantages of Adaptive Delta Modulation

Quantization noise can be observed.

Q. Difference Between Delta Modulation and

Adaptive Delta Modulation
Q. In Delta Modulation, the slope overload
distortion can be minimized by which process?

Delta modulation

Delta modulation is a process mainly used in the

transmission of voice information.

It is a technique where analog-to-digital and digital-to-

analog signal conversions are seen.

In this technique, the difference between consecutive

signal samples is encoded into n-bit data streams. In DM,
the data which is to be transmitted is minimized to a 1-
bit data stream.

But there is a disadvantage in delta modulation is that

when the value of the delta is small, slope overload
distortion is seen, which is a type of noise.

Q. What is Quadrature Phase Shift Keying? Show

Circuit Diagram.

Quadrature Phase Shift Keying is a digital modulation

method. In this method, the phase of the carrier
waveform is changed according to the digital baseband
signal. The phase of the carrier remains the same when
the input logic is the 1 but goes a phase shift when the
logic is 0. In Quadrature Phase Shift Keying, two
information bits are modulated at once, unlike Binary
Phase Shift Keying where only one bit is passed per
symbol.

Here, there are four carrier phase offsets with a phase

difference of ±90° for four possible combinations of two
bits( 00, 01, 10, 11). Symbol duration in this modulation is
twice the bit duration.

Circuit Diagram

Instead of converting bits into a digital stream, QPSK

converts it into bit pairs. This method is also known as
the Double Side Band Suppressed Carrier
modulation method. QPSK modulation circuit consists of
a bit-splitter, 2-bit serial to parallel converter, two
multipliers, a local oscillator, and a summer.
At the transmitter input, the message signal bits are
separated as even bits and odd bits using a bit splitter.
These bits are then multiplied with the same carrier
waveform to generate Even QPSK and Odd QPSK signals.
The Even QPSK signal is phase shifter by 90°, using a
phase shifter, before modulation. Here, the Local
Oscillator is used for generating the carrier waveform.
After separation of bits, a 2-bit serial to parallel converter
is used. After multiplying with the carrier waveform, both
Even QPSK and Odd QPSK are given to the summer when
modulation output is obtained.

At the receiver end for demodulation, two product

detectors are used. This product detectors convert the
modulated QPSK signal into Even QPSK and Odd QPSK
signals. Then the signals are passed through
two bandpass filters and two integrators. After
processing the signals are applied to the 2-bit parallel-
to-series converter, whose output is the reconstructed
signal.

Waveform of Quadrature Phase Shift Keying

After processing of the Even and Odd QPSK signals, they

are applied to the summer where the modulated output
is obtained.

Advantages and Disadvantages

It provides good noise immunity.
Compared to BPSK, bandwidth used by QPSK is reduced
to half.
The information transmission rate of Quadrature Phase
Shift Keying is higher as it transmits two bits per carrier
symbol.
Carrier power remains constant as the variation in the
QPSK amplitude is small.
Effective utilization of available transmission bandwidth.
Low error probability compared to other methods.

The disadvantage of QPSK compared to BPSK is the circuit

complexity.
Q. Find the decision threshold if conditional
probability density functions after addition of noise
are of Gaussian distribution and voltage V₁ represents
symbol s₁ and voltage V2 represents symbol s₂ for no
noise case. Determine the threshold when a priori
probabilities are equal -
a) for bipolar signal with V1 = +V, V2 = - V
b) for unipolar signal with V1 = +V and V2 = 0
Q. State Sampling Theorem. A TV signal has a
bandwidth of 4.5 MHz. The signal is sampled and
converted into PCM signal. Determine the sampling
rate if the signal is to be sampled at a rate of 20%
above Nyquist rate.

Sampling Theorem:

The Sampling Theorem, also known as the Nyquist-

Shannon Sampling Theorem, states that a continuous-
time signal can be completely represented by its samples
and fully reconstructed if it is sampled at a rate fs that is
at least twice the maximum frequency component fm of
the signal:

where fm is the highest frequency in the signal.

Given Problem:

Bandwidth of TV signal ( B ): 4.5 MHz

Sampling rate: 20% above Nyquist rate

Q. Explain the operation of regenerative repeater. If
the bit error probability of such a repeater is 10^-5,
then what will be the overall bit error probability of
45 identical repeaters placed 30 km apart from each
other?
Operation of a Regenerative Repeater:

A regenerative repeater is used in digital

communication systems to enhance signal quality over
long distances. Its primary functions include:

Amplification: It amplifies the incoming signal to restore

its power level lost due to attenuation over the
transmission line.

Equalization: It compensates for distortion introduced by

the transmission medium.
Timing Recovery: It regenerates the clock signal to
ensure proper timing for further stages.

Decision Making: It samples the incoming signal, makes

a binary decision based on the sampling threshold, and
regenerates a clean digital signal.

The regenerative repeater effectively reduces the

accumulation of noise and distortion by regenerating a
clean, noise-free version of the signal at each stage.
Q. An analog signal is express as x(t) = 30cos(10³ πt)
+10sin (500πt) – cos(10^2 πt). Calculate all frequency
components and find out Nyquist sampling rate.
Q. Consider the analog signal x(t) = 10 cos (500 πt).
Determine the minimum sampling 'rate' required to
avoid aliasing.
Q. Derive an expression for mean-square value of
quantization error for PCM system.
Derivation of Mean-Square Value of Quantization
Error for a PCM System

In a Pulse Code Modulation (PCM) system, the

quantization error arises from the process of mapping a
continuous range of input values to a finite set of discrete
levels. The quantization error, denoted as eq, is defined
as the difference between the input signal and the
quantized signal.
Q. Determine the output SNR of a LDM system for 2
khz sinusoidal input signal sampled at 64 khz. Slope
overload distortion is not present and reconstruction
filter has a bandwidth of 4 khz.
Given Data:

Input signal frequency fin=2 kHz

Sampling rate fs=64 kHz

Reconstruction filter bandwidth B=4 kHz

No slope overload distortion

Q. How limitations of LDM are overcome in ADM?
Limitations of LDM:

Slope Overload Distortion:

Problem: Slope overload occurs when the input signal

changes too rapidly for the system to track it accurately.
In LDM, the step size is constant, and if the signal slope
exceeds the step size, the quantizer cannot follow the
signal fast enough, leading to significant distortion.

Effect: This results in poor signal tracking for high-

frequency or rapidly changing signals, causing a
degradation in the reconstructed signal.

Granular Noise:

Problem: Granular noise occurs when the input signal has

very small variations, and the fixed step size in LDM
becomes too large relative to the signal’s small changes.
This causes the quantizer to "overestimate" the signal,
leading to errors.

Effect: This results in a low-level noise in the

reconstructed signal, even for slowly varying signals.

How ADM Overcomes These Limitations:

Adaptive Delta Modulation (ADM) adapts the step size

based on the characteristics of the input signal,
addressing both slope overload distortion and granular
noise.

Dynamic Step Size Adjustment:

Solution to Slope Overload: In ADM, the step size is not

fixed but instead is dynamically adjusted based on the
rate of change of the input signal. If the signal is changing
rapidly (i.e., high slope), the step size is increased,
allowing the system to track fast changes more accurately
and reduce slope overload distortion.

If the signal is changing slowly (i.e., low slope), the step

size is reduced, reducing granular noise by avoiding
overestimation of small variations.

Improved Signal Tracking:

Solution to Granular Noise: By adjusting the step size

according to the signal's variations, ADM reduces the
mismatch between the quantization step and the signal,
minimizing granular noise and enhancing the accuracy of
the signal reconstruction.

Efficiency in Coding:

ADM allows for more efficient use of the available

bandwidth, as the system can use larger steps when the
signal changes rapidly and smaller steps when the signal
is nearly constant. This leads to better utilization of the bit
rate and more precise signal encoding, which is
particularly useful in channels with limited bandwidth.

Q. The information in an analog signal waveform is to

be transmitted over a PCM system with an accuracy of
±0.1% (full scale). The analog voltage waveform has a
BW of 100 Hz and an amplitude range of -10 to +10
volts. Find the number of bits in each PCM word. Also
find the minimum bit rate in the PCM signal and
minimum transmission bandwidth required.
Given Information:

Accuracy: ±0.1% (full scale)

Bandwidth (BW) of the analog signal: 100 Hz

Amplitude range of the analog signal: From -10V to

+10V
Full-scale amplitude range: 20V (from -10V to +10V)

We need to calculate:

The number of bits in each PCM word.

The minimum bit rate in the PCM signal.

The minimum transmission bandwidth required.

Q. A PCM system uses a uniform quantizer followed
by a 7-bit binary encoder. The bit rate of the system is
equal to 50 kbps.
i) What is the maximum message signal bandwidth
for which the system operates satisfactorily?
ii) Calculate the output signal to quantization noise
ratio when a full-load sinusoidal modulating wave of
1-MHz frequency is applied to the input.

Given Information:

PCM system with a uniform quantizer followed by a 7-

bit binary encoder.

Bit rate: Rb=50 kbps

Number of bits per PCM word: b=7

Modulating wave frequency: 1 MHz for part ii.

We need to find:

The maximum message signal bandwidth for which the

system operates satisfactorily.

The output signal-to-quantization noise ratio (SNR)

when a 1 MHz sinusoidal signal is applied.
Q. A band-limited signal m(t) of 3 kHz bandwidth is
sampled at rate of 33(1/3)% higher than the Nyquist
rate. The maximum allowable error in the sample
amplitude (i.e., the maximum quantization error) is
0.5% of the peak amplitude mp. Assume binary
encoding. Find the minimum bandwidth of the
channel to transmit the encoded binary signal.

Given Information:

Bandwidth of the signal m(t): Bm=3 kHz

Sampling rate is 33(1/3)% higher than the Nyquist rate.

Maximum quantization error: 0.5% of the peak

amplitude mp.

Binary encoding is used for the PCM system.

We need to find:

The minimum bandwidth of the channel required to

transmit the encoded binary signal.
Q. A Compact Disk (CD) recording system samples
each of two stereo with a 16-bit analogue-to-digital
Converter (ADC) at 44.1 kb/s.

i) Determine the output signal-to-quantizing-noise

ratio for a full-scale sinusoid.

ii) The bit stream of digitized data is augmented by

the addition of error-correcting bits, clock extraction
bits and display and control bit fields. These
additional bits represent 100 percent overhead.
Determine the output bit rate of the CD recording
system.

iii) The CD can record and hour's worth of music.

Determine the number of bits recorded on a CD.
Given Information:

Sampling rate: 44.1 kHz (44,100 samples per second)

ADC resolution: 16 bits per sample

Stereo system: 2 channels (left and right)

Overhead: 100% (includes error-correcting bits, clock

extraction bits, display, and control bit fields)

Duration of music: 1 hour

We need to determine:

The output signal-to-quantization-noise ratio (SNR)

for a full-scale sinusoid.

The output bit rate of the CD recording system.

The number of bits recorded on a CD.
Q. A spread spectrum system has the following
parameters: Message bit rate = 3kbps, pn sequence
chip rate = 3.027 x 10^6, then find the processing
gain.