solution for principle of analog and digital communication · solution for principle of analog and...

Mumbai University I.T (Semester -III)

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Solution for Principle of Analog

and Digital Communication

December 2015

Index

Q.1)

a) …………………………………………………………………………. 2-3

b) …………………………………………………………………………. 3-4

c) …………………………………………………………………………. 5-7

d) …………………………………………………………………………. 7-8

Q.2)

a) …………………………………………………………………………. N.A

b) …………………………………………………………………………. N.A

Q.3)

a) …………………………………………………………………………. 9-13

b) …………………………………………………………………………. 14-17

Q.4)

a) …………………………………………………………………………. 17-21

b) …………………………………………………………………………. 21-22

Q.5)

a) …………………………………………………………………………. 23-24

b) …………………………………………………………………………. 25-26

Q.6)

a) …………………………………………………………………………. 27

b) ………………………………………………………………………….28-29

c) …………………………………………………………………………. 29-31

d) …………………………………………………………………………. 32-33

e) ………………………………………………………………………… 33-35



Q1)

a) Explain pre-emphasis and de- emphasis

ANS: PRE-EMPHASIS

*It has been proved 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

for higher modulating frequencies

*This can be done by increasing the deviation and it can be increased by increasing

the amplitude of modulating signal at higher modulating 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.

DE- EMPHASIS

*The artificial boosting given to the higher modulating frequencies in the process

of pre-emphasis is nullified or compensated at the receiver by a process called ''De-

emphasis''.

* The artificially boosted high frequency signals are brought to their original

amplitude using the de-emphasis circuit.

*The 75mueseconds de-emphasis circuit is standard .The demodulated FM is

applied to the De-emphasis circuit with increase in fm the reactance of C goes on

decreasing and the output of de- emphasis circuit will also reduce.

b) Explain the following.i)Shot noise ii)Equivalent noise temperature

Ans: The shot noise is produced due to shot effect. Due to the shot effect, shot

noise is produced in all the amplifying devices rather in all the active devices.



*The shot noise is produced due to the random variations in the arrival of electrons

at the output electrode of an amplifying device.

* Therefore it appears as a randomly varying noise current superimposed on the

output .The shot noise 'sounds' like a shower of lead shots falling on a metal sheet.

*The shot noise has a uniform spectral density like thermal noise. The exact

formula for the shot noise can be obtained only for diodes.

# Equivalent noise temperature:

The noise at the input of the amplifier input is given by

Pna =(F - 1)k TOB. This is the noise contributed by the amplifier .This noise

power can be alternatively represented by some fictitious temperature Teq such that,

k TeqB = =(F - 1)k TOB

Thus the equivalent noise temperature of the amplifier is given by,

Teq =(F - 1)TO This equation shows that Teq is just an alternative measure for

F.

(c) A single tone FM signal is given by VFM(t) =10 sin (16 π x 106 t + 20

sin 2π x 103 t) .

Find (a) Maximum frequency deviation

(b)BW of FM by using Carson’s rule



Ans: Note: The sum below is a similar sum for reference .

d) What are the drawbacks in DM system and how these drawbacks can be

overcome.

Ans: The draw backs of delta modulation are as follows:

1) The required bandwidth is increased due to digital technology.

2) System complexity is increased.

3) In order to convert the analog signal to digital prior to transmission and then

from digital to analog at the receiver, we need to use the additional encoders and

decoders circuits.

4)Synchronization is necessary for the digital system .

5) Digital transmission system are not compatible to the older analog transmission

systems.

6) The two distortion are slope overload and granular noise .



Slope Overload distortion :



D.M. Bit Rate (Signaling Rate)

How it can be avoided

The slope overload error can be reduced by using another type of delta modulation

,called as Adaptive delta Modulation

e) Explain the need for modulation

Ans: 1) Reduction in height od antenna : For the transmission of radio signals ,the

antenna height must be multiple of (lambda /4). Here lambda is wavelength =c/f

where c is velocity of light and f is the frequency of the signal to be transmitted.



This antenna can be easily installed practically .Thus modulation reduces the

height of the antenna.

2) Avoid mixing of signals : If the baseband signals are transmitted without using

the modulation by more than one transmitter , then all the signals will be in the

same frequency range i.e. 0 to 20khz.Therefore all the signals get mixed together

and a receiver cannot separate them from each other. So if each baseband sound

signal is used to modulate a different carrier then they will occupy different slots in

the frequency domain.

3) Increase the range of communication : The frequency of baseband signals is low

,and low frequency signals cannot travel a long distance when they are

transmitted. They get heavily attenuated .The modulation process increases the

frequency of the signal to be transmitted .Hence it increases the range of

communication.

4) Multiplexing is possible : Multiplexing is a process in which two or more signals

can be transmitted over the same communication channel simultaneously. This is

possible only with modulation .The multiplexing allows the same channel to be

used by many signals . So many TV channels can use the same frequency range

without getting mixed with each other.

5)Improve quality of reception: With frequency modulation , and the digital

communication techniques like PCM, the effect of noise is reduced to a great

extent. This improves quality of reception.



Q2) A) Explain the basic principles , working of Transmitter and Receiver and

BPSK.Also draw the BPSK waveform for the following binary signal 10110110.

Ans N.A

(b) Draw the data format (line codes) of any five for the given binary signal 10101101

Ans: N.A.

Q3)

a) Explain PWM generation and degeneration method.

Ans: PWM is a type of pulse of analog modulation. In PWM, the width of the modulated pulses varies in proportion with the amplitude of modulating signal. The amplitude and the frequency of the PWM waves remains constant. Only the bandwidth changes. Therefore, the ' information 'is contained in the form of the width variation.

GENERATION:



* A sawtooth generates a saw tooth signal of frequency fc therefore the sawtooth signal in this case is is a sampling signal. It is applied to the inverting terminal of a comparator.

* The modulating signal x(t) is applied to the non-inverting terminal of the same comparator.

*The comparator output will remain hish as long as the instantaneous amplitude of x(t) is higher than that of the ramp signal.

* This gives rise to a PWM signal at the comparator output.

* Note that the leading edges of PWM waveform coincide with the falling edges of the ramp signals.

* Thus the leading edges of PWM signals are always generated at fixed time instants . However the occurrence of its trailing edges will be dependent on the instantaneous amplitude of x(t).

DETECTION :



Detection Circuit of PWM

* The PWM signal received at the input of the detection circuit is contaminated with noise. This signal is applied to pulse generator circuit which regenerates the PWM signal. Thus some of the noise is removed and the pulses are squared up.

* The generated pulses are applied to a reference pulse generator .It produces a train of constant amplitude ,constant width pulses . These pulses are synchronized to the leading edges of the regenerated PWM pulses but delayed by a fixed interval.

* The regenerated PWM pulses are also applied to a ramp generator. At the output of it we get a constant slope ramp for the duration of the pulse. The height of the ramp is thus proportional to the widths of the pwm pulses .

*The constant amplitude pulses at the output of reference pulse generator are then added to the ramp signal at the output of the clipper.

* A low pass filter is used to recover the original modulating signal back from the PAM signal .



Spectrum of PWM

Advantages

Disadvantages



b) Describe the PCM and also explain the PCM encoder and decoder with block diagrams.

ANS: PCM is a type of pulse modulation like PAM , PWM or PPM but there is an important difference between them. PAM, PWM or PPM are 'analog' pulse modulation systems whereas PCM is a digital pulse modulation system.

*That means PCM output is in the coded digital form. It is in the form of digital pulses of constant amplitude, width and position.

* The information is transmitted in the form of 'code words’. A PCM system consist of a PCM encoder and decoder.

* The essential operations in the PCM transmitter are sampling , quantizing and encoding. All these operations are usually performed in the same circuit called as analog- digital converter.

* It should be understood that PCM is not a modulation in the conventional sense.Because in modulation one of the characteristics of the carrier is varied in proportion with the amplitude of the modulation.

PCM ENCODER:

* The analog signal is passed through a band limiting low pass filter ,which has a cut-off frequency fc= 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'.The quantized PAM pulses are applied to an encoder which is basically an A To D converter. Each quantized level is converted into N bit digital word by the A to D converter. The value of N can be 8,16,32.64.

* 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 train of digital pulses .

*A pulses generator produces a train of rectangular pulses with each pulse of duration .The frequency of this signal is 'fs' Hz . This signal act 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 'fs' is adjusted to satisfy the Nyquist criteria.

Waveform



DECODER:

* A PCM signal contaminated with noise is available at the receiver output .

*The regeneration circuit at the receiver will separate the PCM pulses from noise and will reconstruct the original PCM 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 then passed through a serial to parallel converter .

* Output of this block is then applied to a decoder.

*The decoder is a D to A converter which performs exactly the opposite operation of the encoder. 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(t).

* The low pass filter is called as the reconstruction filter and its cut off frequency is equal to the messages bandwidth W.

Q4)

(a) Explain Foster seeley discrimination and compare the performance of foster seeley and ratio detector.



Ans: The phase discriminator of foster seeley discriminator is shown in the figure below.

*If you compare this circuit with the balanced slope detector circuit then you will find that the diode and load arrangement is same in both the circuits.

* But the method of applying the input voltage to the diodes which is proportional to the frequency deviation ,is entirely different . The foster seeley discriminator is thus derived from the balanced modulator.

* Here the primary and the secondary windings both are tuned to the same center frequency ''fc'' of the incoming signal.

*This simplifies the tuning process to a great extent and it will yield better linearity than the balanced slope detector.

Principle of Operation:

*Even though the primary and secondary tuned circuits are tuned to the same center frequency , the voltages applied to the two diodes D1 and D2 are not constant .

*They vary depending on the frequency of the input signal. This is due to the change in phase shift between the primary and secondary windings depending on the input frequency.

* Phase shift between the primary and secondary voltages is not constant but it depends on the input frequency . The effect of this is as explained below:



Output voltage of the discriminator : Output voltage of the phase discriminator is equal to the difference between the outputs of the two diode rectifiers.

therefore, VO = Va'b' =Va'o - Vb,o

As the diode drop are not known , we cannot calculate the output exactly .But it is sure that the output will be proportional to the voltage applied at the input of diode d1 and d2.

1) Output voltage at fin = fc :When the input frequency is equal to the center frequency (fc) , the phase shift between the primary and secondary voltages is exactly 900. Therfore the input voltages to both the diodes will be equal.

2) Output voltage for fin > fc: At input frequencies above the center frequency fc , secondary voltage Vab leads the primary voltage V1 by less than 900 .Hence input voltage to D1 is higher than input to D2 .The output voltage will therefore be positive for fin >f

3) Output voltage for fin < fc : For input frequencies below the center frequency fc , the secondary voltage Vab leads the primary voltage V1 by more than 900 .



Advantages:

1) It is more easy to align than the balanced slope detector as there are only two tuned circuits and both are to be tuned at the same frequency fc.

2) Linearity is better .This is because the operation of the circuit is dependent more on the primary to secondary phase relationship which is very much linear.

Disadvantage

1) It does not provide any amplitude limiting. So in the presence of noise or any other spurious amplitude variations, the demodulator output responds to them and produces errors.



COMPARE :

b) A sinusoidal carrier has an amplitude of 20 V and a frequency of 200 kHz .It is amplitude modulated by a sinusoidal voltage of amplitude 6v kHz. Modulated voltage is developed across a 80 Ω resistance.

(i) Write the equation of modulated wave.

(ii)Determine the modulation index

(iii)Draw the spectrum of modulated wave.

(iv)Calculate the total average power

Ans: Note : The sum below is a similar sum for your reference.



Q5)

a) Explain the generation and demodulation of SSBSC.

Ans:



( b) State and prove the following properties of Fourier Transform.

i)Time shifting ii)Differentiation in Time Domain

Ans:

i))Time shifting :



ii)Differentiation in Time Domain

Some processing techniques involve differentiation and integration of signal x(t) .The property is applicable if and only if the derivation of x(t) is Fourier transformable

Statement :

Let x(t)↔ X(t) and let the derivative of x(t) be Fourier transformable .Then

F

d/dt x(t) ↔ j2πf X(f)

Proof:

Meaning:

Differentiating the signal in time domain is equivalent to multiplying its Fourier transform by (j2πf). Thus differentiation will enhance the high frequency components since | j2πf X(f)| > |X(f)|.



Q6) Answer any four:

a) QPSK(Quadrature or Quaternary phase shift keying)

Ans: QPSK is an example of multilevel phase modulation. In this type of system , the data groups are divided into two or more bits and each group of bits is represented by a specific value of amplitude , frequency or phase the carrier.

* In QPSK system two successive bit in a bit stream are combined together to form a message is represented by a distinct value of phase shift of the carrier .

*The process of combining two successive bits is demonstrated;

* Easy symbol or message contains two bits .So the symbol duration Ts =2Tb.

*These symbols are transmitted by transmitted the same carrier at four different phase shifts

*Since these are four phase shifts involved , this system is called as quadrature PSK or 4-psk system.

Since there are four phase shifts involved ,this system is called as quadrature PSK or 4-PSK.If the symbol 00 is to be transmitted then we have to transmit a carrier is transmitted with a phase shift of 90o . Similarly the message 10 and 11 are transmitted the carrier at 1800 and 2700 respectively.



b) PCM - TDM

Ans When a large number of PCM signals are to be transmitted over a common channel, multiplexing of these PCM signals is required

* The figure shows the basic time division multiplexing scheme is called as the T1 digital system .

* This system is used to convey multiple signals over telephones line using wideband coaxial cable.



Bits/Frame

Synchronization between the transmitter and receiver commutators SW1 and SW2 is essential in order to ensure proper communication.

c) Friss Formula:

Ans: In practice the filters or amplifiers are not used in the isolated manner .They are used in the cascaded manner. The overall noise factor of such cascade connection can be determined as follows:



Conclusion:



D) Image frequency and rejection

Ans: In the broadcast AM receivers the local oscillator frequency is higher than the incoming signal frequency by intermediate frequency i.e.

f0 = fs + I.F.

*If two frequency components f1 and f2 are mixed in a mixer then mixer output consists of following components .If (f1 > f2)

output of mixer : f1, f2 (f1 - f2) and (f1 +f2)

* Out of which the difference component i.e. (f1 - f2) is selected using a tuned circuit after the mixer.



E) Sampling Theorem

Ans: In order to represent the original message signal "faithfully" , it is necessary to take as many samples of the original signal as possible

* Higher the number of samples , closer 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.Therfore 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 as follows:

STATEMENT



1) If a finite energy signal x(t) contains no frequencies higher than 'W' Hz , then it is completely determined by specifying its values at the instants of time which are spaced (1/2W) seconds apart.

2) If a finite energy siganl x(t) contains no frequency components higher than 'W' Hz than it may be completely recovered from its samples which are spaced (1/2W) seconds apart.

solution for principle of analog and digital communication · solution for principle of analog and...

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