Mam Suggestions: -

19.) Comparison between (QM/QAM) and FDM :

Bandwidth efficiency:
" [CH10.1C3] DSB-SC:
By frequency spectrum shifting to passband, BW of
DSB-SC signal is twice of BW of the information
signal.
For example,
BW offf 1tt =WW1 then, BW off f 1 t
COS wWC c t =2WW1

" QM: ldentical carrier frequency forf f 1t t and

ff 21 t
B[CH10.1 C3] Spectrum off f 1t t cos wwc c
I I andf f 2t I sin wwc c I t overlap in
passband.
Share the same band.

B[CH10.1A2] Total BW of QM is equal to single DSB

-SC signal.
B[CH10.1c4) Bandwidth Efficient.

" FDM: Disjoint spectrum in passband

SRequires separate BW for each signals
Bandwidth inefficient
8 Phase error:

" QM:Cross interference betweenf f 1t t and

ff 21 t in addition to SNR loss.
"FDM: No cross interference among the
multiplexed signals. SNR loss only.

5.) How diagonal clipping can be avoided?

Diagonal clipping can refer to several different
phenomena, but in the context of machine learning,
it typically refers to a situation where the gradient of
a neural network's weights becomes aligned with
the principal axes of the weight matrix, leading to a
loss of representational capacity and slower
convergence.

There are several approaches to avoid diagonal

clipping:

Weight Initialization: One way to prevent diagonal

clipping is to initialize the weights of the neural
network in a way that prevents them from
becoming aligned with the principal axes of the
weight matrix. For example, Xavier or He
initialization methods can be used to set the initial
weights in a way that ensures the variance of the
input and output to each layer are balanced.

Batch Normalization: Batch normalization is a

technique used to normalize the inputs to a layer,
which can help prevent diagonal clipping by
ensuring that the inputs to each layer have similar
distributions. This can help to prevent the gradient
from becoming aligned with the principal axes of
the weight matrix.
Gradient Clipping: Another way to avoid diagonal
clipping is to use gradient clipping, which involves
scaling the gradient so that its norm is within a
certain range. This can prevent the gradient from
becoming too large and leading to numerical
instability, while also preventing it from becoming
too small and slowing down convergence.
Weight Decay: Weight decay is a regularization
technique that involves adding a penalty term to the
loss function to encourage the weights to stay
small. This can help prevent diagonal clipping by
limiting the magnitude of the weights, which can in
turn prevent the gradient from becoming too large.
5.) Envelope detector with proper diagram :
Jua
OCopy code

RF 1n -|DiOde

load
C

GND

7.) PLL FM Demodulator (Phase Locked Loop FM

Demodulator): -
PLL FM Demodulator: -
A Phase-Locked Loop (PLL) is basically a negative
feedback system. It consists of three major
components such as re multiplier, a loop filter and a
voltage controlled oscillator (VCo) connected
together in the form of a feedback loop.A VCO is a
sine wave generator whose frequency is determined
by the voltage applied to it from an external source.
It means that any frequency modulator can work as
a VC0.

A phase-locked loop (PLL) is primarily used in

tracking the phase and frequency of the carrier
component of an incoming FM signal.PLL 0s also
useful for synchronous demodulation of AM-SC (i.e.,
Amplitude Modulation with Suppressed carier)
signals or signals with few cycles of pilot carrier.
Further, PLL is also useful for demodulating FM
signals in presence of large noise and low signal
power.
This means that, PLL 0s most suitable for use in
space vehicle-to-earth data links or where the loss
along the transmission line or path is quite large.
Recently, it has found application in commercial FM
receivers.

The block diagram of a PLL is shown in fig.1 below.

FM wave e(t) Loop

s(t) filter
v(t)

b()
VCO

Working Operation
The operation of a PLL is similar to any other
feedback system where the feedback signal tends
to follow the input signal.
If the signal fed back is not equal to the input signal,
the error signal will change the value of the fed back
signal until it is equal to the input signal.
The difference signal between s(t) and b(t) is called
an error signal.A PLL operates on a similar principle
except for the fact that the quantity feedback is not
the amplitude, but a generalized phase (t).
The error signal or difference signal e(t) is utilized
to adjust the VCO frequency in such a way that the
instantaneous phase angle comes close to the
angle of the incoming signal s(t).
At this point, the two signals s(t) and b(t) are
synchronized and the PLL is locked to the incoming
signal s(t).
9.) i) What is image frequency and how it can be
remove by RF sections( explain with proper
equation).i) Why LO frequency is kept higher than
the signal frequency. ii) Why two stage filtering
method is used for SSB generation?

i) Ans:- Localoscillator (L0) frequency is kept

higher than the signal frequency in many
Communication systems, such as in radio
receivers and transmitters, for a few
reasons:Frequency Conversion: One of the
main reasons for keeping Lo frequency higher
is frequency conversion. In radio receivers, the
incoming signal is usually ata very low
frequency, such as in the range of kHz or MHz.
The receiver needs to convert this frequency to
an intermediate frequency (IF) that can be
processed more easily. The LO frequency is
mixed with the incoming signal to produce the
IF. By using a higher LO frequency, the IF can
be selected at a fixed frequency, regardless of
the frequency of the incoming signal. This
 simplifies the design of the receiver, and
reduces the number of components
required.Image Frequency Rejection: Another
reason for keeping LO frequency higher is to
reject the image frequency. The image
frequency is the mirror image of the incoming
signal frequency with respect to the LO
frequency. If the LO frequency is too low, the
image frequency can fall within the band of
interest, causing interference. By using a higher
LO frequency, the image frequency is pushed
farther away from the band of interest, making
iteasier to reject.Spurious Response Rejection:
Keeping the LO frequency higher than the
signal frequency also helps to reduce spurious
responses. These are unwanted responses that
can be generated in the receiver due to non
linearities in the system. By using a higher LO
frequency, the spurious responses are pushed
farther away from the band of interest, making
them easier to reject.Overall, the choice of LO
frequency depends on the specific
requirements of the system, such as the
frequency range of interest, the level of
interference, and the desired performance.

ii) Ans :- Single sideband (SSB) modulation

is a technique used in radio communication to
 transmit audio signals with reduced bandwidth.
In SSB modulation, only one of the sidebands
(upper or lower) is transmitted, along with the
carrier signal. This reduces the bandwidth of
the transmitted signal by half, compared to
traditional amplitude modulation (AM), where
both sidebands are transmitted along with the
carrier.

To generate an SSB signal, a double-sideband (DSB)

signal is first generated using a mixer or modulator.
The DSB signalcontains both the upper and lower
sidebands, along with the carrier. The DSB signal is
then filtered using a bandpass filter to remove one
of the sidebands and the carrier, leaving only the
desired SSB signal.
Atwo-stage filtering method is often used to
generate SSB signals because it provides better
performance compared to a single-stage filter. In
the two-stage filtering method, the DSB signal is
first filtered using a low-pass filter to remove the
carrier frequency and one of the sidebands. The
resulting signal is then fed into a second bandpass
filter, which removes the remaining sideband and
passes only the desired SSB signal. This two-stage
filtering method provides better selectivity, which
means that unwanted signals and noise outside the
desired frequency band are more effectively
rejected.
In addition to better selectivity, the twO-stage
filtering method also provides better attenuation of
unwanted signals and noise compared toa single
stage filter. This is important because unwanted
signals and noise can degrade the quality of the
transmitted audio signal. The two-stage filtering
method also provides better frequency stability, as
the low-pass filter can remove any frequency
variations introduced by the modulator or mixer.
Overall, the two-stage filtering method is a popular
technique for SSB generation because it provides
better performance, including improved selectivity,
attenuation of unwanted signals and noise, and
frequency stability.

12.) How DSBSC signalcan be generated from ring

modulator?
DSBSC (Double-Sideband Suppressed Carrier)
signal can be generated from a ring modulator by
applying two signals to the input ports of the ring
modulator. One of these signals is the carrier signal,
while the other is the modulating signal.
The ring modulator is a non-linear device that uses
a diode ring to mix two input signals. The output of
the ring modulator contains the sum and difference
frequencies of the two input signals. However, the
carrier frequency is suppressed due to the non
linear nature of the ring modulator.
To generate a DSBSC signal using a ring modulator,
the carrier signal must be suppressed while
retaining the sum and difference frequencies. This
can be achieved by adding a DC bias to the input
signal that corresponds to the carrier frequency.
The DC bias will cause the carrier frequency to be
suppressed at the output of the ring modulator.
The resulting signal will be a DSBSC signal with the
same bandwidth as the modulating signal. The
carrier frequency will be absent from the signal, but
the signal will contain both upper and lower
sidebands around the center frequency. The
amount of suppression of the carrier frequency and
the amplitude of the sidebands will depend on the
input power levels and the characteristics of the
ring modulator.

13.) Why macoswmt<<2 for this modulator?

As an Al language model, Icannot provide a
definitive answer without additional context and
information about the specific modulator you are
referring to. However, Ican provide some general
information about the bitwise left shift operator (<<)
and its usage in programming.
In general, the bitwise left shift operator (<<) shifts
the bits of a binary number to the left by a specified
number of positions. This can be useful for
performing multiplication by powers of 2, which can
be more efficient than performing multiplication
using traditional arithmetic operations.
The specific expression "macoswmt<<2" likely
involves a variable named "macoswmt" being
shifted to the left by 2 bits. This could be used in a
variety of ways depending on the context of the
code. For example, it could be used to perform a
multiplication by a factor of 4, or to extract a
specific bit pattern from the variable.
Without additional information, it is difficult to
provide amore specific answer.

14.)For tone modulation determine the upper limit

of RC to ensure that the capacitor voltage follows
the envelope in envelope detector.?
In a tone modulation circuit, the upper limit of RC
(the product of the resistance Rand the
capacitance C) determines the time constant of the
RC circuit,which in turn affects the response time
of the envelope detector.
The envelope detector is a circuit used to extract
the envelope of the modulated signal, which is a low
frequency signal that represents the amplitude
variations of the original modulating signal. The
time constant of the RC circuit should be chosen
such that it is long enough to follow the variations
in the envelope but short enough to prevent
significant distortion of the modulated signal.
Assuming a simple diode-based envelope detector,
the charging time of the capacitor is given by RC,
where R is the resistance of the load resistor and C
is the capacitance of the capacitor. The discharge
time of the capacitor is determined by the time
constant of the resistor and capacitor that form the
RC circuit connected across the load resistor.
To ensure that the capacitor voltage follows the
envelope in the envelope detector, the time
constant of the RC circuit should be at least 5 to 10
times larger than the period of the modulating
signal. This ensures that the capacitor has enough
time to charge and discharge fully during each cycle
of the modulating signal, and reduces distortion of
the envelope.
Therefore, the upper limit of RC can be calculated
as:RC < T/(5 to 10)
where T is the period of the modulating signal.
Note that the specific value of RC that is optimal for
a given tone modulation circuit may depend on
various factors such as the frequency range of the
modulating signal, the input impedance of the
circuit, and the specific components used.
10.) The antenna current of an AM broadcast
transmitter, modulated to a depth of 40percent by
an audio sine wave is 11A. It increases to 12A as a
result of sinusoidal modulation by another audio
sine wave. What is the modulation index due to
second one?
Ans:-The modulation index (m) is defined as the
ratio of the amplitude of the modulating signal to
the amplitude of the carrier signal.Given that the
antenna current of the AM broadcast transmitter,
modulated to a depth of 40 percent by an audio sine
wave is 11A, we can determine the carrier current
as follows:
Carrier current = (100% - 40%) * Antenna current
= 60% * 11A
= 6.6A

Let's assume that the modulating signal for the first

case is represented by m1 and the modulating
signal for the second case is represented by m2.
We can use the formula for AM modulation to relate
the modulation index to the amplitude of the
modulating signal and the carrier current as follows:
Antennacurrent = Carrier current *(1 + m*
cos(2pif*t)
where f is the frequency of the modulating signal
and t is the time.
For the first case, we have:
11A =6.6A *(1 + m1 * cos(2pift)
Simplifying, we get:
m1 = (11A/6.6A -1)/cos(2pift)
For the second case, we have:
12A =6.6A * (1 +m2 *cos(2pift))
Simplifying, we get:
m2 = (12A/6.6A - 1)/cos(2pif*t)
We can now calculate the modulation index due to
the second modulating signal as follows:
m2- m1 =[(12A/6.6A- 1)-(11A/6.6A -
1)J/cos(2pif)
Simplifying, we get:
m2-m1 =0.054/cos(2pif*t)
Therefore, the modulation index due to the second
modulating signal is 0.054/cos(2pif*t).
11.) A
500 Wcarrier is modulated on a depth of
50%.Calculate the total power in the modulated
wave in the following forms of AM: i) DSB with full
carrier. li) DSB with suppressed carrier.?
Ans :-i) DSB-FC (Double Sideband Full Carrier) AM:
In DSB-FC AM, both upper and lower sidebands are
transmitted alongwith the full carrier wave. The
total power in the modulated wave can be
calculated as follows:

Total power = Carrier power + Sideband power

Carrier power = (Ac)^2/2, where Ac is the amplitude
of the carrier wave

Sideband power = (Am)^2/2, where Am is the

amplitude of the modulating signal
Given, carrier power = 500 W
Modulation depth = 50%, which means Am = 0.5 Ac
Substituting the values, we get:
Carrier power = (Ac)^2/2 = 500 W
Amplitude of the modulated signal, Ac(1 + m) =
Ac(1 + 0.5) = 1.5 Ac
Sideband power = (Am)^2/2 = (0.5 Ac)^2/2 =
(Ac^2)/8
Total power = Carrier power + Sideband power
Total power = (Ac)^2/2 + (Ac^2)/8 = (5/8) (Ac)^2
 Equatingthe total power to 500 W, we get:
(5/8) (Ac)^2 = 500
Ac^2 = 800
Ac = V800 28.28 V
Therefore, the total power in the modulated wave in
DSB-FC AM is (5/8) (Ac)^2 =(5/8) (28.28)^2 250
W.

i) DSB-SC (Double Sideband Suppressed Carrier)

AM:

In DSB-SC AM, both upper and lower sidebands are

transmitted without the carrier wave. The total
power in the modulated wave can be calculated as
follows:

Total power = Sideband power

Sideband power = (Am)^2/2, where Am is the
amplitude of the modulating signal
Given, carrier power = 500 W
Modulation depth = 50%, which means Am = 0.5 Ac
Substituting the values, we get:
Sideband power = (Am)^2/2 = (0.5 Ac)^2/2 =
(Ac^2)/8
Total power = Sideband power
Totalpower =(Ac^2)/8
Equating the total power to 500 W, we get:
(Ac^2)/8 = 500
Ac^2 = 4000

Ac = V4000 63.25 V
Therefore, the total power in the modulated wave in
DSB-SC AM is (Ac^2)/8 = (63.25)^2/8 500 W.

16.) Prove that Pt=Pc(1+m2/2) where Pt=total

power in AM signal, Pc=carrier power,
m=modulation index.?

Ans:-To prove that Pt=Pc(1+m^2/2) for an AM

signal, we need to use the expression for the total
power of an AM signal and the definition of
modulation index.The total power of an AM signal
can be expressed as: Pt = Pc (1 + m^2/2)
where Pt is the total power, Pc is the power of the
carrier signal, and m is the modulation index.
The modulation index is defined as the ratio of the
amplitude of the modulating signal to the amplitude
of the carrier signal. Mathematically, it can be
expressed as: m= (Am/Ac)
where Am is the amplitude of the modulatingsignal
and Ac is the amplitude of the carrier signal.
To prove the equation Pt=Pc(1+m^2/2), we need to
substitute the expression for m in terms of Am and
Ac into the expression for Pt:
Pt=Pc (1 + m^2/2)
= Pc (1+ (Am/Ac)^2/2)
=Pc (1+Am^2/(2 Ac2)
=Pc+ Pc (Am^2/(2 Ac^2))
= Pc + Pm

where Pm is the power in the modulated signal.

We can see that the total power of the AM signal is
equal to the sum of the power in the carrier signal
(Pc) and the power in the modulated signal (Pm).
Since the amplitude of the modulated signal is
proportional to the modulation index, we can write:
Pm = Pc (m^2/2)
Substituting this expression for Pm into the
equation for Pt, we get:
Pt =Pc + Pc (m^2/2)
= Pc (1 + m^2/2)
Thus, we have proved that Pt = Pc (1+ m^2/2) for
an AM signal, using the definition of modulation
index and the expression for total power in an AM
signal.
18.) A parallel tuned circuit has a resonant
frequency fr=10MHz.Its Q=20 and the value of
capacitor is 10pf.If the ambient temp. is 178C,
calculate the noise voltage across the parallel
tuned circuit ?
Ans: -To calculate the noise voltage across the
parallel tuned circuit, we need to use the Johnson
Nyquist noise formula: Vn = sqrt(4 *k*T*B*R)
Where:
Vn is the noise voltage
kis theBoltzmann constant (1.38 x 10^-23 J/K)
T is the absolute temperature in Kelvin (17°C =
290K)
Bis the bandwidth (in Hz)
R is the equivalent resistance of the circuit
To calculate the equivalent resistance of the
parallel tuned circuit, we can use the formula:
Q=1/R* sqrt(L/ c)
Where:

Q is the quality factor (given as 20)

Lis the inductance of the circuit (unknown)
Cis the capacitance of the circuit (given as 10pF)
Solving for L:
 Q=1/R* sqrt(L/ C)
20 = 1/R* sqrt(L/10pF)
20 *R= sqrt(L/ 1OpF)
400 * R^2 = L/10pF
L= 400 * R^2 * 1OpF
L= 80nH
Now we can calculate the bandwidth of the circuit
using the formula:
B= fr/Q
B=10MHz / 20
B= 500kHz

Finally, we can calculate the noise voltage using the

Johnson-Nyquist noise formula:
Vn = sqrt(4 *k *T*B* R)
Vn = sgrt(4 * 1.38e-23 * 290* 500e3 *2* pi *
10MHz * 80e-9)
Vn = 2.5 nV

Therefore, the noise voltage across the parallel

tuned circuit is 2.5 nV.
20.) For a receiver with IF and RF frequencies of
455Khz and 900 Khz respectively,determine the
following: i) local oscillator frequency, image
frequency, image frequency rejection ratio for a
preselector Q of 80 ?
Ans:-To determine the local oscillator frequency
and image frequency, we can use the following
formulas:

Local oscillator frequency = RF frequency - IF

frequency
Image frequency = 2 x RF frequency - IF frequency
Therefore, for the given values of RF frequency and
IF frequency, we have:
Local oscillator frequency = 900 kHz - 455 kHz =
445 kHz

Image frequency = 2 x 900 kHz- 455 kHz = 1345

kHz

Now, to calculate the image frequency rejection

ratio, we can use the formula:
Image frequency rejection ratio = f(F) / (f(image) -
f(RF))
Where:
f(IF) = intermediate frequency =455 kHz
f(RF) =radio frequency =900 kHz
f(image) = image frequency =1345 kHz
Q = preselector Q=80
Substituting the values, we get:
Image frequency rejection ratio = 455 kHz / (1345
kHz- 900 kHz) = 455 kHz/445 kHz =1.02
Therefore, the image frequency rejection ratio for a
preselector Q of 80 is approximately 1.02.

21.) Two resistors 10Kohm and 25kohm are at

room temperature (2908k) for a bandwidth of 150
KHZ.Calculate thermal noise for each resistor, if
two resistors are in series and in parallel ?
Ans:-Thethermal noise power generated by a
resistor can be calculated using the following
formula: P = 4*k*T*B *R
where P is the power spectral density of the thermal
noise in watts/Hz, k is the Boltzmann constant
(1.38 x 10^-23 J/K), T is the temperature in Kelvin, B
is the bandwidth in Hz, and R is the resistance in
ohms.

For the first resistor (10kohm):

P=4*1.38e-23 * 290* 150000 * 10000
P= 2.84e-13 watts/Hz
For the second resistor (25kohm):
P=4*1.38e-23 * 290* 150000 * 25000
P=7.1Oe-13 watts/Hz
When two resistors are connected in series, the
total resistance is the sum of the individual
resistances (R_series = R1 + R2). Therefore, the
thermal noise power for the series combination is:
P_series = 4 * 1.38e-23 * 290 * 150000 * (10000 +
25000)
P_series =1.06e-12 watts/Hz
When tWo resistors are connected in parallel, the
total resistance is given by the formula:
1/R_parallel = 1/R1 + 1/R2
Therefore, the thermal noise power for the parallel
combination is: P_parallel = 4 *1.38e-23 * 290 *
150000* (1/10000 + 1/25000)^-1
P_parallel = 2.42e-13 watts/Hz
So, the thermal noise power for the series
combination is higher than the thermal noise power
for the individual resistors and the parallel
combination. This is because the total resistance of
the series combination is higher, which results in a
higher thermal noise power. Conversely, the thermal
noise power for the parallel combination is lower
than the thermal noise power for the individual
resistors and the series combination, because the
total resistance of the parallel combination is lower,
which results in a lower thermal noise power.