DEC-2022

Q1
1) Necessity of de-emphasis & pre emphasis
Ans. De-emphasis and pre-emphasis are signal processing techniques used in audio and
telecommunications systems to improve signal quality and reduce noise.

Pre-emphasis is a technique applied to the audio signal before transmission or recording.

It boosts the higher frequencies of the signal and reduces the lower frequencies. This is
done because high-frequency components of an audio signal are more prone to
interference and noise during transmission or recording. By boosting these frequencies,
pre-emphasis helps to improve the signal-to-noise ratio and enhance the overall
intelligibility and clarity of the audio signal. Pre-emphasis is commonly used in FM
(Frequency Modulation) broadcasting and audio recording systems.

De-emphasis, on the other hand, is the inverse process of pre-emphasis. It is applied to

the audio signal after transmission or playback to restore the original frequency balance.
De-emphasis attenuates the higher frequencies and amplifies the lower frequencies. This
is necessary because during transmission or recording, the high frequencies were
boosted, and without de-emphasis, the audio signal would have an imbalanced frequency
response. De-emphasis ensures that the original audio signal is accurately reproduced and
eliminates the unwanted emphasis introduced during pre-emphasis. De-emphasis is
commonly used in FM receivers and audio playback systems.

Both pre-emphasis and de-emphasis are typically implemented using analog or digital
filters. The specific characteristics of the filters, such as cutoff frequency and slope, are
defined by industry standards or system requirements.

In summary, pre-emphasis and de-emphasis are important techniques in audio and

telecommunications systems to improve signal quality, reduce noise, and maintain the
integrity of the original audio signal throughout the transmission or recording process.

2) AM & FM difference?
Ans.

3) What is aliasing? How to avoid it?

Ans. Aliasing is a phenomenon that occurs in signal processing when a continuous signal is
sampled at a rate that is too low, resulting in the distortion or misinterpretation of the
original signal.

When sampling an analog signal, the continuous waveform is discretized by measuring its
amplitude at regular intervals. The Nyquist-Shannon sampling theorem states that to
accurately reconstruct a continuous signal from its samples, the sampling frequency
should be at least twice the highest frequency component present in the signal. If the
sampling rate is too low, frequencies above the Nyquist frequency (half the sampling rate)
will fold back and appear as lower frequencies, leading to aliasing.

4) A transmitter radiates a power of 9 kW when unmodulated and radiates power of

10.125 kW when modulated. Calculate the depth of modulation.
Ans. To calculate the depth of modulation, we need to determine the modulation index,
which is the ratio of the peak deviation of the modulated signal to the peak amplitude of
the unmodulated carrier signal. The depth of modulation is given by the modulation index
multiplied by 100.

Given that the transmitter radiates 9 kW when unmodulated and 10.125 kW when
modulated, we can calculate the modulation index.
Peak power of the modulated signal = 10.125 kW
Peak power of the unmodulated carrier signal = 9 kW
Let's assume the modulation index as "m."
The peak power of the modulated signal can be calculated using the formula:
Peak power of the modulated signal = (1 + m^2) * Peak power of the unmodulated carrier
signal
Substituting the values:
10.125 kW = (1 + m^2) * 9 kW
Simplifying the equation:
1 + m^2 = 10.125 kW / 9 kW
1 + m^2 = 1.125
Now, solving for "m":
m^2 = 1.125 - 1
m^2 = 0.125
m = √0.125
m ≈ 0.354
To calculate the depth of modulation, we multiply the modulation index by 100
Therefore, the depth of modulation is approximately 35.4%.

5) State advantages of pulse modulation over continuous modulation.

Ans. Pulse modulation, such as pulse amplitude modulation (PAM), pulse width modulation
(PWM), and pulse position modulation (PPM), offers several advantages over continuous
modulation techniques like amplitude modulation (AM) and frequency modulation (FM).
Some of the advantages of pulse modulation are:

1. Improved Signal-to-Noise Ratio (SNR): Pulse modulation techniques can achieve a higher
SNR compared to continuous modulation. This is because pulse modulation allows for
precise sampling and quantization of the signal, resulting in improved noise immunity and
better signal fidelity.

2. Efficient Bandwidth Utilization: Pulse modulation techniques are more

bandwidth-efficient than continuous modulation. By discretizing the continuous signal
into pulses, only the essential information is transmitted, resulting in reduced bandwidth
requirements.

3. Higher Transmission Efficiency: Pulse modulation techniques offer higher transmission

efficiency, especially in digital communication systems. By representing the signal as
discrete pulses, it becomes easier to encode and transmit digital information accurately,
allowing for higher data rates and better utilization of the communication channel.

4. Improved Power Efficiency: Pulse modulation techniques, such as pulse width

modulation (PWM), are widely used in power electronics for applications like motor
control and power conversion. By modulating the width or duration of pulses, PWM
enables efficient control of power devices, reducing power losses and improving overall
energy efficiency.
5. Scalability and Flexibility: Pulse modulation techniques can be easily adapted and scaled
for different applications and requirements. By adjusting parameters like pulse width,
amplitude, or position, pulse modulation allows for flexible customization according to
specific system needs.

These advantages make pulse modulation techniques preferable in various applications

such as digital communication systems, audio encoding, power electronics, and control
systems, where efficient and accurate representation of signals is critical.

Q2

1) Explain indirect fm transmitter.

Ans. Armstrong method of FM generation is the indirect method because the modulating
signal directly varies the phase of the carrier, which indirectly changes the frequency. The
Fig1 shows the block diagram of wideband FM generation through Armstrong method.

2) Explain superheterodyne radio receiver with block diagram and explain each block.
Ans.
Receiving antenna: The receiving antenna receives the signal which was sent by the
transmitter. It sends the received signal for further processing.

RF amplifier: The received signal is fed to the RF amplifier stage so as to amplify it, as the
signal gets attenuated during long-distance transmission. It is tuned in such a way that it
can choose the desired carrier frequency and amplify it.

Local Oscillator: This circuit basically generates a signal with a fixed frequency and the
output is then fed to the mixer. When we talk about AM broadcast systems, the
intermediate frequency is 455 KHz, which simply means that the local oscillator should
select such a frequency which is 455 KHz above the incoming signal frequency.

Mixer: A mixer simply mixes the carrier frequency with the frequency of the signal
generated by the local oscillator.

Here, two different frequencies are to be mixed so as to have another frequency

component of lower value. Now the thing that first comes to our mind is why the mixer
produces a lower frequency value, which is the difference between the two frequencies.
The summation of the carrier and local oscillator frequency at the output of the mixer will
give rise to image frequency which is treated as a type of noise or distortion in the signal.
This is the reason why the mixer generates a frequency difference at its output. This
difference frequency is a constant value irrespective of the variations in the input, known
as the intermediate frequency.

IF amplifier: This section basically amplifies the output of the mixer. IF amplifier provides
sensitivity(gain) and selectivity (bandwidth requirement) to the receiver. As it consists of
several transformers consisting of pairs of the tuned circuit.Here, the sensitivity and
selectivity are uniform and does not show variations as in case of TRF receivers because IF
amplifier’s characteristics are independent of that of the received signal frequency as it
works on the intermediate frequency.Due to this, the system design is quite easy so as to
provide constant bandwidth along with high gain.

Demodulator: Demodulator is placed exactly after the IF amplifier so that the constant
frequency signal is demodulated and the message signal can be extracted from it.

Audio amplifier: The original signal is fed to the audio amplifier which does not hold
distortion or noise so that it can amplify audio signal to a particular level.
Q3

1) What are the different methods for SSB generation? Explain any one.

Ans. The methods employed for SSB generation are as follows:

1. Filter Method
2. Phase-Shift Method
3. Third Method

Phase shift method :

The phasing method of SSB generation uses a phase shift technique that causes one of the
side bands to be canceled out. A block diagram of a phasing type SSB generator is shown in
fig.
2) Explain a balanced slope detector.

As shown in the circuit diagram, 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., fc .

The upper tuned circuit of the secondary (T1) is tuned above fc by Δf i.e., its resonant
frequency is (fc+ Δf).

The lower tuned circuit of the secondary is tuned below fc by Δf i.e., at (fc – Δf).

R1C1 and R2C2 are the filters used to bypass the RF ripple.
Vo1 and Vo2 are the output voltages of the two slope detectors.

The final output voltage Vo is obtained by taking the subtraction of the individual output
voltages, Vo1 and Vo2, i.e.,

Working Operation of the Circuit

The circuit operation can be explained by dividing the input frequency into three ranges
as follows:

(i) fin = fc: When the input frequency is instantaneously equal to fc, the induced voltage in
the T1 winding of secondary is exactly equal to that induced in the winding T2.

Thus, the input voltages to both the diodes D1 and D2 will be the same.

Therefore, their dc output voltages Vo1 and Vo2 will also be identical but they have
opposite polarities. Hence, the net output voltage Vo = 0.

(ii) fc < fin < (fc + Δf): In this range of input frequency, the induced voltage in the winding
T1 is higher than that induced in T2.

Therefore, the input to D1 is higher than D2.

Hence, the positive output Vo1 of D1 is higher than the negative output Vo2 of D2.

Therefore, the output voltage Vo is positive.

As the input frequency increases towards (fc + Δf), the positive output voltage increases as
shown in 3.
If the output frequency goes outside the range of (fc – Δf) to (fc + Δf), the output voltage
will fall due to the reduction in tuned circuit response.

Advantages

(i) This circuit is more efficient than a simple slope detector.

(ii) It has better linearity than the simple slope detector.

Drawbacks

(i) Even though linearity is good, it is not good enough.

(ii) This circuit is difficult to tune since the three tuned circuits are to be tuned at
different frequencies i.e., fc, (fc+Δf) and (fc – Δf).

(iii) Amplitude limiting is not provided.

Q4

2)

Ans. 1) PPM

1. In PPM the amplitude and width of the pulses is kept constant but the position of
each pulse is varied in accordance with the amplitudes of the sampled values of the
modulating signal. The position of the pulses is changed with respect to the
position of reference pulses
2. The PPM pulses can be divided from the PWM pulses. With increase in the
modulating voltage the PPM pulses shift further with respect to reference
3. The vertical treated dotted lines are reference lines to measure the shift in position
of PPM pulses. The PPM pulses go away from their respective reference lines
4. This is corresponding to an increase in the modulating signal amplitude. Then as
the modulating voltage decreases the PPM pulses come progressively closer to
their respective reference lines
Generation of PPM Signal

1. The PPM Signal can be generated from a PWM signal. The PWM pulse obtained at
the comparator output are applied to a mono stable multivibrator
2. Hence corresponding to each trailing edge of the PWM signal, the mono stable
output goes high. It remains high for a fixed time decided by its own RC
comparator.
3. Thus as the trailing edges of the PWM signal keep shifting in proportion with the
modulating signal x(t), the PWM pulses also keep shifting
4. All the PPM pulses have the same width and amplitude. The information is
conveyed via changing the portion of pulses.

Demodulation of PPM Signal

The PPM demodulator block diagram has been shown in fig.4

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 applies
these pulses to the reset pin (R) of a SR flip-flop.

A fixed period reference pulse is generated from the incoming PPM waveform and the SR
flip-flop 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.

The PWM signal can be demodulated using the PWM demodulator.

Q5

1) Draw and explain the FDM transmitter and receiver block diagram along with
applications.

Ans. FDM Transmitter

Each signal that needs to be sent over a communication channel undergoes modulation
with various carrier frequencies, as shown clearly in the diagram below. There are
different kinds of modulation such as amplitude modulation, pulse modulation, frequency
modulation etc.

As the name suggests, the modulation done here is frequency modulation by the FDM
transmitter.

These modulated signals are then added up using a linear adder or a mixer, forming a
composite signal which gets transmitted over a communication channel (single channel).

FDM Receiver
At the receiving end, the single composite signal is received by the FDM receiver.

The receiver then passes the composite signal through various band pass filters.

Each of these band pass filters has a frequency corresponding to the frequencies of one of
the carrier waves.

Each band pass filter will accept the signal whose frequency matches with the frequency
of the carrier signal and rejects all other channels.

The signals coming out of band pass filters pass through a demodulator.

The demodulator does the work of separating the original signal from the carrier signal.

Applications of FDM

FDM is used for FM & AM radio broadcasting. Each AM and FM radio station uses a
different carrier frequency. In AM broadcasting, these frequencies use a special band from
530 to 1700 KHz. All these signals/frequencies are multiplexed and are transmitted in air.
A receiver receives all these signals but tunes only one which is required. Similarly FM
broadcasting uses a bandwidth of 88 to 108 MHz

FDM is used in television broadcasting.

First generation cellular telephone also uses FDM.

2) With the help of a block diagram explain PCM.

Ans. Definition: A technique by which analog signal gets converted into digital form in
order to have signal transmission through a digital network is known as Pulse Code
Modulation. It is abbreviated as PCM.

Basic Elements of PCM

● The transmitter section of a Pulse Code Modulation circuit consists of Sampling,

Quantizing and Encoding, which are performed in the analog-to-digital converter
section.
● The low pass filter prior to sampling prevents aliasing of the message signal.
● The basic operations in the receiver section are regeneration of impaired signals,
decoding, and reconstruction of the quantized pulse train.
● Following is the block diagram of PCM which represents the basic elements of both
the transmitter and the receiver sections.

Low Pass Filter: This filter eliminates the high frequency components present in the input
analog signal which is greater than the highest frequency of the message signal, to avoid
aliasing of the message signal.

Sampler: This is the technique which helps to collect the sample data at instantaneous
values of the message signal, so as to reconstruct the original signal. The sampling rate
must be greater than twice the highest frequency component W of the message signal, in
accordance with the sampling theorem.

Quantizer: Quantizing is a process of reducing the excessive bits and confining the data.
The sampled output when given to Quantizer, reduces the redundant bits and compresses
the value.
Encoder: The digitization of analog signals is done by the encoder. It designates each
quantized level by a binary code. The sampling done here is the sample-and-hold process.
These three sections (LPF, Sampler, and Quantizer) will act as an analog to digital
converter. Encoding minimizes the bandwidth used.

Regenerative Repeater: This section increases the signal strength. The output of the
channel also has one regenerative repeater circuit, to compensate for the signal loss and
reconstruct the signal, and also to increase its strength.

Decoder: The decoder circuit decodes the pulse coded waveform to reproduce the original
signal. This circuit acts as the demodulator.

Reconstruction Filter:

● After the digital-to-analog conversion is done by the regenerative circuit and the
decoder, a low-pass filter is employed, called as the reconstruction filter to get
back the original signal.
● Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it
and samples it, and then transmits it in an analog form.
● This whole process is repeated in a reverse pattern to obtain the original signal.

Q6

2)

i) write a note on delta and adaptive delta modulation.

Ans. Delta modulation is basically analog to digital and a digital to analog signal conversion
technique which is mainly used for data transfer. This type of modulation technique is
used by the satellite business system, and it is also known as Differential pulse code
modulation. This modulation technique is used to achieve a high signal to noise ratio.
Delta Modulation is basically of three types that are Adaptive Delta Modulation(ADM),
Delta-sigma Modulation, and differential modulation. The ADM is a type of Delta
Modulation in which the step size is variable, also known as continuously variable slope
Delta Modulation.

This Modulation is the refined form of delta modulation. This method was introduced to
solve the granular noise and slope overload error caused during Delta modulation.

This Modulation method is similar to Delta modulation except that the step size is variable
according to the input signal in Adaptive Delta Modulation whereas it is a fixed value in
delta modulation.
The transmitter circuit consists of a summer, quantizer, Delay circuit, and a logic circuit
for step size control. The baseband signal X(nTs) is given as input to the circuit. The
feedback circuit present in the transmitter is an Integrator. The integrator generates the
staircase approximation of the previous sample.

At the summer circuit, the difference between the present sample and staircase
approximation of previous sample e(nTs) is calculated. This error signal is passed to the
quantizer, where a quantized value is generated. The step size control block controls the
step size of the next approximation based on either the quantized value is high or low. The
quantized signal is given as output.

At the receiver end Demodulation takes place. The receiver has two parts. First part is the
step size control. Here the received signal is passed through a logic step size control block,
where the step size is produced from each incoming bit. Step size is decided based on
present and previous input. In the second part of the receiver, the accumulator circuit
recreates the staircase signal. This waveform is then applied to a low pass filter which
smoothens the waveform and recreates the original signal.

ii) VSB in television broadcasting

The information stored in LSB (lower side band) and USB ( upper side band) is similar, and
the bandwidth requirement for television signal (audio + video) is large. To reduce the
transmitted power wastage, it is preferred to transmit a single band i.e. either LSB or USB.

Now, SSB can not be used because of these 2 reasons:

1. Receiver of SSB is more complex.

2. Some parts of LSB are shared with USB or vice-versa which creates
distortion.

3. Ideal band pass filters are difficult to produce. So some of the signal gets
lost if the filter is not ideal.
And, DSB can not be used because of the requirement of large bandwidth in television
signals.

To overcome the above mentioned constraints we use VSB i.e. vestigial side band
modulation, in which Vestigial means ‘ extra’. To avoid the loss of information due to the
presence of a single side band, an extra little amount of band is added to it.

Basically, VSB modulation lies between SSB and DSB modulation, whose purpose is to save
the bandwidth, to reduce the transmitted power and to avoid the distortion.

MAY-2022

Q2

1) Explain use of VSB in television broadcasting.

2) A transmitter radiates a power of 9 kW when unmodulated and radiates power of
10.125 kW when modulated. Calculate the depth of modulation. (solved above)

3) Explain types of AGC.

Ans. i. Simple AGC: the gain control mechanism is active for high as well as low value of
carrier voltage.
ii. Delayed AGC: AGC bias is not applied to the amplifiers until signal strength crosses a
predetermined level, after which AGC bias is applied.

4) Define and explain snr, noise figure, noise factor, noise temperature, Friss formula.

Ans. The Signal-to-Noise Ratio (SNR) is a measure of the relative strength of the desired
signal compared to the background noise level in a communication or signal processing
system. It quantifies the quality and clarity of the signal and is usually expressed in
decibels (dB). A higher SNR indicates a better signal quality.

Noise Figure (NF) is a measure of how much additional noise a device or component
introduces into a signal compared to an ideal noiseless component. It quantifies the
degradation in signal quality due to the presence of noise in the system. Noise Figure is
expressed in decibels (dB) and is defined as the ratio of the output noise power of the
device to the input noise power.

Noise Factor (F) is the linear equivalent of Noise Figure. It represents the ratio of the
output noise power of a device to the input noise power, without being expressed in
decibels. Noise Factor and Noise Figure are mathematically related, where the Noise
Figure in decibels is 10 times the logarithm (base 10) of the Noise Factor.
Noise Temperature (T) is a measure of the equivalent noise power in a device or system,
expressed in temperature units (Kelvin). It represents the temperature that a noise source
would need to be at to produce the same amount of thermal noise as observed in the
device or system. A lower noise temperature indicates lower noise levels and better
system performance.

5) Explain how PPM is generated from PWM

Same as the answer of question 4.2 of above.

6) Explain TDM and FDM applications.

Ans. Time Division Multiplexing (TDM) and Frequency Division Multiplexing (FDM) are
both multiplexing techniques used in telecommunications and data transmission to
transmit multiple signals over a single communication channel. While TDM divides the
channel into time slots, FDM divides it into frequency bands. Here are some applications
of TDM and FDM:

Applications of TDM:

1. Digital Telephony: TDM is widely used in digital telephony systems, such as ISDN
(Integrated Services Digital Network) and T1/E1 lines. It allows multiple voice channels to
be combined and transmitted over a single physical line by allocating each channel a
specific time slot. TDM enables efficient and simultaneous transmission of voice calls.

2. Data Communication: TDM is commonly used in data communication systems, such as

synchronous serial communication interfaces. It allows multiple data streams to be
multiplexed onto a single transmission line, increasing the overall data transmission
capacity. TDM is also used in network switches and routers to handle multiple data
streams concurrently.

3. Digital Broadcasting: In digital broadcasting systems, such as Digital Audio Broadcasting

(DAB) or Digital Video Broadcasting (DVB), TDM is used to combine multiple audio or video
signals into a single digital stream for transmission. TDM enables efficient utilization of
the available bandwidth.

Applications of FDM:

1. Analog Television Broadcasting: FDM has been extensively used in analog television
broadcasting systems. Each TV channel is assigned a specific frequency band, and multiple
channels are transmitted simultaneously over the air or cable networks. FDM enables the
simultaneous transmission of different television channels.

2. Radio Broadcasting: FDM is widely used in FM (Frequency Modulation) and AM

(Amplitude Modulation) radio broadcasting. Different radio stations are assigned specific
frequency bands, allowing them to broadcast their signals concurrently without
interference. FDM enables the availability of a wide range of radio stations for listeners.

3. Cable Television (CATV): FDM is employed in cable television systems to deliver multiple
television channels to subscribers. Each channel is allocated a different frequency band,
and these channels are combined and transmitted over coaxial or fiber-optic cables. FDM
enables cable TV providers to offer a wide variety of channels to subscribers.

4. Broadband Internet Access: FDM is used in broadband internet access technologies like
Digital Subscriber Line (DSL) and cable internet. Multiple data channels are allocated
different frequency bands, enabling simultaneous transmission of voice, data, and video
signals over the same physical medium. FDM enables high-speed internet access and
multimedia services.

In summary, TDM and FDM find numerous applications in various communication

systems, including telephony, data communication, broadcasting, and broadband access.
They allow efficient utilization of the available bandwidth, enabling the transmission of
multiple signals simultaneously over a single channel.

Remaining entire paper is same 🤕