1

Unit 3
Q.Define modulation index & Deviation ratio of FM & sketch FM waveform For sinusoidal input.
Modulation Index
Modulation in FM is generally expressed in terms of the modulation index. The modulation index is the ratio of
the frequency deviation to the modulating frequency.

Thus the FM signal in general can be expressed as

Deviation Ratio (DR)

The deviation ratio is the ratio of maximum frequency deviation to maximum modulating signal frequency. i.e.

Thus the deviation ratio is basically the modulation index corresponding to maximum modulating frequency:
Q.Compare frequency modulation with phase modulation.
Sr. No FM PM
1 The equation for FM wave is: v = A sin [ The equation for PM wave is v = A sin [wct+
wct+ mf sin wmt] Mp sin wmt]
2 The frequency deviation is linearly The phase shift of the carrier is linearly
proportional to instantaneous amplitude proportional to instantaneous amplitude of
of the modulating signal. the modulating signal.
3 Frequency modulation is direct method Phase modulation is indirect method of
of producing FM signal. producing FM.
4 The modulation index of an FM signal is The modulation index is proportional to the
the ratio of the frequency deviation to the maximum amplitude of the modulating
modulating frequency. signal.
5 To have better quality of transmission The amount of frequency shift produced by
and reception of higher audio a phase modulator increases with the
frequencies, pre-emphasis and de- modulating frequency. Hence an audio
emphasis circuits are used equalizer is required to compensate this.
6 Amplitude of the FM wave is constant. Amplitude of the FM wave is constant.
7 Noise is better suppressed in FM systems Noise immunity is inferior to that of FM.
as compared to PM system.
8 FM is mainly used for FM broadcasting PM is used in mobile communication
used for entertainment purposes system.
 2
Q.Explain FM generation by Armstrong method with neat block diagram.
• FM generation using Armstrong method is achieved by performing following steps:
1. Generation of PM wave
2. Generation of NBFM from PM wave
3. Generation of WBFM from NBFM
• Fig. Q.23.1 shows block diagram of FM generation using Armstrong method. (See Fig. Q.23.1 on next
page)
• As shown in the Fig. Q.23.1 crystal oscillator is used to generate a stable unmodulated carrier which is
applied to the 90° phase shifter and the summing circuit.
• The 90° phase shifted carrier is applied to the balanced modulator alongwith the modulating signal.
• Thus 90° phase shifted carrier is DSBSC modulated in the balanced modulator giving us only two sidebands
with their resultant in phase with the 90° shifted carrier.
• The two sidebands and the unshifted carrier are applied to a summing circuit to get the resultant of vector
addition of the carrier and two sidebands.
Generating NBFM from PM
• In PM along with the phase variation, some frequency variation alse takes place. Higher modulating
voltages produce greater phase shift which results greater frequency deviation. And higher modulating
frequencies produce a faster rate of change of modulating voltage hence they also result greater frequency
deviation.
• Thus, in PM the carrier frequency deviation is proportional to,
• Modulating voltage and Modulating frequency.
• However, in FM the frequency deviation is only proportional to the modulating voltage regardless of its
frequency.
• Thus, to suppress high frequency modulating signals and to make frequency deviation independent of
modulating frequency the modulating signal is passed through a low pass RC filter (integrator) as shown in
Fig. Q.23.2.
• As a result, the high frequency modulating signals are attenuated but there is no change in the amplitudes of
low frequency modulating signals.
• The output of integrator is then applied to the phase modulator. •With this input the frequency deviation at
the output of the phase modulator will be effectively proportional only to the modulating voltage and
hencewe obtain FM wave at the output of phase modulator. •The NBFM is converted to WBFM by using
frequency multipliers.
 3
Q.Explain pre-emphasis in FM with circuit diagram & frequency response.
Pre-emphasis:
• It has been observed that in FM, the noise has a greater effect on the higher modulating frequencies.
• This effect can be reduced by increasing the value of modulation index (m.) for higher modulating
frequencies (fm).
• This can be done by increasing the deviation “S” and & can be increased by increasing the amplitude of
modulating signal for the modulating signals of higher frequencies.
• Thus if we “boost” the amplitude of higher frequency modulating signals artificially then it will be
possible to improve the noise immunity at higher modulating frequencies.
• The artificial boosting of higher modulating frequencies is called as pre-emphasis.
Circuit diagram:
• Boosting of higher frequency modulating signal is achieved by using the pre-emphasis circuit of Fig.
4.17(a).

• The modulating AF signal is passed through a high pass RC filter, before applying it to the FM
modulator
• As fm increases, reactance of C decreases and modulating voltage applied to FM modulator goes on
increasing.
• The frequency response, characteristics of the RC high pass network is shown in Fig. 4.17(b).
• The boosting is done according to this prearranged curve.
• The amount of pre-emphasis in US FM transmission and sound transmission in TV has beer
standardized at 75 µsec.
• The pre-emphasis circuit is basically a high pass Filter
• The 15 sec indicates the time constant of the RC circuit used for the pre-emphasis.
• The pre-emphasis is carried out at the transmitter. The corner frequency for the RC high-pass network is
2122 Hz as shown in Fig. 4.17(b).
• The pre-emphasis circuit is used at the transmitter as shown in the block diagram of Fig. 4.18.
 4
Q.Classify FM generazation methods
FM signals can be generated using either direct or indirect frequency modulation:
Direct FM modulation can be achieved by directly feeding the message into the input of a voltage-
controlled oscillator.
For indirect FM modulation, the message signal is integrated to generate a phase-modulated signal.
Q.Give comparison between pre-emphasis & De-emphasis in FM.
 5
Q.Sketch PM waveform for sinusoidal input signal. Enlist advantages & Disadvantages of phase
modulation

Advantages:

• Improved Noise Performance: PM exhibits better noise performance compared to amplitude modulation
(AM) or frequency modulation (FM). This is because PM signals maintain a constant amplitude, making
them less susceptible to amplitude-based noise.
• Efficient Bandwidth Usage: PM can be more bandwidth-efficient than some other modulation schemes
like AM. It can transmit information without significantly widening the bandwidth, making it suitable
for certain communication systems where bandwidth conservation is crucial.
• Simple Demodulation Process: Demodulating a PM signal can be simpler compared to some other
modulation techniques, especially when using coherent demodulation techniques.

Disadvantages:

• Sensitivity to Phase Distortions: PM signals are highly sensitive to phase distortions caused by
transmission impairments or environmental factors. Any phase shift in the signal can result in significant
errors in demodulation, leading to signal degradation.
• Complexity in Implementation: In some cases, implementing phase modulation systems with high
accuracy and precision can be more complex compared to other modulation techniques, especially in
systems requiring high modulation indices or stringent phase accuracy.
• Limited Use in Commercial Applications: While PM has advantages, its use in commercial applications
like broadcast systems is limited compared to other modulation schemes like AM or FM. It’s commonly
used in specialized applications such as digital communication systems and certain types of radar.
 6
Q.Differentiate between NBFM and WBFM.
 7
Q.With the help of Block diagram explain superheterodyne FM receiver
The block diagram of typical F.M. receiver is shown in the Fig. Q.27.1

• R.F. Amplifier Stage: Since F.M. signal has a larger bandwidth it is likely to encounter more noise.
Hence to reduce the noise figure of the receiver, an RF amplifier stage is used. The RF amplifier stage
matches the antenna to the receiver.
• Mixer Stage With the help of local oscillator, this stage down Converts the incoming carrier frequency to
I.F., which is 10.7 MHz for F.M. receiver.
• I.F. Amplifier Stage: In the I.F. amplifier stages, the most of the gain of receiver is developed.
• Limiter Stage: To remove the amplitude variations of the signal is the main function of the limiter. At the
output of the limiter stage, we get a constant amplitude signal, even though the amplitude of input signal
may be varying.
• FM Demodulators: FM demodulators, change the frequency deviation of the incoming carrier into an AF
amplitude variation (identical to the one that originally caused the frequency variation
 8
Q.State the types of FM detector & with neat diagram explain Balanced Slope detector.
• Frequency Discriminator (or Frequency Ratio Detector): This detector exploits the relationship between
frequency and amplitude in an FM signal. It converts frequency variations into amplitude variations,
allowing the modulating signal to be extracted through subsequent amplitude demodulation.
• Phase-Locked Loop (PLL) Detector: Utilizes a phase-locked loop circuit that tracks the phase variations in
the incoming FM signal. By locking onto the frequency changes of the carrier wave, the PLL can extract the
modulating signal.
• Ratio Detector: A type of FM demodulator that works by converting frequency variations into amplitude
variations. It involves diode networks and capacitors to achieve the frequency-to-amplitude conversion.
• Quadrature Detector (or Foster-Seeley Discriminator): Utilizes a phase-shift network and a double-tuned
transformer to convert frequency variations into amplitude variations, which are then demodulated to extract
the modulating signal.
• Phase Discriminator (or Phase Detector): Focuses on comparing the phase of the incoming signal with a
reference signal. The phase difference represents the modulating signal, which can be extracted after phase
comparison.
Circuit diagram:
• The circuit diagram of the balanced slope detector is as shown in Fig. 4.24.
• The circuit diagram shows that, the balanced slope detector consists of two slope detector circuits.
• The input transformer has a center tapped secondary.
• Hence the input voltages to the two slope detectors are 180° out of phase.
• There are three tuned circuits. Out of them the primary is tuned to IF i.e. f.
• The upper tuned circuit of the secondary (T_{1}) is tuned above f_{c} by \Delta f i.e. its resonant frequency
is (f_{c}+\Delta f).
• The lower tuned circuit of the secondary is tuned below f_{c} by Af i.e. at (f_{c}-\Delta f)
• R_{1} C_{1} and R_{2} C₂ are the filters used to bypass the RF ripple. \dot{V}_{o1} and V_{02} are the
output voltages of the two slope detectors.
• The final output voltage \dot{V}_{0} is obtained by taking the subtraction of the individual output voltages,

V_{o1} and V_{02}, V_{0}=V_{01}-V_{02}

Operation of the circuit:
 9
• We can understand the circuit operation by dividing the input frequency into three ranges as follows:
Operation for f_{in}=f_{c}
• When the input frequency is instantaneously equal to f_{\alpha} the induced voltage in the T_{3}
winding of secondary is exactly equal to that induced in the winding T_{2}.
• Thus the input voltages to both the diodes D_{1} and D_{2} will be the same.
• Therefore their dc output voltages V_{o1} and V_{O2} will also be identical but they have opposite
polarities.
• Hence the net output voltage V_{0}=0
Operation for f_{c}<f_{in}<(f_{c}+\Delta f)
• In this range of input frequency, the induced voltage in the winding T_{1} is higher than that induced in
T_{2}
• Therefore the input voltage to D_{1} is higher than D_{2}.
• Hence the positive output V_{o1} of D_{1} is higher than the negative output V_{O2} of D_{2}.
• Therefore the output voltage V_{0} is positive. As the input frequency increases towards (f_{c}+\Delta
f) the positive output voltage increases as shown in Fig. 4.25.
Operation in the range (f_{c}-\Delta f)<f_{in}<f_{c}
• If the input frequency is in this range then the induced voltage in winding T_{2} is higher than that in
T_{1}.
• Therefore input voltage to diode D_{2} is higher than that to D_{p}.
• Hence the negative cutput V_{n2} is greater than V_{a1}
• Therefore the output voltage of the balanced stope detector is negative in this frequency range.
• The negative output voltage increases as f_{in} goes closer to (f_{c}-\Delta f) as shown in Fig. 4.25.
• If the output frequency goes outside the range of (f_{c}-\Delta f) to (f_{c}+\Delta fi the output voltage
will fall due to the reduction in tured circuit response.
Characteristics of the ba anced slope detector:
The characteristics \circ^{2} the balanced slope detector is as shown in Fig. 4.25.
 10
Unit 4
Q.State sampling theorem in time domain. Explain sampling process with Block diagram.
Introduction:
• In order to represent the original message signal “faithfully” (without loss of information), it is
necessary to take as many samples of the original signal as possible.
• Higher the number of samples, closer is the representation.
• The number of samples depends on the “sampling rate” and the maximum frequency of the signal to be
sampled.
• Sampling theorem was introduced to the communication theory in 1949 by Shannon.
• Therefore this theorem is also called as “Shannon’s sampling theorem”.
• The statement of sampling theorem in time domain, for the bandlimited signals of finite energy is as
follows:
Statement:
1. If a finite energy signal x (t) contains no frequencies higher than “W” Hz (i.e. it is a band limited signal)
then it is completely determined by specifying its values at the instants of time which are spaced (1/2 W)
seconds apart.
2. If a finite energy signal x (t) contains no frequency components higher than “W” Hz then it may be
completely recovered from its samples which are spaced (1/2 W) seconds apart.
Proof of Sampling Theorem:
Let us now prove the sampling theorem in time domain. The assumptions made for this proof are as follows:
Assumptions:
• Let x (t) be a continuous time analog signal as shown in Fig. 5.3.
• Let x (t) be a signal with finite energy and infinite duration.
• Let x (t) be a strictly bandlimited signal. That means it does not contain any frequency components above
“W” Hz.
• Let s (t) be the sampling function as shown in Fig. 5.3.
• It is a train of unit impulses, spaced by a period of T, seconds.
• This sampling function samples the original signal at a rate of “f,” samples per second. Therefore “T,”
represents the sampling period such that

Continuous Signal -→ Sampling --→ Discrete Signal

Continuous Signal: This represents the original continuous signal or data. It could be an analog signal (e.g.,
sound waves, voltage levels) or a continuous dataset.
Sampling: This step involves taking samples at regular intervals from the continuous signal. A sampler or
sampling mechanism captures the amplitude or value of the signal at specific points in time. The rate at which
samples are taken is called the sampling rate or frequency (measured in Hertz – Hz). The result is a series of
discrete data points.
Discrete Signal: The output of the sampling process is a discrete signal. It represents the original continuous
signal in a series of discrete values or samples. These samples can be stored, processed, transmitted, or analyzed
digitally
 11
Q.Describe with the help of neat sketches of waveforms methods of generation of PDM / PWM. And
PPM.
Principle: The amplitude and width of the pulses are kept constant, while the position of each pulse, with
reference to the position of a reference pulse, is changed according to the instantaneous sampled value of the
modulating signal.
Generation of PWM / PDM and PPM
• The block diagram of Fig. Q.15.1 (a) shows the scheme to generate PDM and PPM. The corresponding
waveforms are shown in Fig. Q.15.1 (b). The scheme of Fig. Q.15.1(a) combines both sampling and modulation
operation.
• The sawtooth generator generates the sawtooth signal of frequency f_{s} (i.e. period T_{5}) . The sawtooth
signal, also called sampling signal is applied to the inverting input of comparator.
• The modulating signal x(t) is applied to the noninverting input of the comparator. The output of the
comparator is high only when instantaneous value of x(t) is higher than that of sawtooth waveform. Thus the
leading edge of PDM signal occurs at the fixed time period i.e. kT_{s} the trailing edge of output of comparator
depends on the amplitude of signal x(t). When sawtooth waveform voltage is greater than voltage of x (1) at that
instant, the output of comparator remains zero. The trailing edge of the output of comparator (PWM) is
modulated by the signal x(t)
• If the sawtooth waveform is reversed, then trailing edge will be fixed and leading edge will be modulated. If
sawtooth waveform is replaced by triangular waveform, then both leading and trailing edges will be modulated.
•The pulse duration modulation (PDM) or PWM signal is nothing kut output of the comparator. The amplitude
of this PDM or PWM signal
 12

Q.Explain Aliasing effect & draw the sampled output for sampling frequency Less than equal to and
greater than maximum frequency of analog signa
Definition of Aliasing:
- The phenomenon of a high frequency in the spectrum of the original signal x (t), taking on the
identity of lower frequency in the spectrum of the sampled signal x (t) is called as aliasing or fold
over error.
Effect of aliasing:
- Due to aliasing some of the information contained in the original signal x (t) is lost in the process
of sampling.
How to Eliminate Aliasing ?:
Aliasing can be completely eliminated by

1. Using an antialiasing or prealiasing filter and

2. Using the sampling frequency f, > 2W.
 13
Q.Compare pulse Amplitude modulation, pulse width modulation and pulse position modulation.
 14
Q.Define Time Division multiplexing. Explain concept of TDM with neat Diagram
There are mainly two types of multiplexers: analog and digital.
If the analog signals are multiplexed, then it is called as analog multiplexing.
If the digital signals are multiplexed, then it is called digital multiplexing.
Definition:
TDM is the multiplexing technique in which many signals are sent in a sequential manner but they occupy the
same band in the frequency spectrum.
Fig. 5.25 explains the concept of TDM.

• In TDM all the signals to be transmitted are not transmitted simultaneously. Instead, they are transmitted
one-by-one.
• Thus each signal will be transmitted for a very short time.
• One cycle or frame is said to be complete when all the signals are transmitted once on the transmission
channel.
• As shown in the Fig. 5.25 one transmission of each channel completes one cycle of operation called as a
“Frame”.
• The TDM system can be used to multiplex analog or digital signals, however it is more suitable for the
digital signal multiplexing.
 15
Q. Describe detection of PPM with block diagram.
Generation of PPM Signal:
• The PPM signal can be generated from PWM signal as shown in Fig. 5.22(a).
• The same block diagram has been repeated in
• Fig. 5.23 as shown.

Operation:
• The PWM pulses obtained at the comparator output are applied to a monostable multivibrator.
• The monostable is negative edge triggered.
• Hence corresponding to each trailing edge of PWM signal, the monostable output goes high.
• It remains high for a fixed time decided by its own RC components.
• Thus as the trailing edges of the PWM signal keep shifting in proportion with the modulating signal x(t),
the PPM pulses also keep shifting as shown in Fig. 5.23.
Detection of PPM:
The PPM demodulator block diagram is as shown in Fig

Operation:
• The operation of the demodulator circuit is explained as follows:
• The noise corrupted PPM waveform is received by the PPM demodulator circuit.
• The pulse generator develops a pulsed waveform at its output of fixed duration and apply these pulses to
the reset pin ® of a SR flip-flop.
• The PWM signal can be demodulated using the PWM demodulator.
• A fixed period reference pulse is generated from the incoming PPM waveform and the SR flip-flo
• Is set by the reference pulses.
• Due to the set and reset signals applied to the flip-flop, we get a PWM signal at its output.
 16
Q.Describe Band limited & time limited signal with suitable example.
Band-Limited Signal:
A band-limited signal is one whose frequency content is confined within a specific range or bandwidth. This
means that the signal’s energy or information exists only within a certain frequency range and is zero outside
that range.
For example, consider a signal in telecommunications that occupies frequencies between 0 Hz and 4 kHz. If all
significant frequencies of the signal exist within this range and there’s no energy at frequencies outside this
band, it’s considered band-limited. Signals in practical applications, like many audio signals and certain
communication signals, often exhibit band-limited characteristics due to limitations in transmission channels or
practical constraints.
Time-Limited Signal:
A time-limited signal is one that exists or is non-zero only within a finite duration or time interval. Outside this
interval, the signal is zero or negligible.
For instance, imagine a pulse signal that lasts from 1 second to 2 seconds. Within this time frame, the signal has
non-zero values, and beyond this interval, it becomes zero. Time-limited signals are often encountered in
various applications, such as radar pulses, specific transmission bursts in communication systems, or certain
types of digital signals that have a finite duration of relevance

Q.Draw & explain spectrum showing aliasing effect & Guard band.

In this spectrum:
• The channels are represented by different segments (Channel 1, Channel 2, Channel 3, and so on).
• Guard bands separate adjacent channels to prevent interference or aliasing between them.
• The Nyquist frequency marks the maximum frequency that can be accurately represented without
aliasing given the sampling rate.
• The unused frequencies in the spectrum represent the guard bands intentionally left without any signal to
prevent interference.
 17
Q.Explain Flat-top sampling with waveforms.
Waveforms:
- The natural sampling is rarely employed in practice.
- Instead the other practical sampling technique called flat top sampling is employed in practice.
- In the flat top sampling technique, the analog signal x (t) is sampled instantaneously at the rate 1
f, and the duration of each sample is T lengthened to a duration “t” as shown in Fig. 5.11(b).
- Thus the amplitudes of these pulses are constant and equal to the corresponding sampled values.
Circuit diagram:
- The flat top pulses can be obtained by using the sample and hold circuit shown in Fig. 5.12(a).

Operation of the sample and hold circuit:

• The sample and hold circuit consists of two FET switches and a capacitor as shown in Fig. 5.12(a).
• The analog signal x (t) is applied at the input of this circuit and the sampled signal s (t) is obtained
across the capacitor.
• A gate pulse will be applied to gate G_{1} at the instant of sampling for a very short time.
• The sampling switch will turn on and the capacitor charges through it to the sample value x (nT_{5}).
• The sampling switch is then turned off. Both the FETs will remain OFF for duration of “t” seconds and
the capacitor will hold the voltage across it constant for this period.
• Thus the pulse is stretched to “t” secords.
• At the end of the pulse interval (t), a pulse is applied to G_{2} i.e. gate terminal of discharge FET.
• This will turn on the discharge FET and short circuit the capacitor.
• The output voltage then reduces to zero. This is as shown in Fig. 5.13.
 18
Q.Distinguish between Ideal sampling, Natural sampling and Flat-Top Sampling.
 19
Unit 5
Q.Draw block diagram of PCM system & Describe working of PCM Transmitter.
Block diagram:
Block diagram of the PCM transmitter is as shown in Fig. 6.1.

Operation of PCM transmitter:

• Operation of the PCM transmitter is as follows:
• The analog signal x (t) is passed through a bandlimiting low pass filter, which has a cut-off frequency f₁
= W Hz.
• This will ensure that x (t) will not have any frequency component higher than “W”.
• This will eliminate the possibility of aliasing.
• The band limited analog signal is then applied to a sample and hold circuit where it is sampled at
adequately high sampling rate.
• Output of sample and hold block is a flat topped PAM signal.
• These samples are then subjected to the operation called “Quantization” in the “Quantizer”.
• Quantization process is the process of approximation.
• The quantization is used to reduce the effect of noise.
• The combined effect of sampling and quantization produces the quantized PAM at the quantizer output.
• The quantized PAM pulses are applied to an encoder which is basically an A to D converter.
• Each quantized level is converted into an N bit digital word by the A to D converter. The value of N can
be 8, 16, 32, 64 etc.
• The encoder output is converted into a stream of pulses by the parallel to serial converter block.
• Thus at the PCM transmitter output we get a train of digital pulses.
• A pulse generator produces a train of rectangular pulses with each pulse of duration “t” seconds.
• The frequency of this signal is “f,” Hz. This signal acts as a sampling signal for the sample and hold
block.
• The same signal acts as “clock” signal for the parallel to serial converter. The frequency “f,” is adjusted
to satisfy the Nyquist criteria.
 20

Q.State types of quantization. Explain uniform quantization with neat Waveform.

Quantization is a process in signal processing and data compression where the continuous amplitude values of a
signal are converted into a discrete set of values. There are mainly two types of quantization: uniform and non-
uniform.
Types of Quantization:
Uniform Quantization:
In uniform quantization, the range of the input signal is divided into uniformly spaced intervals, and each
interval is represented by a single quantization level.
The step size or quantization interval remains constant across the entire range of the signal.
It’s a straightforward method and easier to implement but might lead to quantization errors, especially in areas
of the signal with low amplitudes.
Non-Uniform Quantization:
Non-uniform quantization employs variable step sizes based on the characteristics of the input signal.
It adapts the quantization levels to match the signal’s amplitude distribution, aiming to minimize quantization
errors by allocating more levels to regions with higher signal power and fewer levels to regions with lower
power.
Examples include mu-law and A-law quantization used in audio compression.
Uniform Quantization with a Waveform:
 21
Q.Discuss with neat schematic, transmitter and receiver for DPCM (Differential pulse code modulation).
Block diagram:
The block diagram of a DPCM transmitter and receiver is as shown in Fig. 6.13(a)

• In DPCM scheme, the digital equivalent of each quantized sample is not transmitted by the transmitter.
This is how DPCM is different from PCM.
• Instead the digital equivalent of the “difference” between the present sample value and the previous
sample value is transmitted.
Operation of DPCM transmitter:
• Fig. 6.13(a) shows the block diagram of a DPCM transmitter. X (t) is the analog input signal and x (t) is
its approximated signal.
• What is important to know here is whether x (t) is larger or smaller than x (t) and what is the difference
between x(t) and x (t).
• At each sampling instant the difference amplifier compares x (t) and x (t) and the sample and hold
circuit will hold the result of this subtraction.
• The difference signal at the output of sample and hold circuit is quantized by the quantizer.
• The quantizer output So (t) is then transmitted as it is or it is encoded into a stream of bits as explained
in conventional PCM system.
• The quantizer output is also used to produce the approximated signal x (t) by passing the quantizer
output through a predictor and accumulator.
Reciever
Block diagram:
• The block diagram of DPCM receiver is shown in Fig. 6.13(b).
• The received signal carries the information about the difference between x (t) and x (t) at the transmitter.
• From this signal the “predictor” will predict the approximate value of next signal.
• The accumulator will add all these differences and the filter will smooth out of quantization noise.
 22
Q.Compare Analog and Digital communication.
 23
Q.Draw Block diagram of Delta modulation system & comment on drawback Of Delta Modulation.
Block Diagram:
The block diagram of a delta modulator transmitter is as shown in the Fig. 6.9.

Operation:
The operation of the circuit is as follows:
• X (t) is the analog input signal and x’ (t) is the quantized (approximated) version of x (t). Both these
signals are applied to a comparator.
• The comparator output goes high x(t)>x^{\prime} t) and it goes low if x(t)<x^{\prime}(t)
• Thus the comparator output is either 1 or 0. The sample and hold circuit will hold this level (0 or 1) for
the entire clock cycle period.
• The output of the sample and hold circuit is transmitted as the output of the DM system.
• Thus in DM, the information which is transmitted is only whether x(t)>x^{\prime}(t) or vice versa.
• Also note that one bit per clock cycle is being sent. This will reduce the bit rate anc hence the BW.
• The transmitted signal is alsc used to decide the mode of operation of an up/down counter.
• The counter output incremerts by 1 if S_{0}(t)=1 and it decrements by 1 if S_{0}(t)=0, at the falling
edge of each clock pulse.
• The counter output is converted into analog signal by a D to A converter.
• Thus we get the approximated signal x’ (☺ at the output of the D to A converter.
Disadvantages of Delta Modulation:
1. The two distortions i.e. slope overload error and granular noise are present.
2. Practically the signaling rate with no slope overload error will be much higher than that of PCM.
The slope overload error can be reduced by using another type of delta modulation, called as Adaptive Delta
Modulation (ADM).
 24

Q.Explain working of Adaptive Delta Modulation with block diagram & State advantages of ADM over
DM.
ADM Transmitter:
• In the ADM system, the step size is not constant. Rather when the slope overload occurs the step size
becomes progressive larger and therefore x’ (t) will catch up with x(t) more rapidly.
• The ADM transmitter is as shown in Fig. 6.14.

• If you compare this block diagram with that of the linear delta modulator, then you will find that except
for the counter being replaced by the digital processor, the remaining blocks are identical.
• Let us understand the operation of the digital processor. For that carefully observe waveforms of Fig.
6.15.
Advantages of ADM:
The advantages of ADM over DM are as follows:
1. Reduction in slope overload distortion and granular noise.
2. Improvement in signal to noise ratio.
3. Wide dynamic range due to variable step size.
4. Low signaling rate.

Q.Draw block diagram of Digital communication system & explain function Of each block.
Functions of the Blocks:
Source: Generates the digital information to be communicated.
Encoder/Modulator: Converts digital data into a format suitable for transmission, often by modulating it onto a
carrier wave.
Transmitter: Prepares the modulated signal for transmission by amplification and possibly error correction
coding.
Channel: Acts as the medium for transmitting the signal from the transmitter to the receiver.
Receiver: Amplifies, processes, and prepares the received signal for decoding.
Decoder/Demodulator: Extracts the original digital data from the received signal by demodulating it.
Destination: Utilizes the recovered digital information for its intended purpose

Source>Encoder/modulator>Transmitter>Channel>Reciever>decoder>destination
 25

Q.Compare A-law & u-law compander.

1. μ – Law companding:
• In the u-law companding, the compressor characteristic is continuous.
• It is approximately linear for smaller values of input levels and logarithmic for high levels of input
signal.

• Here z (x) represents the output and x is the input to the compressor. | x | / xrrax represents the
normalized value of input with respect to the maximum value xmax.
• The (sgn~x) term represents \pm1 i.e. positive and negative values of input and output.
• The µ-law compressor characteristics for different values of u are as shown in Fig. 6.7(a).

2.A-Law companding
- In the A-law companding, the compressor characteristic is of piecewise nature, made up of a
linear segment for low level inputs and a logarithmic segment for high level inputs.
- Fig. 6.8 shows the A-law compressor characteristics for different values of A.
- Corresponding to A = 1 we observe that the characteristic is linear which corresponds to a
uniform quantization.
 26
Q.Draw & explain Delta modulation waveform with slope overload & granular Noise.
Slope overload distortion (Startup error): The rate of rise of input signal x ® is so high that the staircase signal
cannot approximate it, the step size ‘8’ becomes too small for staircase signal u(t) to follow the steep segment of
x (t). Thus there is a large error between the staircase approximated signal and the original input signal x(t).
This error is called slope overload distortion. To reduce this error, the step size should be increased when slope
of signal of x (1) is high.

Since the step size of delta modulator remains fixed, its maximum or minimum slopes occur along straight
lines. Therefore this modulator is also called Linear Delta Modulator (LDM).

Granular noise (Hunting): Granular noise occurs when the step size is too large compared to small variations in
the input signal. That is for very small variations in the input signal, the staircase signal is changed by large
amount (8) because of large step size. When the input signal-is almost flat, the staircase signal u(t) keeps on
oscillating by ±8 around the signal. The error between the input and approximated signal is called granular
noise. The solution to this problem is to make step size small. Thus large step size is required to accommodate
wide dynamic range of the Input signal (to reduce slope overload distortion) and small steps are required to
reduce granular noise. Adaptive delta modulation is the modification to overcome these errors.

Q.Explain need of digital communication

Advantages of digital communication
1. Because of the advances in digital IC technologies and high speed computers, digital communication
systems are simpler and cheaper compared to analog systems.
2. Using data encryption, only permitted receivers can be allowed to detect the transmitted data. This is
very useful in military applications.
3. Wide dynamic range is possible since the data is converted to the digital form.
4. Using multiplexing, the speech, video and other data can be merged and transmitted over common
channel.
5. Since the transmission is digital and channel encoding is used, the noise does not accumulate from
repeater to repeater in long distance communication.
6. Since the transmitted signal is digital, a large amount of noise interference can be tolerated.
7. Since channel coding is used, the errors can be detected and corrected in the receivers.
3. Digital communication is adaptive to other advanced branches of data processing such as digital signal
processing, image processing, data compression etc.
Disadvantages of digital communication
• Eventhough digital communication offer many advantages as given above, it has some drawbacks also. But
the advantages of digital communication outweigh disadvantages. They are as follows –
1. Because of analog to digital conversion, the data rate becomes high. Hence more transmission
bandwidth is required for digital communication.
2. Digital commurication needs synchronization in synchronous modulation.
 27
Q.Draw and explain PCM Receiver.
Block Diagram:
Fig. 63 shows the block diagram of a PCM receiver.

Operation of PCM receiver:

• A PCM signal contaminated with noise is available at the receiver input. The regeneration circuit at the
receiver will separate the PCM pu ses from noise and will reconstruct the original POM signal.
• The pulse generator has to operate in synchronization with that at the transmitter.
• Thus at the regeneration circuit output we get a “clean” PCM signal.
• The reconstruction of PCM signal is possible due to the digital nature of PCM signal.
• The reconstructed PCM signal is ther passed through a serial to parallel converter.
• Output of this block is then applied to a cecoder.
• The decoder is a D to A converter which performs exactly the opposite operation of the enccder.
• The decoder output is the sequence of a quantized multilevel pulses.
• The quantized PAM signal is thus obtained, at the output of the decoder.
• This quantized PAM signal is passed through a low pass filter to recover the analog signal, x (1.
• The low pass filter is called as the reconstruction filter and its cut off frequency is equal to the message
bandwidth W.
Waveforms:The waveforms at various points in the PCM receiver are as shown in Fig. 6.4.
 28
Q.Distinguish between DM and ADM.
 29

Q.Draw the following data formats for bit stream 10110100101 [6]
• Unipolar RZ
• Unipolar NRZ
• Polar RZ
• Polar NRZ
• AMI (Alternative mark Inversion)
• Split Phase Manchester
 30
Q.State different Synchronization technique & explain any one in detail With neat diagram.
Definition and need:
- Synchronization is a technique which car make the clocks at the transmitter and receiver, operate
at the same rate.
- The clocks in a digital communication system should be operated at same rate in order have
synchronization.
- The synchronization is of following types:
1.Symbol synchronization
2.Frame synchronization
3. Carrier synchronization.

- The positions of the symbol and bit synchronizers in a binary receiver are shown in Fig. 7.14.
- The bit synchronizer will extract synchronization from the received signal itself, while the frame
synchronizer uses the output message of the regenerator and clock to derive the framing
information.

Q.Define Equalizer. Explain Adaptive equalization with block diagram & State Advantages of Adaptive
equalization.
Equalization: The channel distortion such as ISI or effect of gaussian noise affects the signal being transmitted.
This distortion is corrected with the help of special filters. It is called equalization.
Necessity: Most of the channels are made up of individual links. For example, in the switched telephone
network, the distortion induced depends upon
i) Transmission characteristics of individual links and
ii) Number of links in connnection.
Hence, the fixed pair of transmit and receive filters will not serve the equalization problem completely. The
transmission characteristics of the channel keep on changing. Hence adaptive equalization is used.
Basic Principle: In adaptive equalization, the filters adapt themselves to the dispersive effects of the channel.
That is the coefficients of the filters re changed continuously according to the received data. The filter efficients
are changed in such a way that the distortion in the data is duced.
Block diagram: The adaptive equalizer shown in figure below is a ped-delay-line filter. It consists of set of delay
elements and variable
 31
Q.Explain the working principle of scrambling & unscrambling with example.

Scrambler:
Fig 7.10(a) shows the schematic of a scrambler. It uses a 4-stage shift register with tap gains
\alpha_{1}=\alpha_{2} = 0 and \alpha_{3}=\alpha_{4}=1 .

The clock line has not been shown for our convenience but it is very much present.
M_{k} is the binary message sequence at the input to the scrambler. It is added to the register output,
m_{4}^{\prime\prime} to obtain the scrambled message m
The scrambled message m is also fed back to the shift register input. Looking at Fig. 7.10(a) we can write

Here m represents a scrambled version of m (2)

The scrambled version is completely different from the original message m_{2} and hence the message m_{k}
cannot be recovered unless we have a proper unscrambler at the receiver.
The scrambled signal m_{i}^{\prime} is then transmitted over a suitable communication medium.
Unscrambler:
Fig. 7.10(b) shows the schematic of an unscrambler.
It uses a 4-stage shift register with tap gains a_{1}= C_{2}=0 and \alpha_{2}=\alpha_{4}=1
The unscrambler is used to recover the original
Message signal m_{2} from the scrambled signal m_{+}^{\prime}
Thus by adding (modulo-2 addition), the scrambled signal m_{4}^{\prime} with m_{k}^{\prime\prime} it is
possible to obtain the original signal m.
Note that due to the identical structures of shift registers and same input m_{k}^{\prime} for scrambler as well
as unscrambler, we will get the same sequence m_{k}^{\prime\prime} for the scrambler as well as unscrambler.
 32
Q.Describe eye pattern Graphical Display of Inter Symbol Interference With diagram.
Definition:
- Eye pattern is a pattern displayed on the screen of a cathode ray oscilloscope (C.R.O.).
- The shape of this pattern is very similar to the shape of human eye. Therefore it is called as eye
pattern.
- Eye pattern is used for studying the intersymbol interference (ISI) and its effects cn various
communication systems.
- The eye pattern is obtained on the C.R.O. by applying the received signal to vertical deflection
plates (Y-plates) of the C.R.C. and a sawtooth wave at the transmission symbol rate i.e (1/T) to
the horizontal deflection plates (X-olates) as shown in Fig. 7.26©.

• The received digital signal and the corresponding oscilloscope display are as shown in Figs. 7.26(a) and
(c) respectively.
• The resulting oscilloscope display shown in Fig. 7.26© is called as the “eye pattern”.
• This is due to its resemblance to the human eye.
• The region inside the eye pattern is called as the Eye opening.
• The eye pattern provides very important information about the perform arce of the system.
• The information obtainable is as follows (See Fig. 7.27).
 33
Q.Describe concept of digital multiplexer and Demultiplexer with necessary Diagram.
Digital Multiplexer (MUX):
A digital multiplexer combines multiple input signals into a single output. It’s commonly denoted as a “MUX”
and is represented using a block diagram:
Inputs (I0, I1, I2, …, In): Represent the multiple input signals that need to be combined into one stream.
Control Signals (S0, S1, …, Sm-1): These signals determine which input gets routed to the output. The number
of control signals (m) determines the total number of input lines that the MUX has.
MUX Block: The logic in the MUX interprets the control signals to select the appropriate input to be passed to
the output.
For example, in a 4-to-1 MUX (with 4 input lines), the control signals (S0, S1) might select one of the four
inputs to be transmitted to the output based on their binary combination (00, 01, 10, 11).
Digital Demultiplexer (DEMUX):
A digital demultiplexer performs the reverse function of a MUX, splitting a single input signal into multiple
output lines. It’s represented using a block diagram:
Input (I): Represents the single input signal that needs to be separated into multiple streams.
Control Signals (S0, S1, …, Sm-1): Similar to the MUX, these signals determine which output line receives the
input signal.
DEMUX Block: The logic in the DEMUX interprets the control signals to route the input signal to the
appropriate output line.
 34
Q.Define synchronization & with block diagram explain bit synchronization.
The second disadvantage of closed-loop bit. Synchronizer forces us to search for another technique of
synchronization which is completely independent of zero crossings of the message signal y (t).
The synchronizer which uses this technique is called as early-late synchronizer.
Principle of early-late synchronizer:
The early-late synchronizer operates on the principle that a properly filtered digital signal has peaks at the
optimum sampling times and that the filtered signal is reasonably symmetric on either sides as shown in Fig.
7.16(a).
In Fig. 7.16(a) t is the optimum sampling instant at which the signal amplitude is maximum. |y(t_{k})|
Is the sampled value at the optimum sampling instant.
Due to the symmetry of y (t) on either sides of we can write that,
 35
Q.Explain the properties of line codes.
Line codes: The digital data obtained by PCM, DM, ADM, DPCM etc are represented by different waveforms
or data formats. These are basically different electrical levels to represent the digital symbols. These are called
line codes.
Desirable properties of line codes:
1. The line code should have adquate timing content, so that clock information can be extracted from the
waveform
2. The line code should be immune to channel noise and interference.
3. The line code should allow error detection and correction.
4. The power spectrum of line code should be matched to that of channel to reduce signal distortion
.5. The waveform of the line code should be transparent to the digital data being transmitted.

Q.Draw AT&T hierarchy multiplexing system & explain it in detail.

AT and T hierarchy Fig. Q.13.1 shows AT and T hierarchy. Observe that four T₁ lines are multiplexed in M12
multiplexer. The bit rate of each T₁ line is 1.544 Mbps. The channel bank multiplexes 24 voice PCM signals of
64 kbps bit rate into single T₁ line. It can also be used to transmit digital data, visual telephone and TV signals.

• The M23. Multiplexer at third level multiplexes seven T₂ lines. Each of these lines have 6.312 Mbps bit rate.

At the fourth level M34 multiplexer multiplexes six T3 lines. Each of hese lines have 44.736 Mbps bit rate.
The T_{4} line have 274.176 Mbps bit rate. Thus at the end of 4th level otal voice PCM channels would be
24\times4\times7\times6=4032 Thus T_{4} ine carries 4032 voice PCM channels.
The signaling rate of T_{4} line will be r=274.176\times10^{6} bps. Hence ransmission channel bandwidth will
be,
B_{T}\ge\frac{1}{2}r=\frac{1}{2}\times274.176\times10^{6}=137~MHz