Communication Theory

CHENNAI INSTITUTE OF TECHNOLOGY
Sarathy Nagar, Kundrathur, Pudupedu, Chennai 600 069. Chennai
Department of Electronics and Communication Engineering
Subject Name: Communication Theory UNIT I
Subject Code: EC 2252
AMPLITUDE MODULATION SYSTEMS
1. What is communication? Communication is the process of conveying or transferring messages from one Point to another. 2. What are the types of Communication system? Optical Communication systems Radio Communication systems Duplex Communication Systems Half duplex Communication systems Tactical Communication Systems 3. Define modulation? Modulation is a process by which some characteristics of high frequency carrier signal is varied in accordance with the instantaneous value of another signal called modulating signal. 4. What are the types of modulation? Amplitude modulation, Frequency modulation, Phase modulation. 5. Define depth of modulation(or)modulation index. It is defined as the ratio between message amplitude to that of carri amplitude. carrier ma=Vm/Vc 6. What are the degrees of modulation? Under modulation. m<1 Critical modulation m=1 Over modulation m>1 7. How will you determine the Fourier transform for periodic signal? [Nov-03]
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8. If a 10KW amplitude modulated transmitter is modulated Sinusoidally by 50%, what is the total RF power delivered? [Nov-05] ma=50/100=0.5;Pc=10 kw Pt=Pc(1+m2/2) =11.25kw 9. Define demodulation. Demodulation or detection is the process by which modulating voltage is recovered from the modulated signal. It is the reverse process of modulation. 10. Define Amplitude modulation? Amplitude modulation is the process by which amplitude of the carrier signal is varied in accordance with the instantaneous value of the modulating signal but frequency and phase of carrier wave is remains constant. 11. Define Frequency modulation? Frequency modulation is the process by which frequency of the carrier signal is varied in accordance with the instantaneous value of the modulating signal. 12. Define phase modulation? Phase modulation is the process by which Phase angle of the carrier signal is varied in accordance with the instantaneous value of the modulating signal. 13. As related to AM what is over modulation, under modulation and 100% modulation? [April-04] (OR) When does a carrier is said to be over, under modulated in Amplitude modulation? [Nov-04] In the case of Under modulation, modulation index ma<1(i.e.)Vmc.Here the envelope of Amplitude modulated signal does not reach the Zero amplitude axis. Hence the Message signal is fully preserved in the envelope of the AM Wave. In the case of Over modulation, modulation index ma>1(i.e.)Vm>Vc. Here the envelope of Amplitude modulated signal crosses the zero axis. In the case critical modulation modulation index ma=1(i.e.)Vm=Vc.Here the envelope of the modulated signal just reaches the zero amplitude axis. The message signal remains preserved. 14. Draw the phasor diagram of AM AM-SC signal. [Nov-04]
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15. Draw the Graphical and frequency spectrum and phasor Representation of AM with carrier.Frequency spectrum of AM with carrier:
Graphical representation:
Phasor representation of AM with carrier:
16. What is the power saving in DSB DSB-SC-AM and SSB-SC AM? Power saving in DSB-SC SC-AM is 66.7%. Power saving in SSB-SC SC-AM is 83.3%. 17. What are the advantages of VSB VSB-AM? It has bandwidth greater than SSB but less than DSB system. Power transmission greater than DSB but less than SSB system. No low frequency component lost. Hence it avoids phase distortion. 18. How will you generating DSB DSB-SC-AM? (OR) Give the two methods of generating DSB DSB-SC-AM. There are two ways of generating DSBSC DSBSC-AM such as 1. balanced modulator 2.ring modulators 19. Compare linear and non-linear modulators.
Sl.no Linear modulators 1 Heavy filtering is not required. CHENNAI INSTITUTE OF TECHNOLOGY HENNAI Non-Linear modulators Linear Heavy filtering is required Page 3

2 3
These modulators are used in modulation.
high level
The carrier voltage is very much greater than modulating signal voltage.
These modulators are used in low level Modulation. The modulating signal voltage is very much greater than the carrier signal voltage.
20. What are advantages of ring modulator? 1. Its output is stable. 2. It requires no external power source to activate the diodes. 3. Virtually no maintenance. 4. Long life. 21. Draw the Graphical and frequency spectrum and phasor Representation of DSB-SC-AM? Graphical representation:
Frequency spectrum:
Phasor representation
22. What are the advantages of DSB DSB-SC and SSB-SC. DSB-SC: Suppression of carrier results in economy of power. It is commonly used in carrier current telephony system, in which one sideband is filtered out to reduce the width of the channel required for transmission.<strong> It offers secrecy. It increases the efficiency because the carrier is suppressed. SSB-SC Bandwidth of SSB is half that of DSB SC AM. Thus twice the number of channels DSB-SC can be accommodated at a given frequency spectrum. No carrier is transmitted, hence possibility of interference with other channels are avoided. It eliminates the possibility of fading. Fading occurs due to multipath propagation of electro-magnetic waves. magnetic 23. Give the methods of generating SSB SSB-SC-AM. And mention Some applications of me SSB-SC The two methods of generating the SSB SSB-SC waves are
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Frequency discrimination or Filter method. Phase discrimination method. Applications: Police Wireless communication. SSB telegraph system Point to point radio telephone communication VHF and UHF communication systems. 24. Draw the Graphical and frequency spectrum and phasor Representation of SSB SSBSC-AM? Phasor representation:

Graphical representation: Frequency Spectrum representation: 25. What are the types of AM modulators? There are two types of AM modulators. They are Linear modulators Non-linear modulators Linear modulators are classified as follows Transistor modulator There are three types of transistor modulator. Collector modulator Emitter modulator Base modulator Switching modulators Non-linear modulators are classified as follows linear Square law modulator Product modulator Balanced modulator 26. What is single tone and multi tone modulation? If modulation is performed for a message signal with more than one frequency component then the modulation is called multi tone modulation. If modulation is performed for a message signal with one frequency component ation then the modulation is called single tone modulation. 27. What is the need for modulation? Needs for modulation: Ease of transmission Multiplexing Reduced noise

Narrow bandwidth Frequency assignment Reduce the equipments limitations
28. Compare AM with DSB-SC and SSB-SC.

AM signal Bandwidth=2fm Contains USB, LSB, carrier More power is required for transmission DSB-SC Bandwidth=2fm Contains USB,LSB Power required is less than that of AM. SSB-SC Bandwidth=fm Contains LSB or USB Power required is less than AM &DSB-SC
29. What are the types of AM detectors? 1. Nonlinear detectors 2. Linear detectors 30. What are the types of linear detectors? 1. Synchronous or coherent detector. 2 .Envelope or non coherent detector. 31. A transmitter supplies 8 Kw to the antenna when modulated. Determine the total power radiated when modulated to 30%. ma=30/100=0.3;Pc=8 kw Pt=Pc(1+m2/2) =8.36 kw 32. The antenna current of an AM transmitter is 8A when only carrier i sent. It is increases to 8.93A when the carrier is modulated by a single sine wave. Find the percentage modulation. Solution: Given: Ic =8A It=8.93A m=0.8 2 Formula: It=Ic (1+m /2) 8.93=8(1+m2/2) m=0.701 It=8 (1+0.82/2) It=9.1A 33. A 1MHz carrier is amplitude modulated by 400Hz modulating signal to a depth of 50%.The modulated carrier power is 1KW.Calculate the power of the 0%.The unmodulated signal. Solution:CHENNAI INSTITUTE OF TECHNOLOGY HENNAI Page 6

Pc=1KW, ma=0.5=50% The increase in power is given by 1.125 1=0.125KW is contained in two side bands. 1.125-1=0.125KW 34. What do you mean by Hibert transform and inverse Hilbert Transform? And write few applications of Hilbert transform? It may be observed that the function xh(t) obtained by providing(-/2)phase shift to every frequency component present in x(t),actually represents the Hilbert transform of x(t).This means that xh(t) is the Hilbert transform of x(t) defined as Also, the inverse Hilbert transform is defined as

Few applications of Hilbert transform. For generation of SSB signals, For designing of minimum phase type filters, For representation of band pass signals.
35. Define multiplexing. Multiplexing is defined as the process of transmitting several message signals simultaneously over a single channel. 36.Define sensitivity,Selectivity,Stability. Sensitivity is defined as a measure of its ability to receive weak signals. measure Selectivity of a receiver is defined as its ability to select the desired signals among the various signals.and Stability is the ability of the receiver to deliver a constant amount of output for a given a given period of time. riod 37. What are the drawbacks of emitter modulator? 1. The amplifier is operated in class A mode, thus the efficiency is low. 2. The output power is very small. Thus it is not suitable for generating high level modulation.
38. Define super heterodyne principle. It can be defined as the process of operation of modulated waves to obtain modulated similarly modulated waves of different frequency. This process uses a locally generated carrier wave, which determines the change of frequency. And also we can define that a device performs the frequency translation of a modulated signal is known as a frequency mixer. the operation is often called gnal frequency mixing, frequency conversion, or heterodyning.

16 mark ANSWERS: 1. Write the Expression for AM & calculate its Power and Efficiency (OR) Write down the AM equation and also obtain an expression for its Power saving and efficiency. Let us consider the modulating voltage, Vm(t) = Vm Cos mt Vc(t) = Vc Cos ct According to the definition, the amplitude of the carrier signal is changed after modulation. VAM = Vc + Vm=Vc+Vm Cos mt------------- (1) = Vc [ 1+ (Vm/Vc) Cos mt ] ------------- (2) ma= Em/Ec Where ma is the modulation index (or)depth of modulation. Therfore VAM=Vc(1+ ma Cos mt ) ------------- (3) But the instantaneous amplitude of modulated signal, i.e at any time VAM(t)= VAM Cos ct------------- (4) ------------i.e. the modulated wave contains amplitude of the carrier wave.so that equation(4)is represented by the product of VAM Cos ct. Sub equation(3)in(4) VAM(t)= Vc (1 + ma Cosmt) Cosct = Vc Cosct + ma Vc Cosmt Cosct = Vc Cosct + ma Vc [Cos(c + m)t + Cos(c m)t] 2 = Carrier + USB + LSB Finally the modulated wave contains carrier term and upper side band and lower side band. Frequency spectrum of AM Wave Graphical representation of AM Phasor representation of AM Power relation in AM:

The modulated wave contains three terms such as carrier wave, LSB, USB. The modulated wave contains more power than the unmodulated carrier. Total Power in modulated wave will be, Pt = PC + PUSB + PLSB i.e. total power Pt of AM wave is the sum of carrier power and side band power.

PC - Carrier power PUSB - Upper Side Band power PLSB - Lower Side Band power But we know that Vcarrier=RMS value of carrie voltage. carrier VLSB= VUSB=RMS value of upper and lower side band voltages. R=Resistence in which power is dissipated. The RMS value is VRMS=Vm/ 2 Therefore Pcarrier=V2carrier/R=(Vc/ 2)2/R=Vc2/2R Current relation in AM: It=Total(or)modulated current IC=Carrier current Efficiency: % = Power in side band Total Power =PLSB+PLSB x 100 PTotal = ma2 x 100 2+ma 2 If ma = 1 Then % = 33.33 % In this only one third of total power is carried by the sidebands and the rest two third is sidebands wasted.
x 100

2. Derive the expression for DSB DSB-SC AM and calculate its efficiency. In AM with carrier scheme, there is wastage in both transmitted power and bandwidth. In order to save the power in amplitude modulation the carrier is suppressed modulation because it does not contain any useful information. This scheme is called as the double side band suppressed carrier amplitude Modulation (DSB SC). It contains LSB and USB (DSB-SC). terms, resulting in a transmission bandwidth that is twice the bandwidth of the message signal. Let us consider the modulating voltage, Vm(t) = Vm Cos mt -----------------------(1) Vc(t) = Vc Cos ct -------------------------(2) For obtaining the DSB-SC wave we have to multiply both carrier signal and message SC signal.hence, V(t)DSB-SC = Vm(t).Vc(t) =Vm.Vc cosm t cosc t V(t)DSB-SC=Vm.Vc [ cos(c +m )t + cos(c - m )t 2 USB LSB From this equation we know that the carrier is suppressed in double side band suppressed carrier. Graphical Representation of DSB DSB-SC AM: Frequency spectrum of DSB-SC AM SC It shows that carrier term c is suppressed. It contains only two sideband terms having frequency (c - m) and (c + m). Hence this scheme is called as DSB-SC DSB AM. Phasor representation of DSB- AM: -SC
Power Calculation: w.k.t The total power transmitted in AM is tal Pt =Pcarrier +PLSB +PUSB If the carrier is suppressed , then the total power transmitted is Pt = PLSB +PUSB

w.k.t, PLSB = PUSB = ma 2 Vc2 8R Pt = ma 2 Vc2 + ma 2 Vc2 8R = Power Saving = Pt - Pt x 100 Pt = [ 1+ ma 2 / 2] PC - [ma 2 / 2] PC 8R
[1+ ma 2 / 2] PC = PC [1+ ma 2 / 2] PC % Power Saving = 2 x 100 2+ ma 2 (2/3) x 100 = 66.7 %
If ma = 1, then power saving
i.e 66.7% of power is saved by using DSB DSB-SC-AM. 1. 3. Derive the expression for SSB SSB-SC AM and calculate its Efficiency In AM with carrier both the transmitting power and bandwidth is wasted. Hence the DSB-SC AM scheme has been introduced in which power is saved by SC suppressing the carrier component but the bandwidth remains same. Increase in the saving of power is possible by eliminating one sideband in addition to the carrier component because the USB and LSB are uniquely related by symmetry about the carrier frequency. So either one sideband is enough for transmitting as well as recovering the useful message. The block diagram of SSB SSBSC AM is shown in figure.
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As for as transmission information is concerned only one side band is necessary. concerned so if the carrier and one of the two sidebands are suppressed at the transmitter, no information is lost. This type of modulation is called as single side band suppressed carrier carrier-AM and the SSB system reduces the b band width by half. The block diagram of SSB SSB-SC AM is shown in figure.
The single side band suppressed carrier can be obtained as follows, In order to suppress one of the sideband, the input signal fed to the modulator1 is 90 out of phase with that of the signal fed to the modulator2.
Let V1(t) = Vm cosm t .Vc cos c t V2 (t) = Vm cos(mt + 90 ). Vc cos (c t + 90 ) = Vm sin mt. Vc sin c t Therefore, V(t) SSB = V1(t) +V2(t) = Vm Vc [cosm t. cosc t + sin mt. sinct] V(t)SSB= Vm Vc cos ( c - m) t. ( W.k.t, for DSB-SC AM VDSB-SC (t) = Vm Ec [cos(c +m )t + cos(c - m)t] 2 When comparing equations of V(t) SSB-SC and VDSB-SC (t) , one of the sideband is suppressed. Hence this scheme is known as SSB SSB-SC AM. Frequency Spectrum Of SSB SSB-SC-AM:
The Frequency spectrum shows that only one side band signal is present , the carrier and the other sideband signal are suppressed. Thus the bandwidth required reduces from 2 m to m i.e., bandwidth requirement is reduced to half compared to AM & DSB-SC signals. SC
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Graphical representation of SSB SSB-SC-AM: Phasor representation of SSB sentation SSB-SC-AM: Power calculation: Power in sidebands Pt = PSB = ma2Vc2/8R=ma2 Pc / 4 Power saving with respect to AM with carrier Power saving = Pt - Ptx100 Pt where Pt = Total power transmitted. Power saving = [ 1+ ma 2 / 2] PC - [ma 2 / 4] PC
[1+ ma 2 / 2] PC = PC +[ma 2 / 2] PC - [ma 2 / 4] PC [1+ ma 2 / 2] PC = If ma=1 then %power saving= (5/6) x100=83.3% So we can save 83.3% of power with respect to AM with carrier. We can also find the power in SSB SSB-SC-AM with respect to AM with DSB-SC AM DSB system and that will be, If ma=1 then %power saving=50%.therefore 50%power is saved with respect to DSB DSB-SC System. 4. Write notes on VSB system. (OR) Explain vestigial side band modulation with frequency response Characteristics. [Nov-03] If we are considering the modulated signals of very large bandwidth (such as video signals, T.V and high speed data signals) having the very low frequency components along with rest of the signal. These low frequency components give rise to sidebands very close to the carrier frequency which are difficult to remove it. so it is not possible to go till the equency extreme and fully suppress one complete sideband in case of television signals.

The low video frequencies contain the most important information of the picture, so if we suppress one sideband it becomes unsuitable for handling such ress type of signals. This difficulty has been over come in a scheme known as vestigial sideband (VSB) modulation.i.e. Partially suppressing any one of the sidebands. A VSB system is a compromise b between DSB-SC-AM and SSB-SC-AM, but AM SSB avoids their disadvantages.VSB signals are very easy to generate and the bandwidth is slightly greater than SSB SSB-SC-AM, but less than DSB-SC-AM. VSB modulation is derived by filtering DSB DSB-SC-AM. Here in VSB system one AM. side band is passed almost completely, while a part of other sideband is used. de So mainly in VSB instead of rejecting one sideband completely a gradual cut of one sideband is acceptable. The role off characteristics of filter is such that partial suppression of transmission sideband is VSB. By using VSB system the bandwidth obtained is greater than SSB system but less than DSB. Power transmission is greater than DSB but less than SSB system. No low frequency component lost. Hence avoids the phase distortion. 5. Write short notes on the types of generation methods of AM with Carrier. The generation method of AM waves are broadly divided in to two types, Linear modulator(or)large signal modulator(or)high level modulation Non Linear modulator(or)small signal modulator(or)low level modulation modulator(or)low Linear modulator: In this type of modulators the devices are operated in linear region of its transfer characteristics. Linear modulators are also divided in to two types, Transistor modulator Switching modulator. Non linear modulators: These modulators are operated in nonlinear region. These are used in low level modulation. The types of non linear modulators are, Square law modulator, Product modulator, Balanced modulator.
6. Explain the principle of operation of square law modulator and Mention its merits and demerits. (OR) Explain any one generation method of AM [Nov-05] (OR) Explain the principle of operation of any one non linear Modulators.

Here we are using V-I characteristics of diode is used as a element for non I non-linear modulators. This is suited at low voltage levels because of the fact that current current-voltage characteristics of a diode is highly nonlinear particularly in the low voltage region particularly as shown in figure. A square law modulator has three features shown in figure. Summer-to sum carrier and modulating signal to A non linear element Band pass filter for extracting desired modulating products.
Principle of operation: From the figure the non linear device diode is used to produce low level amplitude modulation. Here the carrier and modulating signals are applied across the diode. A dc battery Vcc is connected across the diode to get a fixed operating point on the V-I characteristics of the diode. I This amplitude modulation that is low level can be explained by considering the fact when two different frequencies are passed through the non linear device. So when we apply carrier and modulating frequencies different fr frequency terms appear at the output of diode. These different frequency terms are applied across the tuned circuit which is tuned to carrier frequency and has a narrow bandwidth just to pass two sidebands along with the carrier and reject other frequencies. So the output of the tuned circuit will be carrier and two sidebands that is amplitude modulated wave is produced. Mathematical analysis: Let the modulating voltage will be Vm= Vm cosm t------------- (1) Let the carrier voltage will be Vc= Vc cosc t---------------- (2) So the input voltage applied to t6he diode will be V1= Vm+ Vc ------------------------------ (3) = Vm cosm t+ Vc cosc t W.k.t the current and voltage relationship of a linear circuit is, i=av i=current through linear device v=voltage through the linear device a=proportionality constant

The current and voltage relationship of a nonlinear circuit may be expressed as, i=a V1+b V12+c V13+ --------- This means that due to non non-linearity in the V-I characteristics of a non-linear I non circuits, the current becomes proportional not only to voltage but also to the square, cube and higher powers of voltage. So i= a1V1+a2 V12+a3 V13+ ---------- ---------- (4) Where a1, a2, a3 are proportionality constants and V1is the input voltage applied to the diode or FET. Sub equn(3)in(4)then i= a1(Vm cosm t+ Vc cosc t)+ a2(Vm cosm t+ Vc cosc t)2+-----------
= a1 Vm cosm t+ a1 Vc cosc t+ a2 Vm2 cosm2 t+ a2 Vc2 cosc2 t+2a2VmVccosm t cosc t+--Neglecting second and higher order terms, we get i(t)= a1 Vm cosm t+ a1 Vc cosc t +2a2VmVc cosm t cosc t+--= a1 Vm cosm t+ a1 Vc cosc t +2a2VmVc [cos(c+m)t + cos(c-m)t] 2 i(t)= a1 Vm cosm t+ a1 Vc cosc t +a2VmVc [cos(c+m)t + cos(c-m)t]
The tuned circuit is tuned to the carrier frequency and it allows only c, c+m, c-m terms and eliminates all other terms.hence we obtain
i(t)= a1 Vc cosc t +a2VmVc [cos( c+m)t + cos(c-m)t] [cos( The main drawback of using diode modulator is it does not provide amplification and a single diode is unable to balance out the frequency completely. These to limitations can be eliminated by using amplifying devices like transistor, FET in a balanced mode. We can also use the square law modulator with FET instead of diode. From the final equation we know that it consists only the carrier and USB and consists LSB frequency components and all the components are removed.
7. Explain the principle of operation of Product modulator. (OR) Explain any one generation method of AM [Nov-05] (OR) Explain the principle of operation of any one non linear Modulators.

In a square law modulator the sidebands of an AM signal contains new frequencies that are not present in the carrier (or) mes message signal. The modulating signal and carrier signal is multiplied and added with carrier frequency in order to obtain the amplitude modulated wave. This is practically applied by using analog multiplier and an op op-amp summer. analog multiplication is done by variable trasconductance method. Here input voltage is applied to variable Tran conductance multiplier. V1 varies with total emitter current. V2 is given to the input to VtoI converter. The output of the multiplier contains only sideband terms which is given to the summer with the carrier which is given to the summer with the carrier which in turn produces an amplitude modulated wave. The output V0(t)=KV1V2 V1=Vm cosm t, V2=Vc cosc t V0(t)=K Vm Vc cosm t cosc t os Then input of summer will be Vs(t)= V0(t)+ Vc(t) Vs(t)= Vc cosc t+ K Vm Vc cosm t cosc t Thus the AM signal with carrier is produces by using product modulator.
8. Explain the principle of operation of balanced modulator for Generating AM with carrier. [Nov03] (OR) Explain any one generation method of AM [Nov-05][Apr-04] [Nov (OR) Explain the principle of operation of any one non linear Modulators. We know that a non-linear device may be used to produce amplitude modulation. linear That is one carrier and two side bands. Here we are using two transistors, non linear devices are identical and the circuit is symmetrical. In this the carrier voltage across the two windings of a centre tap transformers are equal and opposite in phase Vc= -Vc Input to the transistor T1 is given by, Vbe = Vc(t)+ Vm(t)----------(1) Vbe= Vc cosc t +Vm cosm t--------- (2)
Similarly Input to the transistor T2, Vbe = -Vc(t)+ Vm(t)----------(3) Vbe = - Vc cosc t +Vm cosm t--------- (4)
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By using the non-linearity property the collector current can be written as per square law linearity equation ic = a1 Vbe + a2 Vbe2 + . -----------(5) ic = a1 Vbe + a2 Vbe2 +. ---------(6) Hence from this ic = a1 [Vc cosc t +Vm cosm t]+ a2 [Vc cosc t +Vm cosm t]2 ic= a1 [Vc cosc t +Vm cosm t]+ a2V2c cos2c t+a2V2m cos2m t+2a2Vm Vccosm t cosc t---(7) t Similarly ic = a1 [-Vc cosc t +Vm cosm t]+ a2 [-Vc cosc t +Vm cosm t]2 ic= a1 [-Vc cosc t +Vm cosm t]+ a2V2c cos2c t+a2V2m cos2m t-2a2Vm Vccosm t cosc t---(8) t The output AM voltage V0 is K[ ic ic ] That is Vo = K[ ic ic ] Because ic, ic flows in the opposite direction.andKis a constant depending on impedance or other circuit parameters. Vo = 2K a1 Vc Cos ct + 4a2 Vm Vc Cosmt Cosct Cos This equation represents the output modulated voltage which contains carrier and side band terms. The same circuit can also be used to generate the DSB DSB-SC-AM. By using balanced AM. modulator the undesired harmonics are automatically balanced out.
9. Explain the principle of any one of the linear modulators? (OR) What are the types of transistor modulators and explain the Working principle of collector modulator? (OR) Explain any one generation method of AM [Nov-05][Apr-04] [Nov The linear modulators are used in high level amplitude modulation. There are types of linear modulators available, They are 1. Transistor modulators 2. Switching modulators. Transistor modulator having the classification as collector modulator, Base modulation, Emitter modulation.

Collector modulator Collector modulation method is a very popular method for high level AM generation and it is used in high power applications. Transistor T1, makes a radio frequency classC amplifier. and at the ba base T1,the carrier signal is applied. VCC makes the collector supply used for biasing purpose. Transistor T2 makes a class B amplifier which is used to amplify the modulating class-B signal. So class-C is used to amplify the carrier signal, and class B is used to amplify the C class-B a modulating signal. This amplified base band or modulating signal appears in series with the collector supply. The function of the capacitor is to prevent the carrier signal flowing through the modulation transformer. And the supply VCC is used for biasing. Operating principle In class-C amplifier the magnitude of the output voltage is a definite fraction of or C at the most equal to supply voltage VCC. So the unmodulated carrier is amplified by Class C modulated amplifier using Class-C transistor T1 and its magnitude will remain constant at VCC since there appears no ts voltage across the modulating transformer in the absence of base band or modulating voltage. But now if a base band or modulating voltage Vm cosm t appears across the modulating transformer, the signal will be added to the supply voltage VCC. This type of slow variation in carrier supply voltage at the output of the modulated class-C amplifier. So the final output of the carrier modulation transformer will be variation of voltage with respect to modulating voltage with supply voltage VCC. From the figure we observed that an envelope of the output voltage is identical with the base band or modulating voltage and hence an AM signal is generated. Mathematical analysis: When the modulating voltage is applied then it is added up with the supply voltage then, the amplitude of modulated voltage will be This amplitude of modulated voltage with VCC produces a change in carrier supply voltage. So change in carrier supply voltage will be equal to amplitude of modulated voltage. so we can mention the change in carrier supply voltage is VC then, Finally the amplitude modulated output voltage will be obtained from carrier modulating transformer. So final output V0 will be Hence from the above equation we can express the output voltage consists of a carrier term and the side band terms. Hence an AM wave generated.

Like collector modulator in base and emitter modulator the modulating signal is applied to base and emitter junction, and added up with the supply voltage and produces the change in carrier. Finally we will obtain the AM wave. Advantages: Power output per transistor is usually high. Collector efficiency is high. Disadvantages: Large modulating input power is required when the modulating amplifier is a high power amplifier.
10. Explain the principle of any one of the linear modulators? (OR) Explain the working principle of switching modulators? The switching modulators (or) chopper modulators are the linear modulators. Here we are using active device as switches. The devices such as BJT or diode serves as a switch. Here we are using a simple diode circuit. The diode is forward biased for every positive half cycle of the carrier and b behaves like a short circuited switch. For a negative half cycle of the carrier the diode is reverse biased and behaves like an open switch. The signal doesnt reach the filter and no output is obtained. While the circuit is short circuited without applying any modulating signal, then the carrier is carried over to the tank circuit, then the output will be tuned to only carrier frequency. V0(t)=VCcosct---------(1)[Steady state o/p voltage in the absence of Modulating (1)[Steady voltage] Now we are applying the modulating signal with 1: N ratio. Then this signal will modulating be added up with the o/p voltage. then the o/p voltage will be, V0(t)=Vccosct+N.VmVccosmt cosct This means that the o/p voltage contains carrier and side band terms. Thus the amplitude contains modulated wave generated. The tank circuit is used to select certain frequencies and rejects all other frequencies. If the number of turns of transformer is selected properly then the desired modulation has been accomplished without appreciable generation of has undesired components otherwise band pass filter is required to remove the unwanted components. 11. Explain the principle of operation of balanced modulator for Generating DSBDSB SC-AM. [Nov03] (OR) Explain any one modulation method for generating DSB DSB-SC-AM.

We know that a non-linear device may be used to produce amplitude modulation. linear That is one carrier and two side bands. But a DSB-SC signal contains only two side bands. So we are connecting two diodes or two transistors to produce a DSB DSB-SC signal. Here the two non-linear devices are connected in balanced mode. linear The same circuit can be used to generate AM with carrier. The main difference between AM with carrier generation and DSB en DSB-SC AM is the feeding points of AM the carrier and modulating signals are interchanged.
Principle of operation: The modulating signal is applied as the input to the transistor T1 and T2 where as we are applying carrier in case of AM with carrier. Because of centre tap transformers are equal and opposite in phase Vm= -Vm Input to the transistor T1 is given by, Vbe = Vm(t)+ Vc(t)----------(1) Vbe= Vm cosm t +Vc cosc t --------- (2) Similarly Input to the transistor T2, Vbe = -Vm(t)+ Vc(t)----------(3) Vbe = - Vm cosm t +Vc cosc t --------- (4) By using the non-linearity property the collector current can be written as per square law linearity equation
ic = a1 Vbe + a2 Vbe2 + . -----------(5) ic = a1 Vbe + a2 Vbe2 +. ---------(6) Hence from this ic = a1 [Vm cosm t +Vc cosc t ]+ a2 [Vm cosm t +Vc cosc t]2 ic= a1 [Vm cosm t +Vc cosc t ]+ a2V2m cos2m t+a2V2c cos2c t+2a2Vm Vccosm t cosc t---(7) t Similarly ic = a1 [-Vm cosm t +Vc cosc t ]+ a2 [-Vm cosm t +Vc cosc t]2 ic= a1 [-Vm cosm t +Vc cosc t]+ a2V2m cos2m t+a2V2c cos2c t-2a2Vm Vccosm t cosc t---(8) t The output AM voltage V0 is K[ ic ic ] That is Vo = K[ ic ic ]

Because ic, ic flows in the opposite direction.andKis a constant depending on impedance or other circuit parameters. Vo = K [2a1 Vm Cos mt + 4a2 Vm Vc Cosmt Cosct] Cos Here we are using a band pass filter the other frequencies centered at c so it will allow only pass band frequencies. So only the side bands are present in the o/p.Hence the DSB DSB-SC-AM generated. AM The main advantage of this modulator is saving power and efficiency because of the suppressing of the carrier.
12. Explain the principle of operation of Diode balanced modulator (Or) ring modulator for generating DSB DSB-SC-AM. (OR) Explain any one modulation method for generating DSB DSB-SC-AM. Ring modulator is another product modulator. Which is used to generate DSB DSB-SC Signal. Here we are not using any band pass filter at the output; the harmonic frequencies are automatically controlled. In a ring modulator circuit four diode are connected in the form of ring in which ng all the four diodes are point in the same manner. All the four diodes in the ring are controlled by a square wave carrier signal Vc (t). On positive half cycle of the carrier diode D1 and D3 will conduct. And pointa is connected to the input through half of the output transformer secondary. In negative half cycle diode D2 and D4 will conduct. While we applying both modulating and carrier signal on +ve half cycle diode D1 and D2 will conduct. And in ve half cycle diode D3 and D4 will conduct. So the final o/p will be product of signals.
Let us consider the modulating voltage, Vm(t) = Vm Cos mt -----------------------(1) Vc(t) = Vc Cos ct -------------------------(2) For obtaining the DSB-SC wave we have to multiply both carrier signal and message SC signal.hence, V0(t) = Vm(t).Vc(t) =Vm.Vc cosm t cosc t V0(t) =Vm.Vc [ cos( c +m )t + cos(c - m )t cos( 2 USB LSB The o/p contains only two side band terms. Hence an DSB SC wave is generated. DSB-SC

The main advantage is the o/p is stable and long life, no external source to activate the diodes.
13. Explain the frequency discrimination method for generating SSB SC-AM. SSB(OR) Explain the method of generating an SSB.[Nov SSB.[Nov-05][Apr-04]
Frequency discrimination method is otherwise called as selective filtering method. The SSB is generated by filtering the DSB. Here the base band is restricted and approximately related to the carrier frequency. Under this condition the desired side band is selected by an appropriate fi filter by restricting the desired side band. But practically it is not possible to extract completely one of the side band and rejects other, because the practical filter doesnt have ideal characteristics. Hence a compromise has been made to design an appropriate side band filter. appropriate This method is basically consists of a balanced modulator and a suppression filter. The DSB-SC signal is passed through a sharp cut off filter to eliminate the SC cut-off undesired side band. The requirements of a filter are The pass band of the filter should be same as that of the desired side band. The separation region between pass band and stop band should not exceed twice the max frequency component present in the base band. Here we are using band pass filter. Transmission of very high (such as 10MHz) (such frequency as well as very low frequency (50Hz) through side band filtering is impossible. So we are using Mechanical filters or crystal filters. The filter is not able to follow such a change of steep response. To overcome this they have been replaced by mechanical or crystal filters, which may be used in frequency range of 500 KHz to 80 MHz. We are using mechanical filters because they are small in size, good band pass characteristics, very good attenuation. The crystal filters may be cheaper but are preferable only at frequencies greater cheaper than 1MHz. All the filters having the disadvantage, the maximum operating frequencies are below the usual transmitting frequency. Hence a balanced mixer is used. So we are using mixer in order to increase the operating frequency, for that we are increase adding frequency of the crystal oscillator. Thus the frequency will below the transmitted frequency. So that we are using mixer. If suppose the transmitting frequency is two high then two stages of mixing will be required.
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At the mixer part we are using classBamplifier which is more efficient than ClassAoutput.
14. Explain the Phase shift method for generating SSB SSB-SC-AM. (OR) Explain the method of generating an SSB. f [Nov-05][Apr-04] [Nov
This method avoids the prime disadvantage of filtering method. That is requirement of a sideband filter with a narrow transition band and it cannot be used for very low and very high frequencies. The undesired components are removed by generating two side band components ponents out of phase, when added these side band components cancel each other leaving only the desired side band signal. From the figure One of the modulators M1 receives the carrier voltage shifted by 900 and the modulating voltage, where as another balanced modulator M2 receives the modulating voltage shifted by 900 and the carrier voltage. Both balanced modulators produce an output considering only of side bands. Both the upper side bands leads the carri voltage by 900.one of the lower side carrier bands leads the reference voltage by 900 and the other lags it by 900. The two lower side bands are thus out of phase, and when combined together in the adder they cancel each other, and hence they produced the SSB s signal.
Mathematical analysis: For modulator1 (M1) Vc(t) = Vc cos(c t+900) Vm(t)= Vm cosm t For modulator2(M2) Vc(t) = Vc cosc t Vm(t)= Vm cos(m t+ 900) o/p for modulator(M1)isV1 V1=Vc cos(c t+900). Vm cosm t = VcVm [ cos(c t+900+ m t)+ cos(c t+900- m t)] --------(1) 2 o/p for modulator(M2)isV2 V2=Vc cosc t. Vm cos(m t+ 900) = VcVm [ cos(c t+ m t +900)+ cos(c t- m t -900)] --------(2) 2 Final o/p V0= V1 + V2

V0=2VcVm cos(c t+ m t +900)----------(3) Equation (1) contains 900phase shift and equn (2) contains ( 0) phase shift. Hence they (-90 both get cancelled. Thus one of the side band is cancelled where as the other is kept as it is. This cancelled method avoids the use of filters. 15. Explain the Modified Phase shift method for generating SSB SSB-SCAM. (OR) Explain the weavers method of generating an SSB. The modified phase shift method overcomes the limitation of phase shift method. That is AF phase shift network is required to operate over a large range of audio frequencies but also retains the advantage like its ability to generate SSB at any frequency and use of low audio frequency. This method provides both RF and AF oscillator phase shift and also used in low frequency and so it can be used for both audio and radio frequencies. In weavers method we are using four balanced modulators. it is just similar to that just phase shift method. Balanced modulator 1&2 are slightly different from modulator 3&4. Here instead of trying to phase shift the whole range of audio frequencies it combines the AF carrier which lies in the middle of audio frequency. So we are taking the modulating signal Vm= Vm cosm t-------- (1) The A.F carrier(sub carrier signal) V0(t) is the one type of carrier,then Middle of frequency V0(t)= V0 cos0 t 2 So V0(t)= 2V0 cos0 t The R.F carrier Vc(t)= Vc cosc t---------------------(2) 2 So Vc(t)= 2Vc cosc t-----------------------(3) For modulator1 (M1) Vm(t)= Vm cosm t Vc(t) = 2V0 cos(0 t+900) o/p for modulator(M1)is A A= Vm cosm t .2V0 cos(0 t+900) A= VmV0 [ cos(0 t+900+ m t)+ cos(0 t+900- m t)] --------(4) For i/p to modulator2(M2) Vm(t) = Vm cosm t Vc(t)= 2V0 cos0 t o/p for modulator(M2)is B B=2Vm V0 cosm t cos0 t

B= Vm V0[cos(0 t+ m t)+ cos(0 t- m t)]-----------(5) The o/p of LPF1, 2 will be any one of the side band terms. Here the USB is suppressed O/p of LPF1 is C= Vm V0 cos(0 t+900- m t )-----------(6) O/p of LPF2 is D= Vm V0 cos(0 t- m t)-----------(7) The o/p of LPF1,2 are given to B.M3,B.M4. For i/p to modulator3 (C)will be C= Vm V0 cos(0 t+900- m t )] Vc(t) = 2Vc cosc t o/p for modulator(M3)is V1 So V1= 2Vm V0 Vc cos(0 t+900- m t ) cosc t Assume Vm =V0 =Vc=1,then V1= 2cos(0 t+900- m t ) cosc t--------------(8) For i/p to modulator4 (D)will be D= Vm V0 cos(0 t- m t) Vc(t) = 2Vc cos(c t+900) o/p for modulator(M4)is V2 So V2= 2Vm V0 Vc cos(0 t- m t) cos(c t+900) Assume Vm =V0 =Vc=1, then V2= 2 cos(0 t- m t) cos(c t+900)------------------(9) Final o/p VSSB-SC= V1 + V2 SC Hence VSSB-SC=2 cos[(0 + c m)t+900] VSSB-SC=-sin[(0 + c m)t. hence the SSB signal generated. 16. Explain the Principle of operation of AM transmitters, SSB
Antenna
Transmitters and ISB transmitters. AM Transmitters: Block diagram:v AM broad cast transmitters are used in large number of applications such as transmission of music, speech or entertainment for general public. The output power of transmitter varies from 1kw to 100kw. It also works in long ware medium wave and m short wave frequency ranges. v Master oscillator: it generates high frequency waves which are subsequently sued as the carries. The master oscillator should fulfill the following requirements. The circuit must provide carrier of specified frequency Frequency should be drift free. (Slow variations of frequency with time is termed as frequency drift .i.e. it should be easily adjustable. Changes in supply voltages. v Buffer amplifier: the buffer s : stage is a class A voltage amplifier.

v If the buffer amplifier is absent then master oscillator will be directly connected to the power amplifier, hence there may be a chance of change in frequency v Harmonic generators (frequency multiplier): when a transmitter has to be operated nsmitter at the high frequency band (3MHZ 30MHZ) then frequency multipliers are used to obtain (3MHZ-30MHZ) the required output frequency. v Power amplifier: the RF voltage generated by the master oscillator has usually very : small power. This power level is raised to required level by a chain amplifier. v Preamplifier: For noiseless transmission audio amplifier used. For T.V transmission video amplifier is used. The main function of pre amplifier is to raise the amplitude of the signal source to a desired level v Modulated amplifiers: This is generally a push pull class B or class C amplifier and the modulating signal is fed into plate or collector of the amplifier. Generally collector or plate modulator method is used in a high power radio broa broadcast and radio telephone transmitters. v Antenna matching network: The output of the final amplifier is fed to an impedance matching network in order to avoid the amplitude and frequency distortion while passing the side bands. v In case of AM transmitters we are having two types of modulation low level and high level modulation. v In case of AM transmitters, the modulating signal is given to power amplifiers which are modulated and carrier frequency is also given to transmit the signals. Fi Finally the modulated signal is amplified by modulated amplifier and given to antenna. v We have already discussed about the type of AM modulation, so from this we know that we are having low level AM transmitter, high level AM transmitter. SSB transmitters: v The single side band transmission method consists of transmitting only one side band with the suppression of the other side band along with carrier. v It is use in radio telephony systems v This method having high signal to noise ratio compare t DSB. to v Compare to DSB, Bandwidth are also reduced to half. Thus twice the number of channels can be accommodated in a given frequency spec ham v But the main disadvantage of SSB system is designing of SSB transmitter and receiver is more complex v Before taking the SSB transmission, we know the generation of SSB wave by ore frequency discrimination method and phase discrimination method. In frequency discrimination method we are using filters in order to suppress the side band and balanced mixer in order to increase the operating frequency. o v In phasediscrimination method the output of balanced modulators are added together and produced the SSB SC waves.

v In the block diagram of SSB transmitter, the audio frequency and as well as carrier frequency are fed in to the balanced modulator. The output will be the DSB output i.e. quency one carrier is suppressed and two sidebands are present. v The output of the balanced modulator is given to filter in which anyone of the sideband is suppressed either upper side band or lower sideband. er v This output fed in to a balanced mixer with higher frequency in order to increase the operating frequency. From this the harmonics are rejected by band filters. v The output is given to second mixer mixes the signal frequencies with the high frequencies frequency and again filtered and finally the signal is amplified in the final amplifier and transmitted. ISB Transmitter (independent side band) v The ISB is the further extension of SSB principle. I.e. independent side band working with each side band of the transmitted signal carrier a separate message signal. v The carrier transmission is at a reduced level to have more power. So the channel capacity will be wide one. v In the block diagram of ISB transmitter, the audio signals from two channels A &B are from amplified by the channel amplifier and applied to the channel balanced modulators I & II then a 100KHZ carrier signal is generated and applied to the balanced modulators. v The resulting signal is fed to balanced modulator III along with 100KHZ carrier of low along magnitude. v Then the lower side band is filtered by filter and USB is passed to the B.M IV. v The output of B.M IV will be two side bands i.e. LSB and USB while carrier of 15MHZ gets suppressed v Then LSB is again filtered out and the USB is passed on to the power amplifier stage d when the signals are amplified to the required power level and transmitted v This is one type of SSB transmitter is I&B. uses independent sideband working. 17. Explain the principle of operation of square law Detector and Mention its merits and demerits. (OR) Explain any one demodulation method of AM [Nov [Nov-04] (OR) Explain the principle of operation of any one non linear Detectors. v Demodulation or detection is nothing but the process of extracting a modulating or base band signal from modulated signal. Otherwise in other words, demodulation or detection is the process by which the message is recovered from the modulated signal at recovered receiver. v The devices used for demodulation or detection are called as demodulators or detector v For amplitude modulation, the detectors or demodulators are categorized are Square law detectors or nonlinear detectors

Linear detectors v The low level modulated signals are using non linear detectors to recover the original message signal Square law detector. (Non linear detector)
v The square law detector circuit is used for detecting modulated signal of small magnitude v Here the diode is used in V I characteristics of the device i.e. non linear characteristics of the diode. v The square law detector is similar to the square law modulator. The only difference lies in the filter circuit. v In square law modulator we are using band pass filters, in square law detector; a low or pass filter is used. v The DC supply voltage VAA is used to get the fixed operating point in the non-linear non portion of the diode V-I characteristics I v We know the operation is limited to the non linear region of the diode characteristics, non-linear so the lower half portion of the modulated waveform is compressed. v This produces the envelope distortion, so the average value of the diode current is no longer constant and varies with time. The average diode current consists of steady DC tant component and time varying modulation frequency. v Due to nonlinear region the lower half of its current wave form is compressed. This may cause envelope distortion due to this diode current will not be constant, and varies diode with time. I
V v The distorted output diode current is expressed by the non non-linear V-I relationship (i.e. I square law) is, i0=a1 Va+ a2 Va2 Va=i/p modulated signal. And we know the value of AM wave is Va= Vc (1 + ma Cosmt) Cosct We are providing supply voltage VAA to provide operating point.

So Va=VAA+ Vc (1 + ma Cosmt) Cosct (that is it contains the DC component and Time varying modulated signal) i0=a1[VAA+ Vc (1 + ma Cosmt) Cosct ]+a2[VAA+ Vc (1 + ma Cosmt) Cosct ]2 Cos i0=a1VAA+ a1 Vc Cosct (1 + ma Cosmt)+a2V2AA+ a2 Vc2 Cos2ct (1 + maCosmt)2 +2 VAA a2 Vc Cosct (1 + maCosmt) i0=a1VAA+ a1 Vc Cosct (1 + ma Cosmt)+a2V2AA + a2 Vc2 (1-Cos2ct) (1 + ma2 Cos2mt+2 ma Cosmt) 2 +2 VAAa2 Vc Cosct maCosmt) The carrier term is filtered by using a filter then, i0=a1VAA+ a2V2AA+ a2 Vc2 + a2 Vc2 (2 ma Cosmt) The higher frequency term also reduced. The VAA is the supply voltage to get fixed operating point in nonlinear region. So the final output contains a DC component and modulating frequency. Hence original signal is recovered. 18. Explain the principle of operation of Envelope Detector (or) Linear Diode detector and Mention its merits and demerits. (OR) Explain any one demodulation method of AM [Nov [Nov-04] (OR) Explain the principle of operation of any one linear Detectors. Linear detectors are mainly used in commercial receivers. This is classified into two types v Synchronous or coherent detector v Envelope or non coherent detector.
(1
Envelope detector: v It is known that a diode operating in linear region of its V I characteristics can extract V-I the envelope of an AM wave. v This type of detector is known as envelope detector or a linear detector. v This is most popular in commercial receiver circuits since it is very simple and not receiver expensive, and also it gives satisfactory performance.

v The input portion of the tuned transformer provides perfect tuning at the desired carrier frequency. R-C network is the time constant network. C v The diode is working diode is working in a linear portion of V I characteristics of the V-I diode. v Figure shows the idealized linear characteristics of the diode along with the input voltage and output current waveforms. Operating principle: v First let us assume that the capacitor is absent in the circuit, in this case the detector st circuit will work as a half wave rectifier. v While we applying the capacitor, for positive half cycle the diode conducts and attains the maximum peak, when it crosses the peak value , the diode gets reverse biased, and capacitor gets discharged, through the resistor R with a time constant =R C. so the time constant value should be properly chosen. v The voltage across the capacitor c will not fall appreciably during the small period of during negative half cycle and by that time the next positive cycle appears. The positive cycle again charges the capacitor c to the peak value of the carrier voltage and thus this process repeats again and again. v So the output voltage across capacitor is a spiky modulating or base band signal. So the envelope is detected at the output of capacitor. Then if we take the average value, from the peak value we will recover our original signal by extracting the envelope. v We can reduce these spikes to a negligible amount by keeping the time constant RC e large so that the capacitor c discharges negligible amount. v Large (or) small value of time constant makes problem. So time constant is important consideration. v If RC is quite low: Discharge curve during non conductive period is almost Discharge vertical, so fluctuations may occur in output voltage voltage. v If RC is very large: Discharge curve is almost horizontal, so several peaks will missed in output. v So when RC is large the ma will be low and signal becomes clipped at the negative peaks. v If the modulating voltage is faster than the rate of voltage fall across RC combination resulting in distorted output. This type of distortion is called diagonal clipping. v There are two clippings i.e. distortions available Distortion in diode detector; Negative peak clipping Diagonal clipping. v Diagonal clipping: may cause when time constant of detector is not selected properly. v Improper selection of time constan RC. constant v So if modulating frequency is faster than the rate of voltage fall across RC combination resulting in distorted output.

Negative peak clipping: v The 2nd source of distortion in linear diode detector is the curvatures of the diode characteristics. So as a result the efficiency varies. It will be reduced by selecting load cs. resistance value large. v The negative peak clipping provides ac and dc load impedances unequal. v To avoid the clipping mainly diagonal clipping we have to select proper valu of RC value then, voltage across the capacitor in no conduction period v The rate of slope in capacitor due to period discharging is calculated by differentiating VC(t) So Like wise we have to calculate the slope of envelope voltage
The slope is
To avoid the clipping the slope of capacitor voltage VC should be algebraically equal or less than the slope of envelope voltage, so & By simplifying So RC depends on the ma value. So the RC to be selected by satisfying the condition in order to avoid distortion and the modulation index value should be correct, if it is large then it will provide negative peak clipping. So the envelope detector will be changed for smooth functioning by attaching a changed section low pass filter at the output. 19. Explain the principle of operation of Rectifier detector. v In the rectifier detector, the modulated signal is fed to the diode through a resonant circuit tuned to carrier frequency v The resonant circuit has a bandwidth sufficient to accommodate the sidebands used. v The tank circuit in input side needed to prevent ripples entering to the diode v It is also the non coherent detector n v Here we are using simple ohms law in order to calculate the output diode current

Input voltage The output diode circuit id = Rd=Dynamic resistance of the diode. If we find the average current then we will obtain the voltage output by Vav=Iav.R If we find the % then it will be 31.8%.that is one third of o/p is taken at the o/p side. So we want heavy filtering to avoid distortions. For this we are connecting the capacitor across the resistor. 20. Explain the principle of operation of synchronous or coherent lain Detector. (OR) Explain the method of detecting the DSB DSB-SC-AM. The DSB-SC modulated signal reaches the receiver through a transmission SC medium and at the receiver end the original modulating signal is recovered fro modulating signal. The coherent detector uses exact carrier synchronization for retrieving the message signal from modulated signal. These types of detectors are mainly used for detecting DSB&SSB signals. mainly For detecting signal we need a local oscillator at the receiver end. the frequency and phase of the locally generated carrier and transmitter carrier must be synchronized that is exactly coherent. All types of linear modulation can be detected by using synchronous detector. It r consists of a product modulator with LPF. The incoming signal is first multiplied with locally generated carrier and then passed through low pass filter and the local oscillator should be exact exactly synchronized with carrier. Considering the DSB-SC signal I/p V1(t)=Vm.Vc cosm t cosc t V2(t)=Vcosc t e(t)= V1(t). V2(t)= Vm.Vc Vcosm t cos2c t e(t)= Vm.Vc Vcosm t[1+ cos2c t] 2 After passing through the LPF
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Considering the SSB-SC signal I/p SC
V1(t)= Vm Vc cos (c - m) t. V2(t)=Vcosc t e(t)= V1(t). V2(t)= Vm.Vc cos (c - m) t. cosc t e(t)= Vm.Vc V[cosm t+cos(2c t-m t)] 2 After passing through the LPF
Thus by using synchronous detector the original message signal is extracted. But the oscillator frequency must be synchronized one. if there is a phase shift then the result in phase or delay distortion. Assume local carrier to have a phase difference with modulated carrier then er there will be the phase difference in the output also. In order to avoid the distortions and to ensure synchronization a pilot carrier is inserted this is separated and used at the receiver for synchronize the carrier. synchronize 21. Explain the principle of operation of costas PLL detection scheme For DSB-SC-AM. One method of obtaining a practical synchronous receiver system suitable for DSB-SC-AM wave is the costas detectors. AM
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v This receiver contains two coherent detectors supplied by same input signal v The frequency of the local oscillator is adjusted to be the same as the carrier frequency fc v The upper path detector -> In phase I channel coherent detector > v the quadrature phase coherent detector -> lower path detector-> Q channel > v To understand the operation, if suppose the local oscillator signal is of the same phase of the carrier wave then the upper detector have the output and lower detector have zero output. v Suppose there is any phase difference then the upper detector output remains unchanged and lower detector output will be some value and that value will be given to phase discriminator and that difference is adjusted by PLL principle so we will get detected output. 22. Draw the block diagram of a super heterodyne receiver and explain the function of each stage briefly. We are using double frequency conversion AM super heterodyne receiver. We are using carrier frequency tuning in order to select the desired signal.(TVorRadio station) It uses heterodyne principle. Heterodyne means mixing two frequencies. .
R.F. section: v The incoming amplitude modulated wave picked up by the receiving antenna is passed on to the RF amplifier. The RF section generally consists of a preselector and an amplifier. v The preselector is a band pass filter with an adjustable centre frequency that i tuned to is the desired carrier frequency of the incoming signal.

v The primary purpose of preselector is to provide enough initial band limiting to prevent unwanted radio frequency called image frequency and it also reduces the bandwidth for noise and allows the useful signal. llows v The RF amplifier amplifies the incoming signal to the requisite level apart from selecting the desired signal, an also it determines the sensitivity of the receiver. v A receiver can have one or more RF amplifiers or it may not have any depending on the desired sensitivity. The advantage of RF amplifiers are (i) greater gain (ii) better sensitivity (ii) better SNR (iv) better selectivity (v) better image frequency rejection. Frequency changes: v The combination of mixer and local oscillator constitute the frequency changer. Both of them provide heterodyne function, where the incoming signal is converted to a predetermined fixed frequency called the intermediate frequency. This intermediate frequency is lower than the incoming carrier frequency. The result of heterodyning is ncy fIF = fRF - fLO v Since the output of the frequency changer is neither the original input frequency nor the output base band frequency, it is called intermediate frequency. Sometimes the frequency. frequency changer circuits are referred to as first detector. Incase of double frequency conversion, the demodulator becomes the third detector. v Although the modulated signal is translated from RF to IF the shaped of the envelope and the original information contained in the envelope remains unchanged. It is important to note that the bandwidth is not altered in heterodyne process. IF section: v This section consists of one or more stages of tuned amplifiers which amplify the intermediate frequency produced at the output of frequency changer circuit. mediate Demodulator or detector: v The output of the IF section is applied to a demodulator which recovers the base band or message signal. If coherent detection is used, then a coherent signal source must be signal provided in the receiver. v The detector also supplies d.c. bias voltage to RF and IF stages in the form of an AGC circuit. Finally the recovered signal is power amplified enrouted to the loud speaker. Audio amplifiers: v This audio section comprises of several cascaded audio amplifiers and one or more speakers. The number of amplifiers used depends on the audio signal power. Image frequency: v In a super heterodyne receiver the mixer will develop an intermediate frequency when the input frequency is greater than or less than the local oscillator frequency. That is there will be two input frequency [fLO & fin].

v This produces two IF signals at the mixer output and results in simultaneous reception of two signals, differing in frequency by twice the intermediate frequency. Thus the frequency mixer is not able to distinguish between the desired frequency and the image frequency. The image frequency is by . v The curve for this image interference is to employ high selective stages in RF sectio section, in order to favour the selection of the desired signal and reject the unwanted signals. Advantages:

The selectivity of this receiver is better. The design of IF amplifier is relatively simple. It provides better sensitivity throughout the broadcast ba band.
TRF Receivers: It consists of two or three stages RF amp. Tuned together to select and amplify the signals. After the signal was amplified it is demodulated and fed to the loud speaker after passing through audio amplifier. 23. Compare various amplitude modulation systems.[Nov systems.[Nov-04] Comparison table: Comparison spectrum of AM systems: If we draw the power spectrum of the AM systems then it will be,
Description B.W Power saving for sinusoidal Power saving for non sinusoidal Generation methods AM with carrier 2fm 33.33% 33.3% Not difficult DSB SC AM 2fm 66.66% 50% Not difficult SSB-SC-AM fm 83% 75% Difficult VSB-AM Fm<2f1 75% 75% Difficult
Signal to noise ratio Application AM broad cast application Carrier telephony Police wireless mobile etc. TV high speed tx
24. Write short notes on Frequency translation.


v It means changing the frequency, frequency mixing or heterodyning. v If suppose we are having modulated wave S1(t) whose spectrum is centered on a carrier frequency f1 and we need to translate its carrier from f1 to new value f2 v Mixer is a device that consists of a product modulator followed by a BPF. v The received radio frequency signal is shifted to IF frequency band for processing v The received signal is translated to fixed IF , can be easily amplified, filtered and he demodulated. v Normally if we are having the voice signal is modulated signal it will be the RF frequency. So we have to provide the down conversion in order to find the intermediate frequency and we are tuning the frequency range to IF frequency. Then the certain signal is demodulated and detected. v Up conversion f2=fi+fl,Down conversion f2=fi-fl. 25. Explain about Frequency division multiplexing, and time division Multiplexing.[Nov-04] Frequency division multiplexing v Multiplexing is a technique in which several message signals are combined into a composite signal for transmission over a common channel. v To transmit a number of these signals over the same channel, the signals must be kept channel, apart so that they do not interface with each other. v This separation is done by either FDM or TDM v Here we are discussing about the FDM techniques v Here number of signals are combined and uses the constant carrier fr frequencies to modulate and passed through common channel. v The incoming message is given to low pass filter to remove high frequency to remove high frequency terms i.e. high frequency components that do not disturb other message signals v The filtered signals are applied to modulators which shift the frequency ranges of gnals signals. v The necessary carrier supply for frequency translation is given to modulators. v For modulation we mostly use SSB techniques v The band pass filters following the modulators are used to restrict the band of each modulated wave to prescribed range. v The resulting signal output are combined together and passed though common channel and at the receiver side the band pass filters in parallel used to separate the message signals in frequency basis. v Finally the original signal is demodulated and filtered by low pass filters at the destination. Time division multiplexing:
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v In TDM the different frequencies are given low pass filter and then passed to a commutator time basis and provides multiplexing nd v The commutator gives the multiplexed channel to a pulse modulator which converts the multiplexed signals in to suitable form. v Finally the TDM channels are assigned to distinct slots jumbled together in frequency domain. University question:Explain the modulation and demodulation of AM signal. Write any one modulation and demodulation scheme. PROBLEMS 26.A transmitter using AM has unmodulated carrier output power of 10KW and can be modulated to a maximum depth of 90% by a sinusoidal modulating voltage without causing overloading. Find the value to which unmodulated carrier power may be increased without resulting in overloading if the maximum permitted modulating index is ulting restricted to 40 %. Soln: Pc= 10 KW , m = 0.9 PT = Pc [ 1+ m2 ] = 10[1+ (0.9)2 /2] = 14 KW. This is the maximum power which may be handled by the transmitter without using overload. If the modulation index is changed to 40% then the increased modulated carrier is then given by 14 = Pc [1+ (0.4)2 /2 ] Pc = 12.96 KW. 27.A sinusoidal carrier voltage of frequency 1MHz and amplitude 100 V is modulated by er a sinusoidal voltage of frequency 5KHz producing 50% Modulation. Calculate the frequency and amplitude of USB and LSB. Soln: Frequency of USB = 1 MHz + 5KHz = 1005KHz. Frequency of LSB = 1 MHz - 5KHz = 995 KHz. Amplitude of USB and LSB = (ma Ec)/ 2 = 100 /4 = 25 V. Psaving = PDSBFC-PSSB PDSBFC x 100
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= 1.125 Pc 0.0625 Pc 1.125 Pc = 94.5 %
UNIT-II ANGLE MODULATION SYSTEMS II
1. Define frequency modulation. Frequency modulation is defined as the process by which the frequency of the carrier wave is varied in accordance with the instantaneous amplitude of the modulating or message signal. 2. Define modulation index of frequency modulation. It is defined as the ratio of maximum frequency deviation to the modulating frequency. mf= m =KVm /m =d 3. Define frequency Deviation in FM?[ FM?[April-04] The instantaneous frequency of FM signal varies with time around the carrier frequency c.This means that the instantaneous frequency of FM signal varies according to the modulating signal. The maximum change in instantaneous frequency form the average frequency c is form called frequency deviation. 4. What do you meant by multitone modulation, Percent modulation? Modulation done for the message signal with more than one frequency component is called multitone modulation. The term percent modulation as it is used in reference to FM refers to the ratio of actual frequency deviation to the maximum allowable frequency deviation. Percent modulation M= 5. Define phase modulation. [Nov [Nov-04]

Phase modulation is defined as the process of changing the phase of the carrier signal in accordance with the instantaneous amplitude of the message signal. 6. What do you mean by angle modulation? And write their types? Angle modulation may be defined as the process in which the total p phase angle of a carrier wave is varied in accordance with the instantaneous value of the modulating or message signal while keeping the amplitude of the carrier constant. Phase modulation (PM) and Frequency modulation (FM) are the types of angle modulation. 7. Write the expression for the spectrum of a single tone FM Signal. [Nov-04] VFM(t) =Vccos(ct+mfsim mt) 8. How FM wave can be converted to PM wave? The PM wave can be obtained from FM by differentiating the modulating signal before applying it to the frequency modulator circuit. Message signal PM Signal
9. How PM wave can be converted to FM wave? The FM wave can be obtained from PM by integrating the modulating signal before applying it to the phase modulator circuit. Message signal PM Signal
10. Define phase deviation. The maximum phase deviation of the total angle from the carrier a angle is called phase deviation. 11. What are the types of Frequency Modulation? (OR) What do you mean by Narrow Band and Wide band? Based on the modulation index FM can be divided into types. T They are Narrow band FM and Wide band FM. If the modulation index is greater than one then it is wide band FM and if the modulation index is less than one then it is Narrow band FM 12. Compare FM to PM

* The FM system having greater Modulation index results in larger band width. * In FM, the modulation index is increased when the modulating frequency is increased & vice versa.
PM system generally uses a smaller bandwidth because of smaller modulation index. In PM, when the modulating frequency is changed, the modulation index in PM remains constant.
13. Compare AM to FM
* In AM system there are three frequency components and hence the bandwidth is finite. *The amplitude of modulated wave in AM is dependent of modulation index. *In AM, most of the transmitted Power is wasted FM system has infinite number of sidebands in addition to a single carrier. Hence its Bandwidth is infinite. dwidth The amplitude of frequency modulated wave in FM is independent of modulation index. In FM, noise is very less.
14. What is the basic difference between an AM signal and a Narrowband FM signal? In the case of sinusoidal modulation, the basic difference between an AM signal and a narrowband FM signal is that the algebraic sign of the lower side f frequency in the narrow band FM is reversed. 15. Compare Wideband FM and Narrowband FM.[April FM.[April-04]
Parameter/Characteristics Modulation index Maximum Deviation Range of Modulating Frequency Maximum Modulation index Bandwidth Applications Wideband FM Greater than 1 75KHz 30Hz to 15KHz Narrowband FM Less than or slightly greater than 1 5KHz 30 Hz to 3KHz
5 to 2500 Slightly greater than 1 Large, about 15 times higher than Small, approximately same as that of BW of narrowband FM AM FM Mobile Communication like Entertainment broadcasting Police wireless, ambulance etc.
16. What are the advantages of Angle Modulation? Angle modulation has several inherent advantages over Amplitude modulation. 1 Noise immunity 2 Noise performance and signal rmance signal-to-noise improvement. 3 Capture effect. 4 Power utilization and efficiency. 17. What is transmission bandwidth of FM? For nside bands the bandwidth of FM wave is given by

B.W=2nm radians/sec B.W=2nfm Hz 18. What are the two methods of producing an FM wave? Basically there are two methods of producing an FM wave. They are, i) Direct method In this method the transmitter originates a wave whose frequency varies as whose function of the modulating source. It is used for the generation of NBFM ii) Indirect method In this method the transmitter originates a wave whose phase is a function of the modulation. Normally it is used for the generation of WBFM where WBFM is generated the from NBFM 19. What are the disadvantages of FM system? A much wider channel is required by FM. FM transmitting and receiving equipments tend to be more complex and hence it is expensive 20. How will you generate message from frequency modulated signals? rate frequency-modulated First the frequency-modulated signals are converted into corresponding amplitude modulated amplitudemodulated signal using frequency dependent circuits. Then the original signal is recovered from this AM signal. 21. Give the frequency spectrum of Narrow band FM. [April [April-04]
22. List the properties of the Bessel function. The properties of the Bessel function is given by, i) Jn (b)=(-1)nJ-n(b) for all n, both positive and negative. ii) For small values of the modulation index b ,we have J0 (b)=1 J1 (b)=b/2 Jn(b)=0,n>2. iii) S J2n (b)=1 n = -

23. What are the types of FM detectors? Slope detector and phase discriminat discriminator. 24. What are the types of phase discriminator? Foster seely discriminator and ratio detector. 25. What are the disadvantages of balanced slope detector? 1. Amplitude limiting cannot be provided 2. Linearity is not sufficient 3. It is difficult to align because of three different frequency to which Various tuned circuits to be tuned. 4. The tuned circuit is not purely band limited. 26. Define capture effect.[Nov effect.[Nov-05] With FM and PM, a phenomenon known as the capture effect allows a receiver to the differentiate between two signals received with the same frequency, providing one signal atleast twice as high in amplitude as the other; the receiver will capture the stronger signal and eliminate the weaker signal. 27. What is FM thresholding? With the use of limiters, FM and PM demodulators can actually reduce the noise level and improve the signal to noise ratio during the demodulation process. This is called FM thresholding. 28. Define Pre-emphasis and de emphasis de-emphasis. Noise at the higher modulating signal frequencies is inherently greater in higher-modulating amplitude than noise at the lower frequencies. i.e the higher - modulating-signal modulating frequencies have a lower signal signal-to-noise ration than the lower frequencies. To n compensate for this, the high- frequency modulating signals are emphasized or boosted in amplitude in the transmitter. To compensate for this boost, the high-frequency signals frequency are attenuated or deemphasized in the receiver after demodulation has been performed. receiver 29. State the Carsons rule. Carsons rule provides a thumb formula to calculate the bandwidth of a single tone wide band FM.According to this rule the FM bandwidth is given as twice the sum of the frequency deviation and the highest modulating frequency.however, it must ation be remembered that this rule is just an approximation. Mathematically B.W=2( B.W=2(+m) 30. What are the disadvantages of FM?

A much wider channel typically 200 KHz is required in FM as against only 10 KHz in AM broad cast. This forms serious limitation of FM. FM transmitting and receiving equipments particularly used for modulation and demodulation tend to be more complex and hence more costly.
31. What is the use of crystal controlled oscillator? The crystal-controlled oscillator always produces a constant carrier frequency controlled there by enhancing frequency stability. 32. In a FM wave the frequency deviation is 25 KHz. What is the Modulation index when the modulating signal frequency is 1 100Hz &10 KHz? Soln: f = 25 KHz ; mf== when fm = 100 Hz radians when fm = 10,000 Hz = 2.5radians 33.A carrier is frequency modulated with a sinusoidal signal of 2 KHz resulting in a maximum frequency deviation of 5 KHz. Find (i) Modulation index (ii) Bandwidth of the modulated signal. Solution: Given data : Modulating frequency fm = 2 K Hz Maximum frequency deviation = f = 5 KHz i. Modulation index = mf = = ii. Bandwidth of the modulated signal is given by, BW = 2 ( f + fm) = 2 (5 x 103+2 x 103 ) = 14 KHz. 34. A 2 KHz audio signal modulates a 50MHz carrier causing a frequency deviation of 2.5KHz.Determine the bandwidth of FM signal. signal.. fm = 2 K Hz fc = 50MHz f = 2.5 KHz Modulation index = mf = 3 B.W=2 fm=2x2x10 =4KHz.

35. Determine the bandwidth of a wideband FM; given that carrier Signal of 100MHz frequency modul modulates a signal of 5 KHz with the 50 KHz as frequency deviation deviation. B.W for WBFM=2 fm Modulation index = mf = fm=50 KHz fm = 5K Hz fc = 100KHz B.W=2x50KHz=100KHz
DESCRIPTIVE ANSWERS: 1. Derive an expression for a single tone FM signal and draw its frequency spectrum.[Nov-03] Frequency modulation is that type of angle modulation in which the frequency of the carrier is varied in accordance with the instantaneous value of message signal. Let the message signal Vm(t) = VmCosmt Let the carrier signal Vc(t) = VcCosct. It is actually denoted as Vc(t) = VcCos(ct+) Vm Max. Amplitude of message signal. Vc Max. Amplitude of carrier signal. m Angular frequency of mod. Signal c Angular frequency of carrier signal = (ct+) Phase angle of carrier Therefore Vc(t) = Vc Cos i Difference with respect to t to find angular velocity ( c) ( = (ct+) d/dt = c and = cdt ----- (1) (Time independent) In FM the amplitude should be constant to the final frequency modulated wave 2 (final FM contains) Where Vc Amplitude of carrier i Instantaneous phase angle

In angle modulation we are changing frequency for FM. So we have to find the instantaneous frequency ie i. Ie during FM the frequency is varied in accordance with the instantaneous value of modulating signal. So
K Constant proportionality .that is frequently sensitivity factor From equn (1) we may write the equn for instantaneous phase angle
The max. and minimum frequency of instantaneous phase angle is 1 for sine term, so If we take minimum value, so min= c KVm If we take maximum value, so max= c+ KVm So = c min = max c = KVm So KVm = = frequency deviation
Therefore Substituting equn (4) in the final frequency modulated wave then
This equation is the representation of frequency modulation considering the single tone. That is modulation is done by considering single. That is only one frequency is is instead of taking the random signals we are taking single frequency component. While solving these equations by using Bessel functions then it will contain more frequency components. In this according to modulation Index value we will select one carrier with pair of side bands. Compare to power more power absorbed in side bands in case of AM. Like this if modulation is done with more than one message signal is called multitone modulation. So the generalized s spectrum of FM signal is In case of single tone we have to use one frequency component. In case of multitone more than one signal and frequency component used.
2. Derive the expression for PM wave. Phase Modulation is defined as the process by which changing the phase of the carrier signal in accordance with instantaneous value of the message signal and amplitude and frequency remains constant. Let the modulating signal

So is varied with phase angle We are taking the instantaneous phase of the message signal so e Where mp = modulation Index of Phase modulation This is for single tone. For generalized expression
3. What is the relationship between PM and FM. (OR) How will you convert PM to FM, and FM to PM? CONVERSION OF PM TO FM FM: Frequency modulated wave can be obtained from PM. This is done by integrating, the modulating signal before applying it to the phase modulator.

An FM and PM differ only in total phase an angle. In Phase modulation (PM), the phase angle varied linearly with base band signal; In case of FM the phase angle varies linearly with the integral of Base band signal.
To get FM by using PM, we first integrate the base band signal and th apply to then the phase modulator. Let and After integration, After phase Modulation the o/p is, The final output is the FM wave. Conversion of FM to PM: The PM wave can be obtained from FM by differentiating the modulating signal before applying it to the Frequency Modulator circuit. Wkt, After differentiation; Output of FM will be, The o/p is the PM wave. 4. What are the types of frequency modulation? And explain about the Narrow band FM and Wide band FM.[April FM.[April-04] Types of frequency modulation (FM): The bandwidth of an FM Signal depends on the frequency deviation

ie W.K.T = KVm frequency deviation If the frequency deviation is high, the B.w will be large. If it is low the B.w will small. And B.w depends on frequency deviation.
Frequency deviation depends on frequency sensitivity factor K. If K is small B.w will be narrow If K is large B.w will be large or wide.
Depending upon the value of frequency sensitivity K, FM may be divided as under. Narrow band FM : K is small, Hence FM is narrow.
Wide band FM : K is large, Hence FM is wide. Narrow band FM: Generation of Narrow band FM: If modulation index value is small compared to one radian. It is narrow band FM. In Narrow band FM generation the carrier is added with the o/p of balanced modulators. O/p of balanced modulators will be the side band terms. The message signal Vm(t) = VmCosmt And carrier signal Vc(t) = VcCos(ct+) = VcCos
The message signal is passed through integrator. Vm(t)dt = VmCosmtdt y(t) = Vm/mSinmt
The normal carrier is represented as VcCosct passed through phase Shifter Therefore Vc(t) = VcCos( ct+90) = -VcSinct Cos( The o/p of Balanced modulator will be = - VcKSinct.y(t) The o/p of narrow band FM is VFM(t) = VcCosct VcSinct (KVm/m) Sinmt
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VFM(t) = VcCosct Vcmf Sinmt. Sinct Applying SinASinB =[Cos(A =[Cos(A-B) Cos(A+B)]/2 Then VFM(t) = VcCosct [(Vcmf)/2][Cos(c-m)t-Cos(c+m)t]
It is similar to AM wave representation. So the B.w also 2fm. So like an AM we can represent the phase representation of an NBFM. The AM contains the LSB and USB are in same phase. In NBFM the side band frequency are in out of phase with carrier.
Wideband frequency modulation: Generation of WBFM: When the value of modulation index mf is quite large, then in FM a large number of sidebands are produced and hence Bandwidth of FM is sufficiently large. This type of ands FM system is known as wideband FM. Final equn for FM is VFM(t) = VcCos(ct + mf Sinmt) Cos(A+B) = CosACosB - SinASinB VFM(t) = Vc[cosctcos(mfSinmt) SinctSin(mfSinmt)] Assuming modulation index mf is small compared to one radian, and then we can make the following assumptions, Cos(mfSinmt)~1 and Sin(mfSinmt)~mfSinmt (because Sin= if is small)
This is the required expression for single tone narrowband FM.


From the block dgm, the difference betn carrier and o/p of BM produces an Narrow band FM but with some distortion. The FM has constant envelope but the NBFM contains residual amplitude modulation and therefore varies with time. So the modulating wave contains some harmonics. Practically a wideband FM is obtained by multiplying the narrowband FM signal by using stable frequency multipliers. We will consider a single tone FM system, the expression for a single tone FM wave is, VFM(t) = VcCos(ct+mfSinmt) 1
We have to simplify the equation by Fourier transforms. Fourier transform is calculated by representing the signals in complex representation. For representing ng complex representation we are taking the real part of FM. Taking the real part, the exponential phases is given by,
ct sinmt) VFM(t) = Vc e(jct + mf sin 2
e(j) = Cos + jSin
VFM(t) = Vc ejct.emfsinmt) 3 In equn 3 the 2nd term is the periodic section with period T = 1/fm, so it is expanded in the form of complex fourier transform,
sinmt In our expression x(t) is ej mf sin therefore the complex Fourier series is,
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And coefficient Cn is Substituting x = mt in eqn 5 x = 2fmt dx/dt = 2fm dx = 2fmdt when t = 1/2fm x= 2 fm(1/2fm) = and t = -1/2fm, x=-
The integral part of the R.H.S is equal to nth order Bessel function of the 1 kind and argument mf. This f function is represented as Jn(mf)
st
Cn = Jn(mf)----------(7) Substituting eqn (7) in eqn (4) then
Substituting eqn (8) in eqn (3) then By taking real part on right hand side it will provide the expression for FM signals.
Substituting n values and making use of the first property of Bessel function, eqn(10) becomes First property of Bessel function Jn(mf) = J-n(mf)for even n Jn(mf) = -J-n(mf) for odd n VFM(t) = VcJ0(mf) Cosct + VcJ1(mf) [Cos( c+m)t cos(c-m)t] [Cos( + VcJ2(mf) [Cos( c+m)t cos(c-m)t] [Cos( + VcJm(mf) [Cos( c+m)t cos(c-m)t] [Cos(

+ .
------------------ (11)
This is considered as mf is taken that it is greater than one. We are using this type of wide band FM in order to obtain wide bandwidth; it is used in mobile communication. In order to evaluate the value of a given pair of sidebands or the value of the carrier, it is necessary to know the value of the corresponding Bessel function. So from eqn(2) the mag. Of carrier is reduced by by J0m(f).and it is having infinite number of sidebands and each side band is represented by separate Bessel function. In order to reduce the number of sidebands we have to consider to mfvalue. If he mf value is low then only J0(mf) and J1(mf) are existing. So it only having a carrier and a pair of sideband term. And it may produce 100% efficiency. So according to modulation index value the sideband varies. If the mf value is small sidebands are less. mf value is large sidebands are more. 5. Write notes on Transmission Bandwidth of FM signal.[April signal.[April-04] We already know the relationship between no. of side band and mf. So for a particular value of modulation Index mf we can calculate the number of sidebands by using table. So a bandwidth also depends on mf value.
The number of significant side bands n produced by plotting Bessel function Jn(mf). The number of sidebands are separated at fm (or) m(we are representing frequency as fm or m) So the total side band is B.W = 2n mrad/sec 2n B.W = 2nfm n no. of side bands. If n contains large no. of sidebands, then it is related to mf. So n~mf B.w = 2mfm = 2( /m) m B.w = 2( ) (or) mf = /m 2 f
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So the approximate B.w of WBFM is given by twice the frequency deviation. This is true when mf>>1. This is also calculated by Carlsons Rule.
Carlsons Rule provides a thumb formula to calculate the B.w of a single tone WBFM. According to Carlsons rule F.M B.w is given as twice the sum of frequency deviation and the highest modulating frequency. and this rule is just an approximation. Mathematically B.w = 2( +m) = 2(mf. m+m) B.w = 2(1+mf) m Regarding Carlsons rule, consider two special case, 1. If <<m is mf<<1 incase of NBFM then B.w = 2m 2. If >>m is mf>>1 incase of WBFM then B.w = 2(mf) m But mf. m = B.w = 2 Bandwidth of PM: The PM B.w as per Carlsons rule, (B.w)PM = 2 = 2kVm. m 6. Draw the graphical representation of AM, FM, PM wave forms and write the advantages and disadvantages of FM over AM. Advantages of FM over AM: FM receives may be fitted with amplitude limiters to remove the amplitude variations caused by noise It is possible to reduce noise by increasing the frequency deviation. sible Standard frequency allocation provides a guard band betn commercial FM station.
mf= /m

The amplitude of FM is const. It is independent of modulation depth. Where as in AM it depends on ma. All the tran transmitted power is useful in FM. But in AM most of the power is carrier power. Disadvantages of FM over AM: FM transmitter and receiver, modulation, demodulation equipments are more complex and hence more costly. A much wider channel 200 KHz is required, but AM using only 10 KHz channel. 7. Explain the method of generating an FM wave. [Nov [Nov-04] (OR) Explain the direct method of Generating FM. (OR) Explain how varactor diode can be u used for frequency modulation? [Nov-05] Generation of FM waves:
Generation of FM consists of two categories, 1. The direct method (or) Parameter variation Method 2. The Indirect Method (o Armstrong Method. (or) Direct Method (or) Parameter Variation Method: In direct method or parameter variation method the base band or modulating signal directly modulates the carrier is the modulating signal directly varies the instantaneous frequency of the carrier signal by means of a device is known as voltage controlled device. The carrier is generated with the help of an oscillator circuit. and the oscillator circuit uses a parallel tuned L- circuit. So the frequency of oscillation -C c = 1/LC Now c is varied in accordance with the message signal, by varying L and C. An oscillator circuit whose frequency is controlled by an modulating voltage is nothing but voltage controlled oscillator.

So the frequency of vco is varied with message signal by putting a shunt capacitor across the tuned circuit. This variable capacitor is known as varactor ( or) varicap. We can also use FET and BJT in order to vary capacitance ie. Miller capacitance. Electron tubes may also provide variable reactants (either inductive (or) ron capacitance) for FM. We may also change the inductance L of the tuned ckt w.r.to modulating signal is known reactor modulator.
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We may use PIN diode, klystron oscillators and multivibrator as voltage multivibrators controlled devices.
Varactor diode method for FM generation:
Varactor diode is a specially fabricated PN junction diode with a variable capacitor that is a semiconductor diode whose junction capacitance changes with d.c bi as voltage. This varactor diode is connected in shunt with the tuned circuit of the carrier oscillator. The coupling capacitor isolates the tuned ckt from varactor diode. The D.C bias to the varactor diode is regulated in such a way that the oscillator frequency is not affected by varactor supply fluctuations. fected
Mathematical analysis: Vd Total voltage of varactor diode. Cd Capacitance of varactor diode. If we take the voltage, capacitance characteristic curve of varactor diode it will be is That is C 1/v
If bias voltage increases width of the capacitance decreases. It may be written as Cd = K/VD Cd = K(VD)-1/2 But w.k.t Vd = V0 + x(t) [Bias vt + modulating vt] Now the osc. Frequency that is frequency produced by the tank ckt varied according to VD. c = 1/LC Total capacitance of tank ckt is C0+Cd and instantaneous FM output is nstantaneous i = 1/L(C0+Cd)
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So we conclude that instantaneous frequency i of FM signal depends upon VD. VD depends upon x(t). So i depends upon x(t). Thus the voltage across the varactor diode controls the frequency of tank ckt in order to obtain the FM output. Applications of this method is 1)Automatic frequency control 2. Remote tuning
8. Explain the reactance tube modulator method for FM generation? (OR) R) Explain the modulation method of FM [Nov [Nov-04] (OR) Draw the circuit of reactance tube modulator and explain its Principle of operation. Derive an expression for the equivalent Capacitance in terms of the gm of the device and circuit Components. [Nov-05] We know that, a reactance tube (either inductive or capacitive) which is proportional to the modulating voltage. This is also another direct method. And these tubes are used to produce FM wa waves. The basic principle of reactance tube modulator is, one must remember that in any device such as FET (or) BJT or tube, the anode current and voltage are in phase, if it could be made to have a phase difference of 90, the device will be have like reactance. The reactance appearing between terminals xx1, corresponds to the reactance between the drain D and the sources of the FET. The reactance of the FET is controlled by gate. The reactance may be a inductive or capacitive by simple component change. and this will be connected to tuned ckt in order to produce frequency modulated wave. Zxx = Vd/Id --------- (1) we know that Tran conductance gm = id/Vgs id = Vgs.gm According to potential divider rule Vgs = Vd.R R-JXc id = Vd.R.Gm -------------(2) R-JXc Sub eqn (2) in(1) Then Zxx = Vd .R-jxc

Vd.R.gm Zxx = R c ----------- (3) R-jx R.gm
In order to make impedance across the terminal xx is pure reactance that is here capacitance we have to make the following requirements. That is consider Xc>>R, hence R is neglected.
Zxx = R (1-jxcR)/R.gm = R[-j/R]/[R mC.]= -j/2fRgmc j/R]/[R-g Sub Ceq = Rgmc Zxx = -j/2fCeq. Where Ceq = gmRc ------- (4) Again we are having resistance term. If it is so large then the voltage will be dropped across R. So again considering the condition Xc>>R So R is related to Xc. So the gate to drain impedance must be selected, with 5 or 10 time greater than gate to source impedance. So Xc = nR 1/ c = nR
Sub eqn (5) in(4) then 1/n = RC RC = 1/n2f 5 Ceq = gmRc. Ceq = gm/2fmn This equation provides the capacitive reactance or it will control the frequency of oscillation and carrier frequency. We can also vary inductive reactance instead of capacitive reactance. Disadvantage of direct method AM generat generation: In direct method FM generation, it is not easy to get a high order stability in carrier frequency because it directly controlled by message signal. So in indirect method the carrier is controlled indirectly. The non-linearity of the varactor diode produces a frequency variation due to linearity produces harmonics of the modulating signal. So the output should be the distorted one. So we are using Direct method is utilized for high power FM generation in several applications. 9. Explain the indirect method of generation of FM wave and any one Method of demodulating an FM wave?[April wave?[April-04] (OR) With neat block diagram explain Armstrong method of generating
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FM signal. [Nov-03] The indirect method (or) Armstrong method of FM generation: In Armstrong method of FM generation, we can get very high frequency stability in this case the crystal oscillator may be used as a carrier frequency generator. The working principle of Armstrong method is used to generate the NBFM method indirectly by utilizing the phase modulation technique and then changing the Narrow band FM into wide band FM by using frequency multipliers.
By taking the Block diagram of Armstrong method of FM generation.
We are producing WBFM from by using a NBFM. We are using PM because its generation is easy. The frequency multiplier not only multiplies the carrier frequency and also increases the frequency deviation and hence the narrow band FM(with small frequency deviation) is converted into wideband FM(with large frequency deviation)
Practical Armstrong Method for FM generation
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The input is integrated and passed through a narrow band PM in order to generate NBFM output. We are mainly using this method because the message signal varies the tank circuit directly in the direct method. So a stable oscillator cannot be used because crystal oscillator cannot be varied as required in FM. We are using non crystal OSC. We are using Mixer ckt in order to change the output frequency i.e. the carrier frequency is translated suitably without altering frequency deviation. For generating the FM. In Armstrong method the main consideration is to maintain the frequency deviation a small. as We are using amplitude limiter in order to reduce the amplitude by the B.M. 10. Describe the operation of Direct and indirect FM transmitters. FM Transmitters: Like the generation method of FM we are also having a Transmission method. That is 1. Directly modulated FM Transmitter 2. Indirectly modulated FM Transmitter Directly modulated FM Transmitter: The transmitter here having a reactance tube modulator, we may also use the varactor diode for FM generation.

The resulting signal is given to LC osc.that is the modulating frequency varies the frequency of oscillation in order to produce the narrow band FM output. If we want WBFM we are using frequency multiplier. The final output is amplified as transmitted. If the output contains harmonics it is corrected by a AFC circuit. That is the error is gn to mixer ckt and this frequency shift is gn to frequency discriminator which gives dc output corresponds to frequency
Shift. Indirect method FM transmitter
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In the indirect method the modulating signal is transmitted through integrator and it is given to phase modulator. We are using phase modulator because generation of PM is easy. And the output is frequency multiplied. If we want to increase the frequency and then power increase amplified and transmitted. We may use frequency multiplier (or) harmonic generator with a mixer cut in order to produce practical Armstrong method FM transmission.
11. Define FM detection. And write their types.Explian the principle Of operation of Simple Slope detector. (OR) Explain the demodulation method of FM signals. [Nov [Nov-04, 05] The process of getting a modulating or Base band signal from a frequency modulated signal is called demodulation or detection. ulated The FM Demodulator or detector performs the extraction of modulating signal in two steps as follows, 1. It converts the frequency modulated FM signal into corresponding amplitude modulated (AM) signal with the help of frequency dependent circuits. So here we are using frequency discriminators. 2. Then the original modulating or base band signal is recovered from the AM signal with the help of linear detector (or) Envelope det detector. FM discriminators are L C tuned Circuits is in frequency dependent Circuits L-C output voltage depends on input frequency, from which the original modulating signal is recovered is called frequency discriminators. Types of FM Demodulators: s: FM Demodulators having two types: 1. Slope detector a. Single tuned detector circuit (or) Simple slope detector

b. Stagger tuned detector circuit (or) Balanced slope detector. 2. Phase difference detectors a. Foster seely detector b. Ratio detector. Slope detector (Simple slope detector): A simple slope detector depends upon the frequency characteristics curve. The circuit consists of a tuned ckt which is slightly detuned from carrier frequency c. ie we are using two tuned circuits which are tuned to two different frequencies. nd First one is tuned to the incoming FM carrier frequency c where as the 2 one is e tuned to a frequency slightly different from carrier frequency (fct f). This portion of the circuit which contains two tuned circuits tuned to different frequencies is called frequency discriminator. This discriminator circuit converts the FM signal into an AM signal as shown in the slope of detector characteristics curve. Then this AM signal is given into envelope detector (or) linear diode detector in order to getting a original modulating or base band signal. nd We know the function of envelope detector; the AM signal envelope is absorbed at the capacitor, and removed to get the original signal. And it is linear.This linear. advantage of this circuit is it is simple and inexpensive Disadvantage, is 1. The Circuits nonlinear characteristic at the characteristics of discriminator produces a harmonic distortion. 2. The harmonics will add up with the signal. So it produces a amplitude. Variation in FM wave then output wave also varied according to the amplitude variation. But a good discriminator circuit must respond only to frequency variation And not to amplitude variations. 12. Explain the principle of operation of Balanced Slope Detector (or) Stagger tuned circuit. (OR) Explain the demodulation method of FM signals. [Nov [Nov-04, 05] Balanced slope detector: The balanced slope detector is an improvement over the simple slope detector. This Circuit of balanced slope detec detector consists of two L-C circuits. C The frequency response is the frequency discriminator characteristic curve. In this circuit, the two tuned circuits are used in the stagger tuned mode,that is one tuned circuit is tuned above the carrier frequency c ie(fc+ f)(curve e1) and another tuned circuit is tuned below c and(fc- f) (curve e2). The resultant curve (e1+e2) is linear as depicted by the dotted line in the characteristics curve.

Since the characteristic curve of frequency discriminator is linear and envelope detector also in linear region, not harmonic distortions will present. So now the frequency discriminator converts the FM in to AM by linear manner then by using an envelope detector circuit, the original signal is recovered separately and added together. That is (e1+e2). But we are also having disadvantages in this detector circuit. 1. Linearity is not sufficient. 2. It is very difficult to align because of three different frequencies to which Various tuned ckts are to be tuned. uned 3. The tuned ckt is not purely band limited and hence the Low pass Rc filter
of Envelope detector introduces distortion. We know that the linearity of this Circuit is insufficient so the circuit is not purely band limited. ficient 4. Frequency deviation should be less otherwise the circuit will produce Distortion due to non non-linearity. Because increased then curve slope Also increases. 13. Explain the principle of operation of Foster seely detectors. (OR) Explain the demodulation method of FM signals. [Nov [Nov-04, 05] [April-04] Foster seely detector: This type of detector is most widely used. s This circuit consists of an inductively coupled double tuned circuit in which both primary and secondary coils are tuned to the same frequency that is intermediate frequency. The transistor is used as a on, off controller. The centre of the secondary coil is connected to the top of the primary (collector end) through a capacitor c. C performs the following functions: 1. It blocks the d.c from primary to secondary. 1. It couples the signal frequency from primary to centre tapping of the Secondary.
Here there is no need of separate tuning.Therfore the primary voltage v3appears across the inductor. The entire voltage V3 appears across inductor L except a small drop across the capacitor C. The centre tapping of the secondary coil has an equal and opposite voltage across each half winding. Hence V1and V2 are equal in magnitude but opposite in phase.
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Va1 = V3 + V1 Va2 = V3 - V2 So we are having a separate phase difference according to the input frequency.
The conditions are: 1. If input frequency fin = fc fin (Modulated frequency) fc Carrier frequency, centre frequency. In this case V3 is in 90 out of phase
2.If input frequency fin>fc then the phase difference between V3andV1 is 45, and V3 and V2 is 135
3.
If input frequency fin<fc then the phase difference between V3 and V1 is 135 And V3 and V2 is 45
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Voltages Va1 and Va2 depends upon the phase relations between V1 and V2and V3, So according to the Phasor diagram and depending upon the different frequency (ie modulated frequency) we are getting the Amplitude modulated signal from that we are getting our original signal by bypass capacitors. So final output
They are taking V02 V01 because of can be tapped transformer the two voltages are equal in magnitude. And opposite in phase. An either V2 will be high or And V1 will be high. So we are taking V02 V01. If we are taking the dotted lines it will be under linear characteristics of frequency discriminatior.That is fc and fc+f and fc- f.This method is the foster seely method.
14. Explain the principle of operation of Ratio detectors. plain (OR) Explain the demodulation method of FM signals. [Nov [Nov-04, 05] [April-04] Ratio Detector: Ratio Detector is an improvement over the foster seely discriminator and is most widely used. It does not respond to amplitude variations therefore the limiter is not needed. The circuit diagram is similar to the circuit of foster seely discriminator except the following. 1. The polarity of diode D2 has been reversed. ity 1. The output V0 is taken from the center tap of a resistor R which shunts the Load impedance of the two diodes. The output voltage is varied according to FM output as in the case of foster seely detector. Ie V01 and V02 are rectified output then the rectified output is passed through the capacitor in order to obtain original signal. But here the only difference is magnitude is decreased. The voltage V01 and V02 have the same magnitude as in the case of a foster seely discriminator as V02 is reversed in polarity. VR is the sum of V02 and V01. Hence, And the output Voltage V0 is taken across the terminal t1 & t2. From the ckt diagram:

The ratio detected has exactly same behaviour except that its outp is reduced. output But we are not having separate amplitude limiter in this circuit. Advantage of foster seely detector: 1. The phase discriminator is much easier to align than the balanced slope detector. 2. Only two tuned circuits are necessary and both are tuned to same frequency. 3. Linearity is better.
Disadvantages of foster-seely detector: seely 1. It needs separate amplitude limiting circuit. Advantage of Ratio detector: 1. No need of separate amplitude limiter, because if any frequency variation, then automatically amplitude will be limited itself.
15. Explain the principle of operation of FM super heterodyne Receiver.
We know that the carrier frequency tuning, filtering, Amplification are done in Receiver part other than Demodulation. Carrier frequency tuning the purpose of which is to select the desired signal. Filter which is required to separate the desired signal from other modulated signals.
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Amplification Which is intended to compensate for the loss of signal power incurred in the course of transmission. This is the basic block diagram of FM super heterodyne Receiver consists the following. RF section: The function of RF section is similar to that employed in AM Receiver. They employed select the desired signal from many modulated signals and amplify them to the required level. The only difference of FM RF section compare to AM is for FM the receivers are designed to operate at VHF and VHF range. Frequency changer: It is nothing but the combination of mixer and the local oscillator. The function of this frequency changer is same as AM and it produces Intermediate frequency. IF section: The IF section provides the Bandwidth to accommodate the frequency modulated accommodate signal. And the If amplifier amplifies the Intermediate frequency present at the output. Limiter: The limiter circuit is the next block to the IF section. It is provided to limit the amplitude changes mainly noise level. And then Discriminator provides AM signal which level. is detected using the envelope detector. And that signal will be power amplified and given to the destination. But if we are using the Ratio detector there is no need of using Limiter Circuit. Because this circuit itself having the limiter circuit. The signals received by receivers vary in their strength, there are 1. Space diversity reception The Voltage induced in antennas spaced few wavelength apart. 2. Frequency diversity reception Signals from the distant transmitted received y at any receiving point at different frequencies. We are also having high frequency communication receivers are high quality, short wave, multipurpose super heterodyne Receives also having Band spread tuning, Receives Double frequency conversion, Tuning calibration, Noise limiter, AFC, Tuning Indicator. 16. Explain the principle of operation of FM Broad cast Receiver. FM broad cast receiver is a super heterodyne circuit with a tuned RF amplifier so that maximum signal sensitivity is typically between 1 to 10 10V. The RF stage tuned circuits and the local oscillator are tuned by a three ganged variable capacitor controlled from a panel knob. The IF amplifier section is comprised of several gain stages, of which one or more are amplitude limiters. The schematic shown here has one high gain nonlimiting input stage followed by one amplitude limiting stage. In the block diagram of an Fm receiver that uses AGC.AGC may be provided to control the RF and early IF stages so that saturation of the nonlimiting IF stages ontrol does not occur on strong signals.
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In this case a sample of the IF signal is extracted just before the input to the limiting IF amplifier. This sample is applied to a special detector used only to obtain the AGC signal. special The derived AGC signal is applied to control the RF amplifier and the first IF amplifier. its time constant is similar to that used in AM receivers.
To left channel audio amp Composite audio signal
17. Explain the principle of operation of FM Stereo Receivers.
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All FM broad cast receivers are being built with provision for receiving stereo or two channel broad casts. The left(L) and right(R)channel signals from the program material are combined to form two different signals, one of which ids the left left-plusright signal and one of which is the left left-minus-right sinal. The (L-R)signal is double sideband suppressed carrier modulated about a carrier R)signal frequency and (L+R) signal is placed directly and a pilot carrier is added to synchronize the demodulator at the receiver. The output from the FM detector is a composite audio signal containing the frequency multiplexed (L+R) and (L (L-R) signals and a pilot tone. This composite d signal is applied directly to the input of the decode matrix. The composite audio signal is also applied to one input of a phase error detector cuircuit, which is part of a phase locked loop 38 KHz oscillator. The output drives 38-KHz driv the 38 KHz voltage controlled oscillator whose o/p provides synchronous carrier for the demodulator. The oscillator o/p is also frequency divided by 2(in a counter circuit) and applied to the other input of the phase comparator to close the phase locked loop. The phase error signal is also passed to a Schmitt trigger circuit, which drives an indicator lamp on the panel that lights when error signal goes to zero, indicating the presence of a synchronizing i/p signal. The outputs from 38 kHz oscillator and the filtered composite audio signals are applied to the balanced demodualtor, whose o/p is the (L R) channel. The (L+R) (L-R) and (L-R) signals are passed through a matrix circuit that separates the L and R R) signals from each other. These are passed through de mphasis networks and low de-emphasis pass filters to remove unwanted high frequency components and are then passed to the two channel audio amplifiers and speakers. On reception of a monaural signal, the pilot tone and closes the switch to disable the (L (L-R) input to the matrix. he The (L+R) signal is passed through the matrix to both outputs. An ordinary monaural receiver tuned to stereo signal would produce only the (L+R)signal, since all frequencies above 15 KHz are removed by filtering and no demodulator circuitry is present. Thus the stereo signal is compatible with the monaural resent. receivers. 18. Explain about FM Stereo multiplexing. Stereo multiplexing is a form of frequency division multiplexing (FDM) designed to transmit two separate signals via the same carrier. it is widely used in FM radio broad casting to send two different signals via same carrier. It is widely used in FM radio broad casting to send two different elements of a program so as to give a spatial dimension t its perception by a listener at the receiving end. The specification of standards for FM stereo transmission is influenced by two factors: 1. The transmission has to operate within the allocated FM broad cast
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Channels. 2. It has to be compatible with monophonic radio receivers. The first requirement sets the permissible frequency parameters, including frequency deviation. The second requirement constrains the way in which the transmitted signal is configured.
From the block diagram of the multiplexing system used in an FM stereo transmitter, Let m1(t) and mr(t) denote the signals picked up by left-handed and left right-hand microphones at the transmitting end of the system. hand They are applied to a matrixer that generates the sum sig signal, m1 (t) + mr (t), and the difference signal, m1 (t) - mr (t).The sum signal is left unprocessed in its base band form. It is available for monophonic reception. The difference signal and a sub carrier are applied to a product modulator, thereby producing a DSB-SC modulated wave. SC In addition to then sum signal and this DSB SC modulated wave, the multiplexed DSB-SC signal m (t) also includes a pilot to provide a reference for the coherent detection of the difference signal at the stereo receiver. Thus the multiplexed signal is described by lexed m(t)=[ m1(t)+ mr(t)]+[ m1- mr(t)]cosct+Kcosct K-indicates the amplitude of the pilot tone. The multiplexed signal m (t) then indicates frequency modulates the main carrier to produce the transmitted signal. At a stereo receiver the multiplexed signal m (t) is recovered by frequency eceiver demodulating the incoming FM wave. Then m (t) is applied to the demultiplexing system shown in figure. The individual components of the multiplexed signal m (t) are separated by the use of three appropriate filters. The recovered pilot is frequency doubled to produce e the desired sub carrier. The availability of this sub carrier enables the coherent detection of the DSB DSB-SC modulated wave, thereby recovering the difference signal, m1 (t) mr (t) The base band low pass filter is designed to pass the sum signal, m1(t)+ mr(t).finally the simple matrixer reconstructs the left hand signal m1(t) and rightleft-hand hand signal mr(t),except for scaling factors, and applies them to their respective speakers.
19. Explain the operation of a PLL FM demodulator.
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Vd
Demodulated Audio o/p FM I/P
A PLL frequency demodulator requires no tuned circuits and automatically compensates for changes in the carrier frequency due to instability in the transmit oscillator. * After frequency Lock had occurred the VCO would track frequency changes in the input signal by maintaining a phase error at the input of the phase comparator. * The phase comparator produces an error voltage that is proportional to the produces difference in frequency between two input signals. If the PLL input is a deviated FM signal and the VCO natural frequency is equal to the If center frequency , the correction voltage produced at the output of the phase comparator and fed back to the input of the VCO is proportional to the frequency deviation and is, thus, the demodulated information signal. * The low pass filter blocks the two input frequencies (fi and fo) and the sum frequency. Thus, the only signal allowed to pass through to the output of the LPF is the relatively low difference frequency. 20. Write notes on non linear effects in FM systems. Frequency modulation can also done by considering non linear characteristics. Two basic forms of non non-linearity to consider: 1. The non linearity is said to be strong when it is introduced intentionally and in a controlled manner for some specific application. Examples of strong nonlinearity include square-law modulators, hand limiters and frequency multipliers. modulators,

2. The non linearity is said to be weak when a linear performance is desired. The effect of such non linearities is to limit the useful signal levels in a system and thereby become an important consideration. Consider a communication channel, the transfer characteristic of which is defined by non linear input-output relation. output V0(t) =a1 Vi(t)+ a2 Vi2(t)+ a3 Vi3(t) Where Vi(t)and V0(t)are the input and output signals respectively and a1, a2, a3 are constants. V0(t) is the instantaneous function of input signal Vi(t). And the FM signal is defined by Vi(t)= VcCos[ct+(t)] Where For this input signal the equation becomes, V0(t)= a1VcCos[ct+(t)]+ a2V2cCos2[ct+(t)]+ a3Cos3[ct+(t)] Thus the channel consists of a DC component and three frequency modulated signals ists with carrier frequencies fc, 2fc, and 3fc,the latter component are contributed by the liner, second order and third order terms. To extract the desired FM signal from the channel o/p V0(t),that is the particular component with carrier frequency fc,it is necessary to separate the FM signal with this carrier frequency from the one with the closest carrier frequency, 2fc. So the channel o/p is reduced to Therefore the only effect of passing an FM signal through a channel with FM amplitude nonlinearities, followed by appropriate filtering, is simply to modify its amplitude. That is unlike amplitude modulation, frequency modulation is not affected by distortion produced by transmission through a channel with amplitude channel non linearities.An FM system is extremely sensitive to Phase nonlinearities.
21. In a FM wave the frequency deviation is 25 KHz. What is the modulation Index value when the modulating signal frequency is 100Hz and 19 KHz? f = 25 KHz, If fm = m = 10 KHz then, mf = f/ m = 25 KHz/10KHz = 2.5 radians.

22. A 2 KHz audio signal modulates a 50MHz carrier causing a Frequency deviation of 2.5 KHz. Determine the B.W of FM Signal. fm = 2KHz fc = 50MHz f = 2.5 KHz W.k.t mf = f/fm= 2.5KHz/2KHz=1.25 B.w = 2fm = 2 x 2 x103 B.w = 4 KHz
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UNIT=III NOISE THEORY 1. Define noise. Noise is defined as any unwanted form of energy, which tends to interfere with proper reception and reproduction of wanted signal. 2. Give the classification of noise. Noise is broadly classified into two types. They are External noise and internal noise. External noise may be defined as that type of noise which is generated external to the communication system. And it can be classifi into classified 1. Atmospheric noise 2. Extraterrestrial noises 3. Man made noises or industrial noises made Internal noise may be defined as that type of system which is generated internally or within the communication system or receiver. it can be classified into 1. Thermal noise 2. Shot noise 3. Transit time noise 4. Miscellaneous internal noise 3. What are the types of extraterrestrial noise and write their origin? The two type of extraterrestrial noise are solar noise and cosmic noise Solar noise is the electrical noise eminating from the sun. Cosmic noise is the noise received from the center part of our galaxy, other Distant galaxies and other virtual point sources. 4. Define Atmospheric noise and industrial noise? Atmospheric noise, which is also called static, is produced by lightning spheric discharges in thunderstorms and other natural electrical disturbances which occur in the atmosphere. The industrial noise or man made noise is that type of noise which is produced man-made by such sources as automobiles and aircraft ignition, electrical motors, switch uch gears and leakage from high voltage transmission lines and several other heavy electrical equipments. 5. Define shot noise? [Nov-05, Apr 05, Apr-04]

Shot noises arises in active devices due to random behaviour of charge carriers. In electron tubes, shot noise is generated due to random emission of electrons from cathodes, whereas in semiconductor devices shot noise is generated due to random diffusion of minority carriers or simply random generation and recombination of carriers electron-hole pairs.
6. Define partition noise? Partition noise is generated in a circuit when a current has to divide between two or more paths. This means that partition noise results from the random fluctuations in the division. 7. Define flicker noise (or) Low frequency noise. Flicker noise is the one appearing in transistors operating at low audio frequencies. Flicker noise is produced at low frequencies (below few KHz).This noise is also called as flicker noise (1/f noise). 8. Define transit time of a transistor. Transit time is defined as the time taken by the electron to travel from emitter to the collector. 9. Define Transit-Time noise or high frequency noise. Time It is generally observed in semiconductor devices, when transit-time of charge erved transitcarriers crossing a junction is comparable with time period of the signal. Some charge carriers diffuse back to the source, this process gives rise to the input admittance and it affects the conductance with increase in frequency. This conductance conductance produces transit time noise. 10. Define Avalanche noise? The reverse bias characteristic of a diode shows a region where the reverse current increases rapidly with a slight increase in magnitude of the reverse bias voltage. That is voltage increase current also increases. This is because the holes and electrons in the depletion region gain sufficient energy from reverse bias to ionize atoms by collision. This collision provides spikes in current in avalanche region. This noise is called as avalanche noise. valanche 11. Define Thermal noise? The thermal noise or white noise or Johnson noise is the random noise which is generated in a resistor or the resistive component of complex impedance due to rapid and random motion of the molecules, atoms and electrons. ndom 12. Write an expression for thermal noise generated in a resistor. The expression for maximum noise power output of a resistor may be given as

Where
K-Boltzmanns constant=1.38x10-23Joule/deg.K Boltzmanns T-absolute temperature. absolute B-bandwidth of interest in Hz. bandwidth
13. Explain White Noise. Many types of noise sources are Gaussian and have flat spectral density over a wide frequency range. Such spectrum has all frequency components in equal components portion, and is therefore called white noise. The power spectral density of white noise is independent of the operating frequency. And it is also defined as noise in idealized form is known as white noise. This means that in a communication system, the noise analysis is based on an idealized form of noise, is white noise. 14. Give the expression for noise voltage in a resistor. The mean square value of thermal noise voltage is given by square Vn2 = 4 K T B R K Boltz man constant R Resistance T Absolute temperature B - Bandwidth 15. Define equivalent noise bandwidth of an ideal band pass system. Equivalent noise band width may be defined as the bandwidth of an ideal band pass system which produces the same noise power as the actual system does does. 16. Explain why thermal noise is also called as Johnson noise? Thermal noise is also called as Johnson noise after the scientist J.B.Johnson who presented a detailed investigation of it. 17. Define signal to noise ratio. Signal to noise ratio is the ratio of signal power to the noise power at the same point in a system. 18. Define noise figure. S/ N at the input Nose figure F = S/ N at the outp output S/N = Signal power / Noise Power 19. What is narrowband noise?

The receiver of a communication system usually includes some provision for preprocessing the received signal. The preprocessing may take the form of a narrowband filter whose bandwidth is large enough to pass modulated component of the received signal essentially undistorted but not so large as to admit excessive noise through the receiver. The noise process appearing at the output of such filter is called narrow band noise. 20. Give the representation of narrowband noise in terms of envelope And phase components. Narrowband noise in terms of envelope and phase components as
The function r (t) and (t) are called envelope and phase of n (t). t)
21. Define figure of merit? The figure of meritmay be defined as the ratio of o/p signal to noise ration to may input signal to noise ration of a receiver. 22. What is the figure of merit of DSB DSB-SC system? The figure of merit of a DSB DSB-SC system is 2. 23. What is the figure of merit of SSB SSB-SC system? The figure of merit of an SSB- system is 1. -SC 24. An amplifies operating over the frequency range from 18 to 20 MHz has a 10Kh input resistor. Calculate the RMS voltage at the input to this amplifier if the ambient temperature is 27C. Given data:R = 10 K , K = 1.38X10-23J/K(Boltzman constant) B.w B = (18 20) that is (20 18) MHZ = 2MHz T = 27C + 273 = 300K The RMS voltage is given by th expression, the 4RKTB = 4X10X103X1.38X10-23X2X106X300 4X10X10 11 = 4X1.38X3X2X10-11 = 1.82X10-5V = 1.82V. 25. Two resistors of 20k and 50k are at room temperature of 15 15c or 288k. For a given B.w of 100KHz determine the thermal noise voltage generated by i).Each resistor, ii).The two resistors in series, iii). The two resistors are in parallel. 1. thermal noise in First resistor Vn1 = 4KTBR1 = 4X1.38X10-23X288X100X103X20X103.

Vn1 = 5.6V And in 2nd resistor Vn2 = 4KTBR2 = 4X1.38X10-23X288X100X103X50X103 Vn2 = 8.91V. ii. when two resistors in series Vnr = 4KTB (R1+R2) = 4X1.38X10 -23X288X100X103X (20+50) X103. = 10.55V iii. when two resistors are in parallel Vnr = 4KTB Rpar 1/Req = 1/R1 + 1/R2 = (14 3) X 103 -23 = 4X1.38X10 X288X100X103X14.3X103 = 4.7V. 26. What is the signal-noise-ratio at the output of DSB ratio DSB-SC system With coherent demodulation? [Nov [Nov-03]
27. Write the advantages of Super heterodyne receiver over TRF Receivers?[April-04,Nov 04,Nov-04] The super heterodyne receiver has the following advantages over TRF receivers. 1. Improved selectivity in terms of adjacent channels. 2. More uniform selectivity in terms of adjacent channels. 3. Improved receiver stability. 4. Higher gain per stage because IF amplifiers are operated at a lower frequency. 5. Uniform band width because of fixed intermediate frequency. DESCRIPTIVE ANSWERS: CRIPTIVE 1. Define noise and write notes on shot noise?[Nov noise?[Nov-04,05] Noise may be defined as an unwanted form of electrical signal or energy tends to interface with the proper reception and reproduction of transmitted signals. For eg. 1. In receivers several electrical disturbances produce noise and thus ceivers modifying the required signal in an unwanted form. 2. In TV Receivers noise may produce super imposed images. 3. In pulse communication, noise may produce unwanted pulses or Cancel the required pulses. Shot noise: Shot noises arises in active devices due to random behaviour of charge carriers. Eg. In electron tubes, shot noise generated due to the random emission of electrons from cathodes.
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In semiconductor devices shot noise is generated due to the random diffusion of minority carriers or simply random generation or recombination of electron hole electronpairs.
The current in electron devices (i.e. tubes or solid state device) flows in the form of discrete pulses. Every time a charge carrier moves from one point to another (i.e. cathode to plate). It seems to be continuous but it is a discrete one. At lower voltages the electrons emitted by the cathode does not reach the anode, does some electrons remains inside the region between anode and cathode. Hence this region is called space charge region. When we take the value, i.e. total current i(t) it will be i(t) = I0 + in(t) I0 constant current is average or mean value of current i.e. current pulses at s Every time. in(t) Noise current .it fluctuates about the mean value I0. This current in(t) is shot noise current and it is not visualized by normal instrument and it is viewed by fast sweep oscilloscope. The fluctuation may be smoothened if the emission rate decreases because if emission rate decreases the space charge is reduced and more electrons reach the plate properly. Power density spectrum of shot Noise: We know the shot noise current is given by i (t) = I0+in(t) And the function in(t) is the noise current and it is a random function. And this is a in deterministic function. And we have to calculate the power density spectrum of this indeterministic function. This noise is also approximated by using Gaussian process. We can treat the total diode current as the sum of current pulses and each pulse is formed by the transit of an electron from the cathode, to the anode. The power density spectrum of statistically independent non-interacting random interacting noise current in(t) is expressed as Si() = qI0 ) q Electronic charge, I0 Mean value of the current in amperes
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That is leakage charge in current will be power spectral density of noise current. density This power density spectrum is frequency independent. This type of frequency Independence is only up to a frequency range determined by the transit time of an electron. Beyond this it is frequency dependent i.e. poles will vary.
The transit time of an electron in a diode depends upon anode voltage V and may be expressed as, d spacing between anode and the cathode And the shot noise voltage is given by
The mean square noise component is proportional to the dc flowing and for most devices the mean square, shot noise current is given by Where Idc is the direct current in amperes,qe the magnitude of electron charge(=1.6x1019 C).and Bn is the equivalent noise bandwidth in hertz. 2. Write short notes on Thermal noise. [Nov [Nov-04] The thermal noise white noise or Johnson noise is the random noise which is generated in a resistor or the resistive component due to random motion of electron (or) molecule and atoms. The random noise voltage will not be a instant value; it will be a rms value. will The noise power produced in a resistor is proportional to absolute temperature. And also the noise power is proportional to Band width over which the noise is measured. The maximum power Pn is calculated as PnT.B (or) Pn=K.T.B Where K Boltzmanns constant 1.38 X 10-23J/k T absolute temp B Band width in Hz. If we are taking a resistive component, if we connect a d.c volt meter across any given resistor at room temperature i.e. 270 C, then no voltage is displayed by a D.C volt meter. A very sensitive electronic component is used to display some reading.

(Thevenins equivalent Voltage model of Noisy resister)
If we are taking a resistor R then the noise voltage is produced by the random moment of electrons, within the resistor constituting a current. When measuring over a long period of time, the as many electrons arrive at one end as of other, if we are taking a random moment; the random voltage across the taking resistor is measured at a temperature. This random noise voltage is expressed in terms of rms value, rather than instantaneous value. According to maximum power transfer theorem, the power of noise voltage Vnto a load Resistor RL = R, then We know the noise power Pn = V2/(R+RL) (P=V2/R) 2 That is Pn = V rms/(R+RL) And W.k.T Vrms = Vm/2 = Vn/2
K Boltzmanns constant R Resistor T Temperature B Bandwidth B over which noise is being measured. Nortons theorem may be used to find an equivalent current model of a noisy resistor. If we take the current model of noisys resistor then
By using G = 1/R the rms noise current In for current model of a noisy resistor is expressed as In2 = 4GKTB Addition of noise due to several sources in series:
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Let us consider the thermal noise for sources R1,R2,R3 etc in series producing noise voltage Vn1,Vn2,Vn3 etc then w.k.t the rms voltage of R will be
Addition of noise due to several sources in parallel: Let us consider the thermal noise source is resistors R1,R2,R3 are connected in parallel then the noise rms voltage will given by
3. Write notes on Spectral densities We know about the power spectral density of a noise wave form which will be a random one and not able to predetermined. That means the total average Power is equal to the product of power spectral density and Bandwidth. If we are calculating the power spectral density of the noise current due to free he electrons is given by Where K Boltzmanns constant average number of collision/second T Temperature in degrees Kelvin. G Conductance of the conducti medium. conducting
If we take the variation in power density then the Power density spectrum is frequency Independent is almost constant in the frequency range. That is the range covers almost all the practical applications in communication systems. So for all practical purposes the Si() is frequency Independent and ) constant. So it is represented as Si () = 2KTG
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So the contribution of any circuit is determined by the Bandwidth of the circuit. Thermal noise has a power density spectrum which is constant at all frequencies hence it is also referred to as white noise it is also referred to as Johnson noise after the scientist J.B.Johnson who inversed this details.
4. Derive the expression for thermal noise using power spectrum Density. For calculating the expression of noise power by using power spectral density we are taking the voltage model and current model of the Resistor circuit. current Here the resistance source is represented interms of Thevenins voltage source or Nortons current source. And we know that the noise voltage generated by a resistor is given as Vn(t) = in(t).R Now since the power density spectrum is a function of the square of voltage or current, the relationship between the power density spectrum Si() as Sv() with ( in(t) and Vn(t) is expressed as
Therefore Sv() = R2Si() W.k.T Si () = 2KTG
( / is constant)
Sv () = 2KTGR2 Sv () = 2KT (1/R)R2 (G=1/R) Sv () = 2KTR W.k.T the total power P in terms of power spectral density spectrum S( S() is given as (Power and Psd relation formula by F.T) And it is changed as is noise power in terms of power spectral density is given as
For infinite Band width width------ (1) If we are having a finite Bandwidth form ( f to f) (Then Pn = Sv ().B = 2KTR. (2 f) Pn = 4KTR f -------- (2) Therefore noise voltage and power relationship is Pn Vn2 Equation 1, 2, 3 are the noise power voltage using psd.

5. Calculate the thermal noise for a single noise source. If we are having a single noise source the then
Due to that single source the output noise voltage Vno(t) is determined. W.k.T Vni (t) input noise voltage, Sni (t) input power spectral density Vno (t) output noise voltage Sno (t) output power spectral density r H () transfer function. Relationship between input and output psd is Sno () = |H () |2 Sni ( d ()
The output noise power P0 may be calculated by integrating output power spectral density Sno( ) Power is related to mean square value V2rms Therefore
Equation (1) and (2) is the single source noise power and voltage using power spectral density 6. Calculate the Thermal noise in RC circuits. Let us consider an RC Circuit, and let us obtain the equivalent Circuit of noiseless and Resistance, by considering a Thevenins model.
So a noise source is connected in seriously to a Resistor. We know the power spectrum density of an Resistor with input noise voltage is Sni () = 2KTR. This input noise source is passed through the RC filter. So if we are finding the transfer function of RC filter it will be
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If we take the conjugate then
Then the next step is the calculation of output power spectral density Sno () = |H () |2 Sni( ) Sno () = 1/ (1+ 2R2C2) x 2KTR ------- (1) For calculating output noise power from power spectral density, the next step is
tan-10 = 0, tan-1 = /2 And noise voltage interms of P0 is P0 V02 and V0 = P0 There fore
If we calculate the thermal noise for multiple noise sources then the power spectrum density due to several sources are Sy () = Sy1 () + Sy2 ( +..+Syn () () 7. Write notes on White noise. (OR) Explain about white noise. Noise in an idealized form is known as white noise. As we know that white light consists of all colour frequencies, in the same manner, white noise contains all frequencies in equal amoun amount. The power density spectrum of a white noise is independent of frequency. This means that white noise consists of all the frequency components in equal amount. If we are taking that is specifying the Gaussian distribution in terms of white noise then it is called as white Gaussian noise. t The power spectral density of white noise is expressed as (Half the power for +ve frequencies, half the power Will be for ve frequencies) The dimensions of N0 are in watts per Hertz. The parameter N0 is usually referenced to the input stage of the receiver of a communication system. It may be expressed as

N0=kTe Where k is the Boltzmann constant and Te is the Equivalent noise temperature of the receiver. ture The equivalent noise temperature of a system is defined as the temperature at which a noisy resistor has to maintained such that, by connecting the resistor to the input of a noiseless version of the system, it produces some available noise power at the output of the system as that produced by all the sources of noise in the actual system. The important feature of the equivalent noise temperature is that it depends only on the parameters of the system. Since the autocorrelation function is the inverse Fourier transform of the power function spectral density, it follows that for white noise
That is auto correlation function of white noise consists of a delta function weighted by the factor N0/2 and occurring at t=0, as in the figure. If the white noise W (t) is Gaussian then two samples are statistically independent. In a sense, White Gaussian noise represents the ultimate in randomness. White noise has infinite average power and as such, it is not physically realizable.
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The utility of a white noise process is parallel to that of an impulse function analysis of linear systems. We may state therefore, that as long as the bandwidth of a noise process at the input of a system is appreciably larger than that of the system itself, then we may model the noise process as white noise. Ideal low pass filtered white noise Suppose that a white Gaussian noise w(t) of zero mean and power spectral density N0/2 is applied to an ideal low pass filter of band width B and pass band magnitude response of one. The power spectral density of the noise n(t) appearing at the filter output is therefore
The auto correlation function of n(t) is the inverse Fourier transform of the power spectral density
The autocorrelation function is plotted.RN (t) has its maximum value of N0B at the origin and it passes through zero at t = k/2B, Where k=1, 2, 3
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Since the input noise w (t) is Gaussian, it follows that the band-limited noise n (t) limited at the filter o/p is also Gaussian. Suppose now that n (t) is sampled at the rate of 2B times per second. The joint probability density function of a set of noise samples obtained in this way is equal to the product of the individual probability density functions.
8. Write notes on Noise Equivalent Band Width. With the help of equivalent Band width, the noise power can be specified at the output of linear band pass system. Let us consider a linear band pass system; the system is excited by an input noise band voltage source Vni (t) producing an o/p Vno (t).Let H () be the transfer function of ) 2 linear band pas system. The parameter | H ( | , the square of the transfer function () is plotted.
(Linear band pass system)
In the above figure the negative half of the curve is symmetrical about the vertical axis and hence the positive half of the plot | H () |2 has beens shown. We know that the power spectrum density of noise voltage at the input and power spectrum density at the output are related as Sno ( = Sni ().| H () |2 () Sni () input power spectral density ) Sno () output power spectral density The mean square value of output noise signal, that is o/p noise power P0 may be evaluated simply by integrating Sn0() over the band width under consideration, ) that is
For practical purpose, the input noise power spectrum density is taken as constant with frequency. Let this constant value be denoted by C. Hence, Sni () =C Thus the output noise power is given by And therefore Area under the curve | H () |2 The ideal band pass system having the characteristics. The area under this ideal rectangular characteristic is given by AxBN. 2 Here A=| H () | Equating the areas of the actual and ideal systems, we get

From above equation, the expression for equivalent noise bandwidth is given as That is BN=Area under the | H ( |2 curve () Maximum value of the | H ( |2 curve ()
Then, the output noise power may be obtained as Where C is the constant equal to the power spectrum density of noise voltage at the input of the system.Hence, equivalent noise band width may be defined as follows: Equivalent noise bandwidth may be defined as the Bandwidth of an ideal band pass system which produces some noise power as the actual system does. duces Likewise we can able to calculate the equivalent Bandwidth of Low pass filter. Available power: The available power of any source may be defined as the maximum power which can be drawn from the source. 9. Write short notes on signal signal-to-noise ratio? The ratio of signal power to the associated noise power is defined as signal to noise ratio. In other words, signal to noise ratio is defined as the ratio of signal power to the noise power at the same point in the system. he It is represented as (S/N) And S =Power spectrum density of signal voltage N Power spectrum density of noise voltage In an ideal noiseless system the input S/N ratio and output S/N ratio are identical. But for a practical system there will be some noise included in the inside of the circuit. So there must be change in input and output S/N noise level. In fact the comparison of S/N ratio at the input and the output of any two port network gives arison the noise indication of that network. 10. Define Narrow band noise and explain the representation of Narrow Band Noise in terms of In In-Phase and Quadrature Components and envelope and phase components.[Nov s components.[Nov-03] The receiver of a communication system usually includes some provision for preprocessing the received signal. The preprocessing may take the form of a narrow band filter whose band width is just large enough to pass the modulated component of the received signal essentially undistorted but not so large as to admit excessive noise through the receiver. The noise process appearing at the output of such a filter is called as Narrow band noise. And this noise is concentrated at mid frequencies such as fC. concentrated

To analyze the effects of narrow band noise on the performance of a communication system, we need a mathematical representation of it. Depending on the application of interest, there are two specific representations of narrow band noise. The narrow band noise is represented in terms of a pair of components called the In-phase and Quadrature components. phase The narrow band noise is defined in terms of two other components called the envelope and phase. Representation of Narrow Band noise in terms of In phase and Quadrature In-phase components: Considering a narrow band noise n(t) of band width 2B centered on frequency fC .And we may represent n(t) in the canonical(standard)form: n (t) = nI (t) cos (2 ct) nQ (t)sin(2fct) (2f Where nI (t) In-phase component phase nQ (t) Quadrature component Both nI (t) and nQ (t) are low pass signals. Except the mid band frequency fC,these two components are fully representative of the narrow band noise n(t).Given the narrow band noise n (t), we may extract its in phase and Quadrature components by using low pass in-phase filters. It is assumed that the two low pass filters used in this scheme are ideal, each having a band width equal to B. The in-phase and Quadrature components having the following properties: 1. The in-phase component nI (t) and nQ (t) of narrow band noise n (t) have zero mean. 2. If n (t) is Gaussian then nI (t) and nQ (t) are Gaussian 3. If n (t) is stationary then nI (t) and nQ(t) are jointly stationary 4. Both nI (t) and nQ (t) have same power spectral density, which is related to the Power spectral density SN (f) of the narrow band noise n (t). 5. nI (t) and nQ (t) having same variance as the narrow band noise n (t). 6. The cross spectral density of the nI (t) and nQ (t) is purely imaginary. 7. If narrow band noise n (t) is Gaussian with Zero mean and a power spectrum Density SN (t) is symmetric about the mid frequency fC, then the in-phase phase Component nQ (t) are statistically impendent. For example if we consider a white Gaussian noise of zero mean and power spectral density N0/2, which is passed through an ideal band pass filter of pass band magnitude response equal to one, mid band frequency fC, and band width 2B.And to determine the auto correlation function of n (t) and its in in-phase and
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Quadrature components we are taking the inverse Fourier transform of the power spectral density. And finally the auto correlation function of in in-phase and Quadrature components are RNI(t)=RNQ(t)=2N0Bsinc(2Bt) ponents Representation of Narrow Band noise in terms of Envelope and Phase components: We know that the Representation of a narrow band noise n (t) in terms of in in-phase and Quadrature components. We may also represent the noise n (t) in terms of its noise envelope and phase components. n (t) = r (t) cos [2fCt+y(t)] Where
The function r(t) envelope of n(t), y(t) phase of n(t) We may also represent that n (t) = nc (t) Coswct nS (t) Sinwct nc (t) In phase ns (t) Quadrature Component. The envelope r(t) and phase y(t) are both sample functions of low pass random processes. The time interval between two successive peaks of the envelope r(t) is app approximately 1/B,Where 2B is the band width of the narrow band noise n(t). The probability distributions of r(t) and y(t) may be obtained from those of nI(t) and nQ(t).Let NI and NQ denote the random variables obtained by observing the random processes repre represented by the sample functions nI(t) and nQ(t) respectively. We note that NI and NQ are independent Gaussian random variables of zero mean and variance s2,and so we may express their joint probability density function by
Accordingly the probability of the joint event that NI lies between nI and nI+d nI and NQ lies between nQ and nQ+d nQ is given by,
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By using the coordinate system representation of narrow band noise
In a limiting sense, we may equate the two incremental areas shown shaded in figure (a) and thus we may write
Now let R and y denote the random variables obtained by observing the random processes represented by the envelope r (t) and phase y(t) respectively. Substituting the equation (3),(4),(5) in equation(2),we find that the probability of uation the random variables R and y lying jointly inside the shaded area of the figure(b) is equal to
That is the joint probability density function of R and y is
This probability density function is independent of the angle ,which means that the function random variables R, are statistically independent. We may thus express fR, (r, ) as the product of fR(r) and fy().In particular, the random variable representing phase is uniformly distributed insi the range 0 to 2p,as shown by inside
This leaves the probability density function of the random variable R as
Where s2 is the variance of the original narrow band noise n(t).A random variable having the probability density function of equation(9) is said to beRayleigh be distribution. if we represent in graphical representation then . fV(v) =s fR (r) Then we may rewrite the ray Leigh distribution of equation(9)in the normalized form
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Unlike Gaussian distribution, ray Leigh distribution is zero for negative values of v.This is because the envelope r(t) can assume only non negative values.
11. Explain about sine wave plus narrow band noise? If we add the sinusoidal wave A Cos (2pfCt) to the narrow band noise n(t),where A and fC both constsnts.We assume that the frequency of the sinusoidal wave is the same as the nominal carrier frequency of the noise. A sample function of the sinusoidal wave plus noise is then expressed by x (t) = A Cos(2pfCt)+n(t)------(1) Where A is the carrier amplitude. Representing the narrow band noise n(t) in terms of its in phase and Quadrature in-phase components, we may write We assume that n(t) is Gaussian with zero mean and variance s2.accordingly we may state the following: 1. Both nI(t) and nQ(t) are Gaussian and statistically independent. 2. The mean of nI(t) is A and that nQ(t) is zero. 3. The variance of both nI(t)and nQ(t) is s2. We may therefore express the joint probability density function of the random variables NI and NQ,corresponding to nI(t) and nQ(t),as follows: Let r(t)denote the envelope of x(t) andy(t) denote its phase. from equation(2) we thus find that
For the derivation of the ray Leigh distribution, we find that the joint probability density function of the random variables R and y,corresponding to r(t) and y(t) for some fixed time t,is given by
Here R and y are dependent random variables for nonzero values of the amplitude A of the sinusoidal wave component. To determine the probability density function we have to integrate the equation (7) over all possible values of y obtaining the marginal density,

The integral in the right hand side of equation (8) can be identified in terms of the identified defining integral for the m modified Bessel function of the first kind of zero order that is, Thus,x=Ar/s2,and we may rewrite the equation(8) as,
This relation is called the Rician dist distribution.when we simplifying,
fV(v) =s fR (r) Then we may express the Rician distribution of equation (10) in the normalized form
When a is zero, the rician distribution reduces to the ray Leigh distribution. 12. Explain the characteristics of superhetrodyne receivers? [Nov-04,05][April-04] In a broadcasting system, irrespective of whether it is based on AM or FM, the receiver not only has the task of demodulating the incoming modulated signal, but it is also required to perform some other system functions. 1. Carrier frequency tuning to select desired signal 2. Filtering To separate the desired signal. 3. Amplification which is intended to compensate the loss of signal power during transmission. The super heterodyne Receiver is a special type of receiver which is particularly used in all radio and T.V receivers. The receiver consists of a RF section a mixer and local OSC, an intermediate frequency (IF) section, Demodulator and power amplif amplifier. and there is separate frequency ranges available for AM and FM

RF carrier range Midband frequency of IF section 0.455MHz IF B.W 10KHz

Antenna
AM Radio 0.535 1.605MHZ
FM Radio 88-108MHZ 108MHZ 10.7MHz 200KHz
If we take the Block diagram of a AM super heterodyne Receiver,
The incoming amplitude modulated wave is picked by the receiving antenna and amplified and tuned to the particular incoming carrier frequency by RF section. The combination of mixer and Local Oscillator provides a heterodyning function, where by the incoming signal is converted into predetermine fixed intermediate frequency, usually lower than the incoming carrier frequency. This frequency translation is achieved without disturbing the relation of the sidebands to the carrier Therefore fIF = fLO fRF FLO Local oscillation frequency, FRF Incoming carrier frequency from RF section. Normally the mixer Local oscillation is called as first detector and Demodulator is the 2nd detector. This IF section provides best selectivity in the receiver and this will be detected by an envelope detector and the message signal is amplified and given to Loud Speaker.
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The basic difference between AM and FM super heterodyne Receiver is in FM detectors we are using amplitude limiters and frequency discriminator because in FM we are making the amplitude as constant. So any change in amplitude will be limited by amplitude Limiter. The other functions are same.
13. Write short notes on Signal to noise ratio. SNR (Signal to Noise Ratio): Signal to noise ratio is nothing but signal power to the noise power. We know that in a communication system, the message signal travels from the transmitter to the receiver through a medium called channel. r Now, noise is present in every communication system, in fact the channel introduces an additive noise in the message signal and thus the message which is received at the receiver is distorted. Since the receivers receives both the message eceivers and noise signal. And the o/p contains the noise by the reproduction of message signal with noise. So we are calculating the signal power and noise power. And the noise calculation in common system is carried out in the form of figure of merit (g). The figure of merit g may be defined as the ratio of o/p signal to noise ratio to input signal to noise ratio to input signal to noise ratio of the receiver.
If g is high, noise will be low.
For calculating figure of merit we are taking the following assumptions as
i. Channel noise is always Gaussian that is independent on frequency that is constant. ii. Channel noise is always additive, that is addition of signal x(t) and noise n(t)

iii. The i/p to the Demodulator is a Band pass noise. e 14. Explain the noise in DSB DSB-SC receiver using synchronous or Coherent detection?[April detection?[April-04] (OR) Calculate the figure of merit g for a DSB-SC system?
We know that the DSB Sc system uses coherent (or) synchronous detection DSB-Sc scheme at the receiver and figure shows the block diagram of DSB DSB-SC System from noise point of view. and then we calculate the figure of merit as
S0o/p Signal power, N0 o/p Noise power at detector Sii/p signal power, Ni i/p Noise power at detector We know that we are detecting our original signal by synchronous detection method, where as it is multiplied with a local Oscillator frequency which is in same phase with the DSB DSB-Sc signal i/p carrier frequency.
In order to calculate the figure of merit in noise analysis we have to calculate the i/p and o/p signal power, i/p and o/p noise power.
i. Input Signal Power: W.k.T the modulated DSB SC signal is S(t) = x(t) Cos wct And the signal power is the mean square value of x (t) Hence Si=mean square value of S (t) (Neglecting higher frequency terms) ii. Input Noise power Ni: The noise at the input of the receiver is a Gaussian noise in nature and this noise is passed through the Band Pass filter.
Then the o/p of Band pass filter is converted as Band pass noise and the band pass noise is given as,
And the input noise power is calculated as,
Neglecting higher order terms and equating the inphase component and equating Quadrature component.
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iii. output signal power: W.k.T the DSB-SC signal at i/p is multiplied with synchronous carrier Coswct. SC Thus the signal at the multiplier output is So (t) = e (t) = x (t) Coswct. Coswct The signal is passed through Low Pass Filter are higher order terms is reduced.
The o/p signal power is calculated by mean square values. S0 = mean square value of S0(t) iv. Output noise power: Now we will find the o/p noise po power from detector. The i/p noise from narrow band filter is narrow band noise is multiplied with locally generated carrier. The o/p noise n0(t) is
When this signal is passed through Low pass filter, the higher order frequencies are neglected, then O/p noise power N0 will be N0 = mean square value of no (t)
15. Explain the noise in SSB-SC receiver using Coherent detection? (OR) Calculate the figure of merit g for a SSB-SC system? Noise in SSB SC system using coherent detection:

W.k.T the receiver of an SSB SC System is similar to that for DSB-Sc system SSB-SC DSB using coherent Detection.
The only difference between DSB SC and SSB Sc system is Bandwidth is reduced Sc halfely compare to DSB SC system. SC So the evaluation procedure of figure of merit is similar to that of the DSB SC system.
S0o/p Signal power, N0 o/p Noise power at detector Sii/p signal power, Ni i/p Noise power at detector i. Input signal power: W.k.T the incoming signal to the SSB SC receiver may be expressed as T SSB-SC S (t) SSB = x (t) Coswct xh (t) Sinwct And xh (t) Hilbert transform I.e. we know the o/p of SSB system is the single side band and carrier suppressed o/p i.e. o/p is Cos(wc m) t this means the upper sideband is suppressed. w (Or) o/p = Cos (wc+wm) t this means lower side band is suppressed. Therefore Cos (wc wm) t = CoswmCoswct + SinwmSinwct 1 Cos (wc +wm) t = CoswmCoswct - SinwmSinwct 2
When we are combining both the equations
S (t)SSB =CoswmCoswct SinwmSinwct
(+) represents lower sideband, ( indicates upper sideband. (-)
i.e. Sinwmt = Cos (wm /2) SinwCt = Cos (wCt /2)

/2 indicates the phase shift. /2 So the general expression is

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S (t) SSB = x (t) CoswCt xh (t) SinwCt Where x (t) = Coswmt, xh (t) = Phase shift of Coswmt and (+) signal corresponds to LSB and (-) corresponds to USB The I/p signal power Si = mean square value of S(t)
I/p noise power Ni: The noise power is same as DSB DSB-SC system. So according to the o/p of Band pass filter, the noise is represented as,
And the input noise power is calculated as,
Neglecting higher order terms and equating the inphase component and Quadrature component.
O/p signal power (S0): W.k.T the incoming SSB signal is multiplied by a synchronous local carrier Coswct. There fore O/p will be e (t) = S0 (t) =(x (t) Coswct xh (t) Sinwct).Coswct e (t) = S0 (t) =x (t) Cos2wct xh (t) Sinwct.Coswct O/p noise power No: W.k.T the o/p noise power is calculated as same as DSB DSB-SC.

The input band pass noise is multiplied with synchronous carrier signal Coswct. Therefore the multiplied signal nd (t) = ni (t). Coswct nd (t) = [nc (t) Coswct nS (t) Sinwct].Coswct nd (t) = n0 (t) =nc (t) Cos2wct nS (t) Sinwct.Coswct There fore the figure of merit of SSB SSB-SC is
From the figure of merit value of DSB DSB-SC and SSB-SC we may note that there is SC no improvement because DSB SC having better SNR but i/p noise power is twice DSB-SC that of SSB.And SSB having Less SNR compare to DSB SC.
UNIT-IV PERFORMANCE OF CW MODULATION SYSTEMS IV 1. Compare the noise performance of an AM and FM system? The figure of merit of AM system is 1/3 when the modulation is 100 percent and that of FM is (3/2) mf2.The use of FM offers improved noise performance over AM when (3/2)mf2 > 1/3.mf modulation index in FM. modulation 2. What is threshold effect in AM recei receivers? The loss of the message signal x (t) in an envelope detector due to the presence of the large noise is known as the Threshold effect.

The threshold effect is also defined as when a noise is large compared to the signal at the input of the envelope detector, the detected output has a message signal completely mingled with noise. it means that if the i/p signal to noise ratio(Si/Ni) is below a certain level called threshold level. ow
3. Write the figure of merit (g) for an AM system? (OR) What is the for AM (with envelope detection in small noise Case? [Nov-05]
4. Define Noise-Quieting effect? Noise power of FM detector varies inversely with carrier power. The decrease in noise power with an increase in carrier power is called noise quieting effect. noise-quieting 5. Write the figure of merit (g) for an FM system?
6. Define capture effect in FM? [Nov n [Nov-05] When the interference is stronger then it will suppress the desired FM input. When the interference signal and FM input are of equal strength, the receiver fluctuates back and forth between them .This phenomenon is known as the capture effec effect. We may also define as in low noise case; the distortion produced by the noise at the o/p of FM detector is negligible in comparison to the desired modulating signal. And noise almost suppressed by the signal. This phenomenon is called as capture effect. 7. Define FM threshold effect?[April effect?[April-04] As the input noise power is increased the carrier to noise ratio is decreased the receiver breaks and as the carrier to noise ratio is reduced further crackling sound is heard and the output SNR cannot be predicted by the equation. This predicted phenomenon is known as threshold effect. It is also defined as when the SNR becomes even slightly less than unity, an impulse of noise is generated. This noise impulse appears at the output of the detector in the form of click sound. If the SNR ratio is further decreased so that the sound. ratio is moderately less than unity, the impulses are generated rapidly and clicks merge in to spluttering sound. This phenomenon is known as threshold effect. 8. How is threshold reduction achieved in FM systems?[Nov-03,04]

Threshold reduction is achieved in FM system by using an FM demodulator with negative feedback or by using a phase locked loop demodulator. Such devices are referred to as extended-threshold demodulators.
9. What is Pre-emphasis? The pre modulation filtering in the transistor, to raise the power spectral density of the base band signal in its upper frequency range is called pre emphasis (or pre upper-frequency distortion) Pre emphasis is particularly effective in FM systems which are used for Transmission of audio signals. 10. Define de-emphasis. A de-emphasis in the receiver used to restore relative magnitude of different emphasis improvement in AF signal and to suppress noise is called de de-emphasis.
11. What is the figure of merit of a AM system with 100 percent Modulation? The figure of merit of an AM system with 100 percent modulation is 1/3.This means that other factors being equal an AM system must transmit three times as much average power as a suppressed system in order to achieve the same quality of noise erage performance. 12. What is Nyquist rate? The sampling rate of 2fm samples per second for a signal bandwidth of fmHertz is called the Nyquist rate. Its reciprocal 1 m is called the Nyquist interval. 1/f 13. Compare AM to FM
* In AM system there are three frequency components and hence the bandwidth is finite. *The amplitude of modulated wave in AM is dependent of modulation index. *In AM, most of the transmitted Power is wasted FM system has infinite number of sidebands in addition to a single carrier. Hence its Bandwidth is infinite. The amplitude of frequency modulated wave in FM is independent of modulation index. In FM, noise is very less.
14. Draw the phasor representation of FM noise?[Nov noise?[Nov-05] r (t) and q(t) are the amplitude and phase of the band pass signal noise. And y(t) is the relative phase.
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15. Draw the phasor representation of AM noise using envelope Detection?
DESCRIPTIVE ANSWERS: 1. 1. Discuss the Noise performance of an AM system using envelope Detection?[Nov-04,April 04,April-04] Noise Performance in Amplitude modulation systems in envelope detector method. We know that in amplitude modulation system, a large carrier is accompanied with the two upper and lower sidebands. For calculating the noise performance we have to take the input and output signal power and noise power and then we have to calculate the figure of merit by using the equation, S0o/p Signal power, N0 o/p Noise power. Sii/p signal power, Ni i/p Noise power. Input signal power Si The i/p signal with carrier and two sidebands is Vm (t) = VmCoswmt VC (t) = VCCoswCt VAM (t) = VC (1+Vm/VcCoswmt) Coswmt = VC CoswCt+VmCoswmt Take VC=A and vmcoswmt =x (t) S (t) AM = [A+x (t)] CoswCt. Therefore the input power is the mean square value of carrier and message component then
Input noise power Ni W.k.t in general especially in AM, FM the i/p noise Narrow Band noise is i/p assumed as white Gaussian noise. And we know the power spectral density of white noise, is

Ni=hfm Output power:
The o/p of envelope detector contains the detected o/p with noise, hence equation noise, will become
S (t) AM = [A+x (t)] CoswCt+ni (t) Where (Output contains Band pass filter) There fore S (t) AM = [A+x (t)] CoswCt + nC(t) CoswCt-nS (t) SinwCt S (t) AM = [A+x (t) + nC (t)] CoswCt -nS (t) SinwCt
If we take trigonometric expression then A(t),f(t) will be varying amplitude and phase and The time varying amplitude A(t) is the envelope of S(t)AM and therefore the output of the envelope detector will be envelope A(t). The envelope A (t) contains both signal and noise components. We are taking the analysis in two cases i. small noise case ii. Large noise case
Small noise case: In small noise case the noise is taken to be much smaller than signal. that is Ni (t) << [A+x (t)] If we draw the phasor representation of In phase of Quadrature component then

It is clear from the phasor diagram, that if nS (t) is much smaller then f(t) is also much smaller. So it may be assumed to be zer zero. So A (t) = A+x (t) +nC (t)
This equation implies the o/p of envelope detector contains a carrier component and useful information x (t) and noise component nC(t).
Large noise case:
In large noise case the noise is taken as ni(t) >> [A+x(t)] Under large noise case the noise is dominated mostly hence the performance of the envelope detector varies fully. That is
It is clear that a from the above equation w.k.t the o/p x (t) (modulating signal) is multiplied with large noise. So it carries no useful information. The loss of the message x (t) is an envelope detector due to the presence of the large noise is known as the threshold effect. The threshold effect is defied as when a noise is large compared to the signal compared at the i/p of the envelope detector, the detected o/p has a message signal completely mingled with noise. It means that if the i/p signal to noise ratio (Si/Ni) is below a certain level called threshold level.
O/p signal power (S0): (In small noise case) n


[Useful information is only in x (t). A and nC (t) are rejected Component. So o/p signal power is mean square value of s(t).] O/p noise power(N0):
The o/p noise signal n0(t) at the o/p of the dete detector is nC (t). so the power spectrum is
[Contains only nC (t) terms, so Sno (w) =Snc (w).so o/p power spectral density is h] There fore N0=h.2fm=2hfm Figure of merit (g):-
From this equation it is clear that the noise performance improves in reduction in carrier amplitude A. And w.k.t if figure of merit is increased noise will be reduced. So the greatest values of g that may be achieved depend on the minimum possible value of carrier amplitude A. When we use synchronous detection method rather than envelope detection method for AM demodulation the noise performance is identical in small noise case but it is very quite and complex compare to Envelope method.
Discuss the Noise pe performance of an FM system. (OR) Derive the noise power spectral density of the FM demodulator And explain its performance with diagram.[Nov diagram.[Nov-03] We have discussed the noise performance of amplitude modulated systems. In a similar manner, we will discuss the noise performance of angle modulated systems, i.e. FM systems.
1. 2.
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An FM detector consists of a frequency discriminator followed by envelope detector. And the o/p signal at the detector is x(t).
Let the modulating signal Vm (t) = VmCoswmt VC (t) = VCCos (wC t+q) = VCCosf = ACosf Where f =wC t+q And df/dt = wC We know that the final output Where Vc Amplitude of carrier i Instantaneous phase angle. And it is calculated from Where There fore
Input signal power (Si)
Output signal power (S0):
The o/p signal is dependent on insta instantaneous frequency wi. So
In which useful information is only in x (t) not in carrier. So S0 = Mean square Value of S0 (t)
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Noise power: Input noise power (Ni) W.k.t the power spectrum density of a white Gaussian noise
Output noise power (N0) The output noise power may be computed assuming the message signal term x (t) to be zero. I.e. assume the signal at the detector gi (t) = ACoswCt+ni(t) [ The i/p signal to the detector will have a carrier + noise term The message is inside the carrier]
There fore gi (t) = ACoswCt+nC(t) CoswCt nS(t) SinwCt tgi (t) = [A+nC(t)] CoswCt - nS(t) SinwCt Writing in trigonometric form
In which we are not considering about amplitude parameter A(t), we are considering only frequency component q(t). And it can also written as gi(t) = A(t) Cos(wC t+ q(t)) gi(t) = A (t) Cosf(t) There fore f(t)=wC(t)+q(t) Now we will consider two noise case. That is whether the noise signal is low or high compare to carrier amplitude, in q(t)
i.e. 1) A>>nC (t) and nS (t) (Small noise case) 2) A<< nC (t) and nS (t) (large noise case) i) Small noise case If A>>nC (t) and nS (t) then nC (t) can be eliminated

Therefore For small angleq, tanq = q then
W.k.t the o/p noise power also depends on instantaneous frequency (wi) So nd (t) wi And nd (t) =K wi We have to calculate wi from q(t).and we know that
Sno (w) =SnS (w) (Or) Sno (f) =SnS (f)
Now to determine average o/p noise power the noise at discriminator nd (t) o/p is proportional to the derivative of nS (t). so nd(t) is calculated by passing nS(t) through differentiator and w.k.t the property of Fourier transform that the differentiation w.r.t time tcorresponds to multiplication of Fourier transform by j2f.
Therefore And due to Gaussian noise SnS (f) = h Therefore Sn0 (f) = And Sn0 (f) =
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Therefore
Figure of merit g for FM: Noise quieting effect: Equation (1) states that noise power of FM detector varies inversely with carrier power. The decrease in noise power with an increase in carrier power is called noise quieting effect
Figure of merit by considering single tone FM signal If we are taking x (t) = Vmcoswmt Then
3. Compare the Noise performance of FM with AM W.k.t


4. Write short notes on Capture effect and FM threshold effect? Capture effect: The phase deviation produced by the noise signal in FM is much smaller than the phase deviation produced by the modulating signal provided that noise is smaller than carrier. Hence in low noise case, the distortion produced by noise at the output of FM detector is negligible in comparison to the desired modulating signal. And noise is almost suppressed by the signal. This phenomena is called capture effect
Whenever FM signals from two transmitters operated on the same or nearly same henever carrier freq. The signal of a weak magnitude is suppressed by the strong signal. The weak signal is noise.
FM Threshold effect:
The output signal to noise ratio of an FM receiver is valid only if the carrier to receiver noise ratio, measured at the discriminator input, is high compared with unity. The input nose power is reduced by increasing carrier noise - ratio. If the carrier carrier noise- ratio is less, then it will produce a crackling sound. This phenomenon is known as the threshold effect. For a qualitative discussion of the FM threshold effect, consider the first case when there is no signal present, so that the carrier wave is unmodulated. Then the frequency discriminator input will be or x (t) = (A+nC (t))CoswCt nS(t) SinwCt If we take the phasor diagram if carrier to noise ratio is large, nC (t) and nS (t) is smaller than carrier amplitude. If it is small, then nC (t) and nS(t) will be large. W.k.t the o/p signal is taken as the receiver o/p measured in the absence of noise. The carrier to noise ratio is calculated when there is unmodulated carrier signal. It is represented as r Practically the threshold effect may be avoided when carrier to noise ratio is more avoided than 20dB Phasor representation of FM: FM:Page 112

5. Write short notes on FM threshold reduction? (OR) How Threshold reduction is achieved in FM systems?
In communication systems for reducing the threshold level we are using FMFB technique. (Frequency modulated feedback loop technique). This contains a VCO (Voltage controlled oscillator).
From the above diagram we may note that a conventional FM Receives Local Oscillator may be replaced with a VCO. W.K.T The operation of the block diagram. Here we are providing a negative feed back from output, because this process provides a high signal noise-ratio, because signal-noise Local Oscillator frequency is take from demodulated output. taken
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So it provides high SNR at the input side for that we are using FMFB technique for threshold reduction.
6. Explain Pre-emphasis and De emphasis De-emphasis in FM. [April-04] [Nov-05] (OR) Explain the significance of Pre he Pre-emphasis and De-emphasis in FM Systems. [Nov-03] The noise Power density at the demodulated output rises parabolically with increase in frequency.
I.e. The power spectral density of the message usually falls off at the higher frequencies.
The Power spectral density of the noise is usually increases with frequency. Therefore the high frequency components of the message signal suffers. In order to avoid this we are using pre pre-emphasis and De-emphasis circuit. emphasis We are increasing the magnitude of high frequency components of the modulating signal by pre-emphasis circuit. Hence the signal to noise ratio is increased. As a emphasis De-emphasis circuit is used to restore the original information. t
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In this process the high frequency components of the noise at the discriminator Output are also reduced. Therefore the o/p signal to noise ratio is increased. This is used in commercial FM Radio transmission and reception. In order to produce an undistorted version of the original message at the receiver output, the pre-emphasis filter in the transmitter and de emphasis filter in the emphasis de-emphasis receiver must ideally have frequency responses that are the inverse of each other. inverse That is ,if HPe(f) designates the frequency response of the pre emphasis filter, then pre-emphasis the frequency response Hde(f) of the de-emphasis filter must ideally be(ignoring emphasis transmission delay) Hde (f) =
Hde (f) Frequeny response of De De-emphasis filter HPe (f) frequeny response of pre emphasis filter We are taking a simple RC filter as pre and De emphasis filters.
The HPe (f) = assuming R<<r And the Hde (f) = From the Simple RC network network.
But generally from our noise analysis in FM systems, assuming high carrier to noise ratio the power spectrum density is calculated as | Hde (f) |2 Snd (f) =
Therefore the average o/p noise power is N0 = Because of pre-emphasis and de emphasis the output signal to noise ratio emphasis de-emphasis improvement factor is
There fore

So by using a simple pre pre-emphasis and de-emphasis circuit we can provide emphasis significant improvement in noise performance of the receiver. The use of simple linear and non non-linear pre-emphasis and de-emphasis filters is emphasis used to calculate the differences between the signals and noise in the system. These simple filters are also used in auto tape recording. Specially non-linear preSpecially non emphasis and de-emphasis filters have been applied successfully to tape recording. emphasis These techniques use a combination of filtering and dynamic range compression to reduce the effects of noise, particularly when the signal level is low.
7. Compare the performance of AM and FM systems? [Nov [Nov-05]

S.NO Amplitude modulation Amplitude of the carrier is varied according 1 to amplitude of modulating signal Am has poor fidelity due to narrow band 2 width Most of the power is in carrier hence less 3 efficient 4 Noise interference is more 5 6 7 8 9 10 Frequency modulation Frequency of the carrier is varies according to amplitude of the modulating signal Since the band width is large, fidelity is better All the transmitted power is useful.
Noise interference is minimum. Adjacent channel interference is avoided due to Adjacent channel interference is present guard bands. AM broad cast operates in MF and HF range. FM broadcast operates in VHF and UHF range. In AM only carrier and two side bands are Infinite number of sidebands are present. present. The transmission equipment is simple. The transmission equipment is complex. Transmitted power varies according to Transmitted power remains constant irrespective of modulation index. modulation index. Depth of modulation has limitations. It can Depth of modulation has no limitation. It can be be increased above 1. increased by increasing frequency deviation.
8. Write the advantages and disadvantages of FM over AM. FM has the following advantages over AM.
The amplitude of FM is constant. It is independent of depth of modulation. Hence transmitter power remains constant in FM whereas it varies in AM.
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Since amplitude of FM is constant, the noise interference is minimum in FM.Any noise superimposing an amplitude can be removed with the help of amplitude limits. Whereas it is difficult to remove amplitude variations due to noise in AM. The depth of modulation have limitation in AM. But in FM the depth of lation modulation can be increased to any value by increasing the deviation. This does not cause any distortion in FM signal. Since guard bands are provided in FM, there is less possibility of adjacent channe channel interference. Since space waves are used for FM, the radius of propagation is limited to line of sight. Hence it is possible to operate several independent transmitters on same frequency with minimum interference. A special case of narrow band FM (or) PM where mf is less than one has a limited number of side band pairs. Hence FM and PM is similar to AM except that it has superior noise behaviour than AM. Since FM uses UHF and VHF ranges the noise interference is minimum compared to AM which uses MF and HF ranges.
There are some disadvantages of FM compared to AM as follows: follows: The band width requirement of FM is much higher than that of AM.

The FM transmitting and receiving equipment is more complex and costly. Since FM uses UHF and VHF range of frequencies, its area of reception is limited only to line of sight. This is much lower than area covered by AM.
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UNIT INFORMATION THEORY UNIT-V 1. 1. A Source is emitting symbols x1 , x2 and x3 with probabilities, respectively 0.6 , 0.3 , and 0.1.What is the entropy of the source?[Nov source?[Nov-03] Let p1=0.6, p2=0.3, p3=0.1 1. 2. What is the channel capacity of binary symmetric channel with error probability of 0.2?[Nov 0.2?[Nov-03]
W.K.T P=0.2, Hence 1 1-P=0.8 Channel capacity for binary symmetric channel C=1+Plog2P+ (1-P) log2 (1-P) P) =1+ 0.2log20.2+ 0.8 log2 0.8 =1-0.4644-0.2575 0.2575 C=0.278 bits/message. 1. 3. What is entropy?[ entropy?[April=04] The average information per source symbol in a particular interval is called entropy. It is denoted by H and given as, Here P is the probability of occurrence of Kth message. 1. 4. What is discrete memory less source? The symbols emitted by the source during successive signaling intervals are statistically independent. That source is called discrete memory less source. Here memoryless, means that the symbol emitted any time is independent of previous choices. 1. 5. What is amount of information?
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The amount of information gained after observing the event S=SK, which occurs with probability PK, as the logarithmic function. Amount of information I(SK)=Log(1/PK) Unit of information is bit.
1. 6. What is mean by one bit? s One bit is the amount of information that use we gain, when one of two possible and equally likely (equal (equal-probability) events occurs. 1. 7. What is information rate? Information rate Ris represented as the average number of bits of information per second. R=r.H(x)information bits/second Where H(x)is entropy r is rate at which messages are generated. 1. 8. What is meant by Source encoding? The efficient representation of data generated by a discrete source. This pro process is called Source coding. The device that performs the representation is called a source encoder. 1. 9. Write Source encoding theorem? Given a discrete memoryless source of entropy H(x),the average code word length for any distortion less source encoding is bounded as H(x) Here the entropy H(x) represents the fundamental limit on the average number of bits per source symbol (). 1. 10. Name the two source coding techniques.[Nov techniques.[Nov-04] Prefix coding or instantaneous coding Shannon fano coding Huffman coding. 1. 11. When is the average information delivered by a source of alphabet size 2 maximum?[Nov-04] 04] The average information is maximum when the alphabets are equally probable.i.e p1=p2= 1. 12. What is meant by prefix code?[Dec code?[Dec-03] Prefix coding is variable length coding algorithm. It assigns binary digits to the messages as per their probabilities of occurance.Prefix of the codeword means any

sequence which is initial part of the code word. A prefix code is defined as a code in which no code word is the prefix of any other code word. 1. 13. Write about data compaction? For efficient signal transmission, the redundant information should be removed from the signal prior to transmission. This operation with no loss of information is operation ordinarily performed on a signal in digital form. This refers to Data compaction (or)Lossless data. 1. 14. What is channel redundancy?[April redundancy?[April-04] Redundancy is given as, Redundancy (g) =1-Code efficiency Code Redundancy (g) =1-h The redundancy should be as low as possible. 15. An event having six possible outcomes with probabilities Find entropy of the system. [April [April-05]
16. What is mutual information? The mutual information is defined as the amount of information transferred when Xi is transmitted and YJ is received. It is represented by I(Xi , YJ) and given as, 17. Write about channel capacity? The channel capacity of the discrete memoryless channel is given as maximum average mutual information. The maximization is taken with respect to input verage probabilities P(xi) C=max I(x; y) {P (xi)} 18. Define lossless channel. The channel described by a channel matrix with only one nonzero element in each matrix column is called a lossless channel. In the lossless channel no sources information is lost in transmission. 19. Define Deterministic channel A channel described by a channel matrix with only one nonzero element in each nonzero row is called a deterministic channel and this element must be unity. 20. Define noiseless channel.

A channel is called noiseless if it is both lossless and deterministic. The channel matrix has only one element in each row and in each column and this element is unity. The input and output alphabets are of the same size. 21. Explain Shannon-Fano coding. Fano An efficient code can be obtained by the following simple procedure, known as Shannon- Fano algorithm. 1. List the source symbols in order of decreasing probability. 2. Partition the set into two sets that are as close to equiprobable as possible, and sign 0 to the upper set and 1 to the lower set. 3. Continue this process, each time partitioning the sets with as nearly equal probabilities with as possible until further partitioning is not possible. 22. State the properties of mutual information. 1. I(X;Y)=I(Y;X) 2. I(X,Y)0 3. I(X;Y)=H(Y)-H(Y/X) 4. I(X; Y) =H(X) +H(Y) +H(Y)-H(X; Y). 23. Give the relation between the different entropies. n H(X; Y) =H(X) +H(Y/X) =H(Y) +H(X/Y) H(X) - entropy of the source(Y/X), H(X/Y) H(X/Y)-conditional entropy H(Y)-entropy of destination H(X, Y) - Joint entropy of the source and destination. 24. What is source coding and entropy coding? A conversion of the output of a DMS into a sequence of binary symbols is called source coding. The design of a variable length code such that its average code word length approaches the entropy of the DMS is often referred to as entropy coding. 25. What is information theory? Information theory deals with the mathematical modeling and analysis of a communication system rather than with physical sources and physical chann channels. 26. What is the channel capacity of a BSC and BEC? For BSC the channel capacity C=1+plog2 p + (1-p) log2 (1-p). For BEC the channel capacity C= (1 (1-p) 27. What happens when the number of coding alphabet increases? When the number of coding alphabet increases the efficiency of the coding technique decreases. 28. What is channel diagram and channel matrix?

The transition probability diagram of the channel is called the channel diagram and its matrix representation is called the channel matrix. 29. Write the expression for code efficiency. [Nov-05] 30. Define band width efficiency.
31. Is the trans information of a continuous system non non-negative? If So, why? [Nov-05] The mutual information is always positive.i.e, I(X; Y) 0 32. State Shannon Hartley theorem. (Or)Channel capacity theorem For a continuous channel. The capacity C of a additive Gaussian noise channel is B= channel bandwidth, S/N=signal to noise ratio. But N=N0B There fore
33. What is the channel capacity of the channel having infinite Bandwidth? The channel capacity of such channel is given as, Here is the signal to noise ratio of Gaussian channel. 34. Define rate of information transmission across the channel. te Rate of information transmission across the channel is given as, Dt= [H(X)-H(X/Y)] r bits/sec Here H(X) is the entropy of the source. H(X/Y) is the conditional entropy. 35. State the channel coding theorem for a discrete memoryless Channel. [Dec-03] Given a source of Mequally likely messages, with M>>1,which is generating information at a rate R.Given channel with capacity C.Then if, R C There exists a coding technique such that the output of the source may be ng transmitted over the channel with a probability of error in the received message which may be made arbitrarily small.

36. A discrete memoryless source is capable of transmitting three Distinct symbols m0, m1 and m2.Their probabilities are Respectively. Calculate the source entropy. Given that 37. Prove that the mutual information of a channel is symmetric. That means I(X; Y) =I(Y; X) [May [May-05] Let us consider some standard relationships from probability theory. These are as follows P (xi ,yi) = P (xi /yi) P (yi) ------------ (1) And P (xi ,yi) = P (yi / xi) P (xi) -------------- (2) From equation (1) and (2) we can write, P (xi /yi) P (yi) = P (yi / xi) P (xi) ------------ (3) There fore The average mutual information is given by Hence we can write I(Y; X) as follows. By considering equation (1) the above equation is written as,
Thus the mutual information of the discrete memoryless channel is symmetric. 38. Prove that the following, I(X; Y) =H(X) +H(Y)-H(X, Y) H(X, We know the relation H(X, Y) =H(X/Y) +H(Y) There fore H(X/Y) =H(X, Y) -H(Y) --------- (1) Mutual information is given by I(X; Y) =H(X) - H(X/ Y) --------- (2) Substituting equation (1) in (2) I(X; Y) =H(X) + H(Y) - H(X/ Y) Thus the required relation is proved. 39. For an AWGN channel with 4 KHz band width and noise power Spectral density ,the signal power required at the Receiver is 0.1 mW.Calculate capacity of this channel.

The equation of the channel capacity Bits/sec.
B= 4000 Hz S=0.1x10-3 W And noise power can be obtained as, But N=N0B=10-12x2x4000=8x10-9 W DESCRIPTIVE ANSWERS: 1. What is uncertainity? Explain the difference between uncertainity s And information? The words uncertainity, surprise and information are all related to each other. Before an event occurs, there is an uncertainity, when the event occurs there is an amount of surprise and after the occurrence of an event there is a gain of information. Consider the source which emits the discrete symbols randomly from the set of fixed alphabet i.e. X={x0, x1,x2,. xK-1} The various symbols in Xhave probabilities of p0, p1, p2etc, which can be written as, P(X=xK) =PK K=0, 1, 2 K-1 The set of probabilities satisfy the following condition,
The idea of information is related to UncertainityorSurprise.Considering the emission of symbol X=xK from the source. If the probability of xK is PK=0, then such a symbol is impossible. Similarly when probability PK=1, then such symbol is sure. In both cases there is no Surpriceand hence no information is produced when symbol xK is emitted. As the probability PK is low, there is more surprise or uncertainity.Before the event X=xK is emitted, there is an amount of uncertainity. When the symbol X=xK occurs, there is an amount of surprise. After the A occurrence of the symbol X=xK there is the gain in amount of information. The essence of which may be viewed as the resolution of uncertainity.
2. Write notes on information (or) measure of information (or) Amount of information and explain their properties. Let us consider the communication system which transmits messages

m1,m2, m3 with probabilities of occurrence p1, p2, p3,. The amount of Information transmitted through the message mK with probability pK is given as, Amount of information: PK=Probability of occurrence of mK.The unit of information is bits. Properties of information Property 1:If there is more uncertainity about the message, information carried is also more. Property 2:If receiver knows the message being transmitted, the amount of information carried is zero. Proof:-Here it is stated that receiverknowsthe message. This means only one Here message is transmitted. Hence probability of occurrence of this message will be PK=1.This is because only one message and its occurrence is certain (probability of certain event is 1).The amount of information carried by this type of message is, Taking PK=1 IK=0 bits This proves the statement that if receiver knows message, the amount of information carried is zero. As PK is decreased from 1 to 0,IK increases monotonically from 0 to infinity. This shows that amount of information conveyed is greater when receiver correctly identifies less likely mes messages. Property 3:If I1 is the information carried by message m1, and I2 is the information carried by m2, then amount of information carried componently due to m1and m2 is I1+ I2. The definition of amount is, The individual amounts carried by messages m1 and m2 are, he Hence P1 is the probability of message m1 and P2 is the probability of message m2. Since messages m1,m2 are independent, the probability of composite message is p1p2. Therefore information carried compositely due to m1,m2 is,
Property 4:If there are M=2N equally likely messages, then amount of information carried by each message will be N bits. Since all the messages are equally likely and independent, probability of occurrence of each message will be 1/M. The definition of amount is,

Here the probability of each message is,.Hence the above equation will be, .We know that M=2N,hence equation will be, Thus the amount of information carried by each message will be Nbits. 3. (i) Calculate the amount of information if PK=. (ii)Calculate the amount of information if binary digits (binits) Occur with equal likelihood in binary PCM. Solution:(i)Given that the PK= We know that amount of information (ii)We know that in binary PCM, there are only two binary levels, i.e. 1 or 0.Since they occur with equal likelihood, their probabilities of occurrence will be, P1(0level) = P2(1level) = Hence the amount of information carried will be given by 4. Define Entropy. Explain the properties of entropy with a Suitable Example and Explain the entropy of binary memoryless Source. And write short notes on information rate. [April [April-04] Definition The average information per source symbol is called entropy.
Properties of entropy Property 1:Entropy is zero if the event is sure or it is impossible.i.e, H=0 if PK=0 or 1. Proof: We know that Since PK=1, the above equation will be When PK=1 Thus entropy is zero for both certain and most rare message.

Property 2:When PK=1/M for all the Msymbols, then the symbols are equally Likely. For such source entropy is given as H=log2M. Proof: We know that for Mnumber of equally likely messages, probability is, This probability is same for all Mmessages.i.e, ty And entropy is Hence substituting the value of P
Hence adding these terms above equation becomes,
Entropy for binary memoryless source Consider a binary source transmits two independent messages with probabilities of P and (1-P) respectively. Prove that the entropy is maximum when both the messages are P) equally likely. Proof:We know that For two messages above equation will be, Here we have two messages with probabilities P1=P and P2=1-P.Then above equation P.Then becomes, Information rate:If the time rate at which source X emits symbols is r(symbol S),the information rate R of the source is given by R=rH bits/second. R-is the information rate. is H-is the entropy or average information is r-is the rate at which messages are generated. is 5. A discrete memoryless source (DMS) X symbols x1,x2,x3,x4 with Probabilities P (xi) = 0.4, P(x2) = 0.3, P(x3) = 0.2, P(x4) = 0.1 i. calculate H(x)
has
four
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ii. find the amount of information contained in the message x1,x2,x1,x3 and x4,x3,x3,x2,and compare with the H(x) obtained in part(i) Given that P (xi) =P1= 0.4 P(x2) = P2=0.3 P(x3) = P3=0.2 P(x4) = P4=0.1 We know that (i) (ii) I(x1, x2, x1, x3) = I(x1) +I( x2) +I( x1 )+I(x3) We know that amount of information I(x1, x2, x1, x3) = 6.703 bits/symbol Thus I(x1, x2, x1, x3) < 7.4 [= 4H] bits/symbol (r = 4 because there are four outcomes At the particular time) I(x4,x3,x3,x2) = I(x4) +I( x3) +I( x3) +I(x2) Substituting the values I(x4,x3,x3,x2) =9.70 bits/symbol > 7.4 [=4H]bits/symbol. 6. A high resolution black and white TV picture consists of about 2x106picture elements and 16 different brightness levels. Pictures are repeated at the rate of 32 per second. Al the picture elements are assumed to be independent and all level have equal likelihood of occurrence. Calculate the average rate of information conveyed by this TV picture source. onveyed P (xi) =1/16 =4 bits/element And r=2 x 106(32) =64 x 106 elements/second R=rH=256 x 106 bits/second=256 Mb/second
7. The probabilities of the five possible outcomes of an experiment are given as P(x1) =1/2, P(x2) =1/4, P(x3) =1/8, P(x4) =P(x5) =1/16. Determine the entropy and information rate if there are 16 outcomes per second. The entropy of the system is given as,
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r= 16 outcomes/ second. Therefore R= rH= 16x1.81 =29 bits/second. 8. An analog signal band limited to 10 KHz is quantized in 8 levels of A PCM system with problem of 1/4, 1/5, 1/5, 1/10, 1/10, 1/20, 1/20 And 1/20 respectively. Find the Entropy and rate of information. We know that according to sampling theorem
The signal is band limited and sampled. So r = 2 fm Therefore R = rH= 20x103x2.84=56800 bits/sec 9. Write short notes on discrete memoryless channels, and lossless Channel, deterministic channel, noiseless channel, Binary Symmetric channel. Discrete Memory less source and Discrete Memory less channels: A discrete memory less source is memory less if output depends on the current input symbol only not on previous one. A channel is a transmission medium of command systems. The channel may be continuous channel, discrete channel. If the source of information is continuous then we are using continuous channels. A discrete memory less channel is a statistical model with an input X and an Output Y the channel is discrete when X and Y are both finite. The above diagram consists of m inputs and n outputs have been illustrated in figure.

The input X consists of input symbols x1, x2.xm. Each possible i/p to o/p path is indicated along with a conditional probability (or) transition probability P (yi/xi) and it is called as channel transition probability.
The channel is completely specified by the channel matrix
P(Y/X) is called channel matrix. Properties: 1. Each input channel results in some output, each row of channel matrix must be

Unity i.e. For all i. 2. If input probabilities are represented by row matrix then P[x] = [P(x1).P(x2).P (xm)] And if o/p probabilities are represented by row matrix P[Y] = [P (y1).P (y2).P (yn)] P[Y] = [P(x).P(y/x)] 3. If P(x) is represented as a diagonal matrix, then we have Then P[x, y] = [P(x)]d [P(y/x)] Lossless channel: A channel described by channel matrix with only one non zero element in each non-zero column is called lossless channel. E.g.: for lossless channel is
In loss less channel no source of information is lost in transmission
Deterministic channel: A channel described by a channel matrix with only one non zero element in each non-zero row is called deterministic channels.
Noiseless channel:
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A channel is called noiseless if it is both loseless and deterministic. And it contains only one element in each row and in each column. That is m = n for the noise less channel.
Binary symmetric channel (BSC) The binary symmetric channel (BSC) is defined by channel diagram shown in figure and its channel matrix is given by
The channel has two inputs and two o/p. This channel is symmetric because the probability of receiving a 1 if a 0 is sent is same as the probability of receiving a 0 if a 1 is sent. This is denoted by P.
10. Consider a binary channel shown in figure.
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i. find the channel matrix of the channel ii. Find P (y1) and P (y2) when P(x1) = P(x2) = 0.5 iii.find the joint probability P(x1,y2) and P(x2,y1) When P(x1) = P(x2) = 0.5 W.K.T the channel matrix is given by
ii).W.K.T P(x1) = P(x2) = 0.5 We have to calculate P (y1) and P (y2) W.K.T P(y) = P(x). P(y/x) And given P(x1,x2), there fore [P(y1), P(y2)] = [P(x1), P(x2)]
iii). P[x, y] = P(y/x)
11. Define mutual information and write the properties of mutual Information? Mutual information: Mutual information is the measure of uncertainity about the channel input that is resol resolved by observing the channel output. It is represented as, I(x; y) = H(x) H(x/y) b/symbol H(x) uncertainity about the channel input before the channel o/p is observed that is entropy of channel i/p H(x/y) uncertainity about the channel input after the channel o/p is observed that is information lost in a channel when Reception is confirmed. Properties of mutual Information: Property:1 I(x; y) = I(y; x) mutual information of a channel is symmetric. Proof:Let us consider some standard relationships from probability theory. These are as follows ider P (xi / yJ) P (yJ) = P (yJ / xi) P (xi) There fore

The average mutual information is given by Hence we can also write I(X; Y) as The above equation can be written as, Thus the mutual information of the discrete memoryless channel is symmetric. Property:2 I(x; y) 0 mutual information is always non negative. Proof:Mutual information is given by equation as But from the equation we have already derived mutual information is equal to information Putting the above value in the above equation
Next a direct application of the fundamental inequality yields the desired result I(x; y) 0 With equality if, and only if,
For all j and k Property:3 The mutual information of a channel is related to the joint entropy of the channel input and channel output by I(x; y) =H(x) +H(y)-H(x, y) Proof:We are having the relation as H(x, y) =H(x/y) +H(y) Therefore H(x/y) =H(x, y)-H(y) And also we know that mutual information is given by Mutual information is given by equation as utual I(x; y) =H(x)-H(x/y) Putting the value of H(x/y) in the above equation then I(x; y) =H(x) + H(y) - H(x, y) 12. Verify the following expression. H(x, y) =H(x/y) +H(y)

We know that And
13. Define channel capacity and write the channel capacity for Different special channels. [Nov/Dec [Nov/Dec-03, 04] Channel capacity Channel capacity of a discrete memory less channel as the maximum mutual information I(x; y) in the signaling interval. Therefore the channel capacity C is given by ore C = bits/symbol Where the maximization is over all possible input probability distributions {P (xi)} on X.Note that the channel capacity C is a function of only the channel transition probabilities which define the c channel. C = bits/symbol The transmission efficiency of the channel is defined as h= h= The redundancy of the channel is R = 1-h R= Channel capacities for special channels: Lossless channel: For a lossless channel H(x/y) = 0 and I(x; y) = H(x) Thus the mutual information (information transform) is equal to the input entropy and no source of information lost during transmission. Channel capacity C= C= log2 m

Deterministic channel: Be a deterministic channel H(y/x) =0 for all input distributions p (xi) and I(x; y) = H(y) This information transfer is equal to output entropy C= C= log2 n Noiseless channel: Since a noiseless channel is both lossless and deterministic, we have, I(x; y) = H(x) = H(y) Therefore channel capacity per sym symbol is C = log 2 m = log 2 n Binary symmetric channel For a BSC channel the mutual information is I(x; y) = H(y) +Plog 2P + (1-P) log 2 (1- P) P) And the channel capacity per symbol is C=1+ Plog 2P + (1-P) log 2 (1- P) H(y) max. Value of H(y) and is 1.In binary bits n=2 therefore log22 = 1 Binary Erasure channel (BEC): The binary erasure channel has two I/p and three o/ps where n is noise (0,n,1)
Channel matrix P(y/x) = Mutual information I(x, y) = H(x) H(x/y) Therefore I(x, y) = H(x) (1-p) H(x) = P H(x) p)

Channel capacity C= C=P 14. Write notes on binary symmetric channel, discrete channel Capacity, continuous channel capacity and channel capacity Theorem. [April-04] Binary symmetric channel (BSC) The binary symmetric channel (BSC) is defined by channel diagram shown in figure and its channel matrix is given by
The channel has two inputs and two o/p. This channel is symmetric because the probability of receiving a 1 if a 0 is sent is same as the probability of receiving a 0 if a 1 is sent. This is denoted by P. Discrete channel capacity: Channel capacity of a discrete memory less channel as the maximum mu mutual information I(x; y) in the signaling interval.
Therefore the channel capacity C is given by C = bits/symbol Where the maximization is over all possible input probability distributions {P (xi)} on X.Note that the channel capacity C is a function of only the channel transition probabilities which define the channel. C = bits/symbol If r symbols are being transmitted per second, then the maximum rate of transmission of information per second is rC.This is the channel capacity per second. Channel capacity per second=r.C b/s pacity Continues channel capacity:

A number of communication systems uses the continues sources and they uses channel continuously i.e. AM, FM, PM. Consider a continuous random variable X described by its probability density function fX(x). The average amount of information Per sample value of x(t) is measured by
The entropy H(x) defined as the differential entropy of x. The average Mutual Information is a continuous channel is defined as, Information
I(x; y) = H(x)-H(x/y) I(x; y) = H(y)-H(y/x) And
And
If we are having a random variable x uniformly distributed over the interval (0,a)
The probability density function of x is
If we are applying this in H(x) then
Thus this example shows that unlike a discrete random variable the differential entropy of a continuous random variable can be negative hence the Transinformation of a continuous system is non non-negative value.
Channel capacity theorem (or) Shannon Hartley theorem: ty
Considering an arbitary pair of random variables x and y , whose probability density function are respectively denoted by fx(x) and fy(y), if we are considering a Gaussian channel,
And

But
If we are considering the additive white Gaussian channel the channel o/p y is given by Y=X+n X channel I/p and n additive band limited white Gaussian noise
Then if we are calculating the channel capacity then for calculating this the signal is band limited and the samples are taken 2B per second. Hence rate of information will be, R(X) = rH Therefore R(X) =

If fx(x) is a band limited white Gaussian noise with average power N, then
R (n) = Blog2(2peN) Wheres2=N And R(Y) = Blog2[2pe(S+N)] because o/p contains signal and noise. The channel capacity C= Max {I(x, y)} I.e. C = Max (R(y) R (n)} Therefore C = B log2[1+S/N] bits/sec. Here B-is the channel bandwidth is S-is the signal power N-is the noise power within the channel bandwidth. is We know that the signal power is given as, Power P=Here B is the band width. And power spectral density of white noise is .Hence noise power N becomes, Noise Power N= There fore N=N0B And And substituting N value Extra question:CHENNAI INSTITUTE OF TECHNOLOGY HENNAI Page 138

1. Derive the channel capacity for a continuous band limited white Gaussian noise channel. [Nov-03, 04, 05, April 03, April-05] Answer:-Write channel capacity theorem (or) Shannon Hartley theorem. Write 15. State and explain source coding theorem (Shannons first Theorem). What is coding efficiency? An important problem in communications is the efficient representation of data generated by a discrete source. The process by which this representation is accomplished is called source encoding. The device that performs the ccomplished representation is called a source encoder. For the source encoder to be efficient, we require knowledge of the statistics of the source. In particular, if some source symbols are known to be more probable than others, then we may exploit this others Coding: A conversion of the output of a DMS into a sequence of binary symbols (ie binary code word) is called source coding. The device that performs this conversion is called the source encoder. Mainly we are going to encoding Cu Decoding is to minimize the average bit rate required for representation of the source by reducing the redundancy of the information source. The coding efficiency Q = H(x)/Llog(2)m = L min/L Where H(x) Entropy Lmin=H(x) q = H(x) / L log(2)m But m = 2 So H(x) = -(m,i=1) p(xi) log(2) p(xi) (m,i=1) H(x) = -(m,i=1) p(xi) log(2) 1/p(xi) (m,i=1) L length of the code Therefore L per source symbol (average code code-word length) L = (m,i=1) p(xi)ni P(xi) probability of event. Ni corresponding code length
so q = H(x)/L

We are having no. of classification in codes as fixed length code, variable length code, distinct codes, prefix free codes, uniquely decodable codes, instantaneous codes, optimal codes. Entropy coding Shannon-fano coding: Algorithm: 1. List the source symbols in order of decreasing probability. e 2. partition the set in to two sets that are close to equiprobables as possible and assign O to the upper set, 1 to the loser set. 3. continue this process, each time partitioning the sets with as nearly equal probabilities as possible until further portioning is not possible. Example problems: 1. A DMS x has four symbols x1,x2,x3 cu x4 with P(x1) = 1/2 , P(x2) = and P(x3) = P(x4) = 1/8 construct the Shannon-fano code for x./ show that his code has the optimum property fano that ni=I(xi) cu the code efficiency is 100 percent. Xi X1 X2 X3 X4 p(xi) 0.5 0.25 0.125 0.125 0 1 1 1 1 1 0 0 1 3 3 code 1 2
H(x) = (4,i=1) p(xi) log(2)1/p(xi) (4,i=1) = p(x1) log(2) p(x1) + p(x2) log(2) p(x2) + p(x3) log(2) p(x3) + p(x4) log(2) p(x4)

= -[0.5log(2)0.5 + 0.25log(2)0.25 + 0.125 log(2)0.125 + 0.125 log(2)0.125] = -[-0.5000 0.5000 0.3750 0.3750] = - [ - 1.75] = 1.75 bits/symbol. L = (4,i=1) p(xi) ni = 0.5(1) + 0.25(2) + 0.125(3) + 0.125(3) L = 1.75 Q = H(x)/Llog(2)m = 1.75/1.75 log(2)2 = 1 Therefore q = 100% 2. A DMS x has five equally likely symbols i. construct a Shannon fano code for x, and calculate the efficiency of the code. P(x1) = p(x2) = p(x3) = p(x4) = p(x5) = 0
Xi
p(xi)
code
X1 X2 X3 X4 X5
0.2 0.2 0.2 0.2 0.2
0 0 0 1 1 0 1 1 0 1 1 1
2 2 2 3 3
H (x) = 5 P (xi) log2 P(xi) = 5( -0.2 log2 0.2) I=1 H(x) = 2.32 b/symbol L = 5p(xi) ni = 0.2 (2+2+2+3+3) p(xi)

I=1 L = 2.4 .percentage = H (x) =2.32 . ------- ------- 2 Llog m2 2.4log 2
=96.67./.
3. a DMSX has five symbols x1,x2,x3,x4,x5 with P(x1) =0,4, P (x2) =0.19, p(x3) =0.16, P(x4) =0.15 and P(x5) =0.1. construct a Shannon fanocoding are calculate the efficiency of the code.
xi x1 x2
P(xi) 0.4 0.19 0 0
code 2 2 2 3 1 1 1 3
x3
0.16
0 1 -------1 0 1 1 0
x4
0.15
x5 0.1 H (x) =5 (xi) log2 P(xi) i=1 = - [0.4 log2 0.4 + 0.19 log2 0.19+0.16 log20.16+ 0.15log20.15 +0.1 log20.1] = - [ - 0.5288-0.45520.4230 0.4230-0.4105 -0.3322 ]
H(x) = 2.15 b/ symbol L= 5 P(xi) ni= 0.4 (2) =0.19(2)+ 0.16(2)+ 0.15(3) +0.1(3) i=1

L=2.25 Therefore percentage =H(x) = 2.15 ----- -------2 Llogm 2.25 log2 2
4. Apply shannons fano coding Procedure for the following message ensemble. message Take m=2.
X= [ x1,x2,x3,x4,x5,x6,x7,x8] P= [ , 1/8 , 1/16, 1/16, 1/16, , 1/16, 1/8 ]
Message X1 X6 X2 X8 X3 X4 X5 X7
P(xi) 0.25 0.25 0.125 0.125 0.0625 0..0625 0.0625 0.0625 0 0 ---0 1 -------1 0 0 1 1 1 1 0 1 1 1 1 0 1 1 0 1 1
code oo 0 1 100 101 1100 1101 1110 1111
Length 2 2 3 3 4 4 4 4
H(x) = - P (xi) log 2 P (xi) i=1
Page 143

= -[ 0.25 log2 0.25+ 0.125 log2 0.125 + 0.0625 log2 0.0625+ 0.0625 log2 0.0625+0.0625 [ log2 0.0625 + 0.25log2 0.25 +0.000625 log2 0.0625 + 0.125 log2 o.125] .0625
= - [ - 0.5000- 0.3750-0.2500- 0.2500 -0.2500-0.5000-0.2500 0.2500-0.3750]
H(x) = 2.75 b/symbol = P(xi) ni=0.25 (2) +0.125 (3) +0.125 (3)+ 0.0625(4) =0.0625(4) +0.0625(4)
= 2.75 Letter/ message.
Percentage = H(x) 2.75 =100 percentage ----------------2 Llogm2 2.75 log2
5. Apply Shannon fano coding for the following message enables.
X= [ x1,x2,x3,x4,x5,x6,x7] P= [ 0.4 0.2 0.12 0.08 0.08 0.08 0.04 ] Take m=3 [ - 1 0 1]
Xi X1 X2 X3
P (xi) 0.4 0.2 0.12 -1 ------0 -1 code 1

Page 144

2 0 X4 X5 0.08 1 0.08 1 0 2 X6 X7 0.08 0.04 1 1 1 1 -1 0 3 3 -1 2 0 2
H(x) = - P(xi) log2 P(xi) I=1 = 2.42 = H (x) = 2.42 -------- -------. 3 = 88.7 percentage Llog2m 1.72log2
7 L= P(xi) ni i=1 L = 1.72 LETTER/ MESSAGE.
Huffman coding ----------------------Huffman coding results an optimum code. Thus it is the code that has the highest efficiency. The Procedure is as follows, re 1.List the source symbols in order of decreasing Probability. 2. Combine the Probabilities of the two symbols having the lowest Probabilities, and reorder the resulted Probabilities. This step is called reduction. The same Procedure is repeated until there are two ordered Probabilities remaining. 3. Start encoding with the last reduction which consist of exactly two ordered Probabilities. Assign as the first digit in the code words for all the source symbols associated with the first Probability and assign to the second PROBABILITY. 4. Now go back and assign 0 and 1 to the second digit for the two Probabilities that were combined in the Previous reduction step, retaining all assignments made in step 5. Keep regressing this way until the first column is reached.

Problem:A DMS x has five symbols x1,x2,x3,x4,x5 with P(x1) =0.4,P(x2) =0.19, P(X3) = 0.16, p(x4) = 0.15 cu P(x5) = 0.1 Calculate the efficiency by using Huffman coding Xi M1 M2 P(xi) x1 x2 0.4 m1 0.19 m2 0.4m1 0.25 (m4+M5) 0.19 m2 M3 x3 0.16 m3 0.16m3 M4 M5 ] x4 0.15 ] m4 x5 0.1
0.4 0.6 0.35] ] 0.25] 0.4 0 1
code H(x) = 5 P(xi) log2 P(xi) x1= 1 =1 = - [ 0.4 log2 0.4 + 0.9 log2 0.19 + 0.16 log2 0.16+ 0 X2= 0 0 0 = 3 0.15 log2 0.15+0.1 log20.1] = - 0.5288- 0.4552 -0.4230 0.4105 -0.3322] =-[ 0.3322] X3 = 0 0 1 =3 X4=0 1 0 = 3 H(x) = 2.15

X5= 0 1 1 = 3 = P(xi) ni i=1 = (0.4) (1)+ (0.19) (3) + 0.16 (3) + 0.15(3) + 0,1(3)
L = 2.2 = h(X) = 2.15 ------- ------2 log2m 2.2 log2 = 97.7 percentage.
= 97.7 percentage.
2. A DMS x has five equally likely symbols P ( x1) = p(x2) = P(X3) =p(x4)= P(x5) =0.2 construct the Huffman code cu calculate the efficiency.
Xi
p(xi)
0.4(m4+m5)
0.4(m2+m3)
X1
0.2m1
0.2m1
0.4(m4+m5)
X2 X3
0.2m2 0.2m3
0.2m2 0.2m3
0.4 m1
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0 X4 X5 0.2m4 1 0.2m5
0.6 0(m1+m4+m5) 0.4 1(m2+m3)
code x1 x2 x3 x4 x5 04 10 11 000 001
Length 2 2 2 3 3
H(x) = - (5,i=1) p(xi)log(2)p(xi) (5,i=1) = - [(0.2 log2(0.2)X5] H(x) = 2.32 L = (5,i=1) p(xi) ni =(0.2) (2+2+2+3+3) L = 2.4 Therefore %q = H(x)/Llog2(M) = 2.32/2.4log2(2) = 96.7% 3. A discrete memory less source has an alphabet of seven symbols whose probabilities of occurrence

Symbol Prob/-
S0
S1
S2
S3
S4
S5
S6
0.25 0.25 0.125 0.125 0.125 0.0625 0.0625
Compute Huffman code cu compute the efficiency Source prob/S0 0.25 S0 0.25 0.25(S3+S4)
S1 S2
0.25 0.125
S1 0.25 S5+S6 0.125
0.25 S0 0.125 S1
S3 S4 S5 S6
0.125 0.125 0 0.625 1 0.625
S2 S3 S4
0.125 0 0.125 1 0.125
0.125 (S5+S6) 0.125 S2 1 0
0.25(S5+S6+S2) 0.375(S0+S1) 0.25(S3+S4) 0.25 0.25 S0 0.125 S1 0 0.25 1 0 0.5 1 0.037
S0 = 1 0 (2)
S5 = 0 0 0 0 (4)
Page 149

S1 = 1 1 (2) S2 = 0 0 1 (3) S3 = 0 10 (3) S4 = 0 1 1 (3) H(x) = -(6,i=1) P(xi) log(2)p(xi) (6,i=1)
S6 = 0 0 0 1(4)
= - [0.25 log2(0.25)+0.25 log2(0.25) + 0.125 log(2) 0.125 + 0.125 log(2) 0.125 + 0.0625 log2(0.0625) + 0.0625log2(0.625)] =-[-0.5000 0.5000 0.3750 0.3750 0.3750 0.2500 0.2500] H(x) = 2.625 b/symbol L = (6,i=1) (6,i=1) p(xi) 0.25(2)+0.25(2)+0.125(3)+0.125(3)+0.125(3)+0.0625(4)+0.0625(4) L = 2.625 %q = H(x)/L log2(m) = 2.625/2.625 log2(2) = 100% 4. A source transmitting 6 messages with prob. 0.30, 0.25, 0.15, 0.10 cu 0.08 resp. calculate q by Huffman coding Source prob. ni =
M1 M2
0.30 0.25
m1 0.30 m2 0.25
0.30(m1) 0.27 m3+m4
M3
0.15 0 0.125
m5+m6 0.13 0 0.15(m3)
0.25 m2 0.18
Page 150
M4 0.12 (m5+m6)

1 m5 0 m6 0.10 1 0.08
0.43(m5+m6+m2) 0 0.30(m1) 0.27 (m3+m4) 1
0.57 (m1+m3+m4) 0.43 (m5+m6+m2)
m1 = 0 0 (2)
m5 = 1 1 0 (4) m6 = 1 1 1(4)
m2 = 1 0 (3) m3 = 0 10 (3) m4 = 0 1 1 (3)
H(x) = -(6,i=1) P(xi) log(2)p(xi) (6,i=1) = - [0.3 log2(0.3)+0.25 log2(0.25) + 0.15 log(2) 0.15 + 0.12 log(2) 0.12 + 0.10 log2(0.10) + 0.08log2(0.08)] =-[-0.5211 0.5000 0.4105 0.3671 0.3322 0.2915] H(x) = 2.46 b/symbol L = (6,i=1) p(xi) ni = 0.3(2)+0.25(2)+0.125(3)+0.15(3)+0.12(3)+0.10(3)+0.08(3) (6,i=1)

L = 2.45 %q = H(x)/L log2(m) = 2.4/2.45 log2(2) = 97.9% cu redundancy R = i q = 0.021. i-q
5. A DMS source having five symbols cu probl/. Of 0.4,0.2,0.1,0.1 resp. find %q by Huffman coding.
M1 x1
0.4
m1 0.4
0.4 (m2+m3)
M2 x2 M3 x3 M4 x4 (m5+m6) m5 x5
0.2 0.2 0.1 0
m4+m5 0.2 0 m2 0.2 m3 0.2 1
0.4 m1 0.2 m4+m5 0.18
0.1 1
0.6(m1+m4+m5) 0.4 (m2+m3)
x1 = 0 0 (2) x2 = 1 0 (2)
x5 = 0 1 1 (3)
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x3 = 1 1 (2) x4 = 0 1 0 (3)
H(x) = -(5,i=1) P(xi) log(2)p(xi) (5,i=1) = - [0.4 log2(0.4)+0.2 log2(0.2) + 0.2 log(2) 0.2 + 0.1 log(2) 0.1 + 0.1 log2(0.1) =-[-0.5288 0.4644 0.4644 0.3322 0.3322] H(x) = 2.122 b/symbol L = (5,i=1) p(xi) ni = 0.4(2)+0.2(2)+0.2(2)+0.1(3)+0.1( (5,i=1) 0.4(2)+0.2(2)+0.2(2)+0.1(3)+0.1(3) L = 2.2 %q = H(x)/L log2(m) = 2.122/2.2 log2(2) = 96.45% 6. solve by Huffman coding [x] = [x1,x2,x3,x4,x5,x6,x7] [p] = [0.5,0.2,0.12,0.08,.0.08,0.08,0.04] x1 0.4 0.4
x2 0.4
0.2
0.2
x3
0.12 x4 0.08 0 1
0.12 0.12
0.2
0.16 x5 0.08
1 0.08 0.12
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x6
0.08
0.08
0.12
x7 0.04 0.4 0.36 0.60
0.24
0.41
x1 = 1 (1) x2 = 0 0 0(3) x3 = 0 1 1 (3) x4 = 0 0 1 0 (4)
x5 =0 0 1 1 (4) x6 = 0 1 0 0 (4) x7 = 0 1 0 1 (4)
H(x) = -(7,i=1) P(xi) log(2)p(xi) ,i=1) = - [0.4 log2(0.4)+0.2 log2(0.2) + 0.12 log(2) 0.12 + 0.08 log(2) 0.08 + 0.08 log2(0.08)+0.08log(2)0.08+0.04log(2)0.04] =-[-0.5288 0.4644 0.3671 0.2915 0.2915-0.2915-0.1857] H(x) = 2.42 bits/symbol L = (7,i=1) p(xi) ni = 0.4(1)+0.2(3)+0.08(4)+0.08(4)+0.08(4)+0.04(4) L = 2.48 %q = H(x)/L log2(m) = 2.42/2.48 log2(2) = 97.58% 7. solve by Huffman coding if m = 3 ( (-1,0,1)

P = (0.5,0.2,0.12,0.08,0.08,0.08,0.04) x1 0.4 0.4
x2 1
0.2
0.2
0.4
x3 x4 x5
0.12 0.08 0.08 -1 x6 0.08
0.2 -1 0.12 0.08 1 0 0
0.2 0 0.16 1
x7 0.04 x1 = -1 (1)
1 x5 =1 -1 (2) x6 = 1 0 (2) x7 = 1 1 (2)
x2 = -1 -1 (2) x3 = -1 0 (2) x4 = -1 1 (2) H(x) = -(7,i=1) P(xi) log(2)p(xi) (7,i=1)
= - [0.4 log2(0.4)+0.2 log2(0.2) + 0.12 log(2) 0.12 + 0.08 log(2) 0.08 + 0.08 log2(0.08)+0.08log(2)0.08+0.04log(2)0.04] =-[-0.5288 0.4644 0.3671 0.2915 0.2915-0.2915-0.1857]
H(x) = 2.42 bits/symbol


L = (7,i=1) p(xi) ni = 0.4(1)+0.2(2)+0.12(2)+0.08(2)+0.08(2)+0.08(2)+0.04(2) (7,i=1) L = 1.6 %q = H(x)/L log2(m) = 2.42/1.6 log2(3) = 2.42/2.536 = 95.42%
H(x) = 2.42 bits/symbol
L = (7,i=1) p(xi) ni = 0.4(1)+0.2(2)+0.12(2)+0.08(2)+0.08(2)+0.08(2)+0.04(2)
L = 1.6
%q = H(x)/L log2(m) = 2.42/1.6 log2(3) = 2.42/2.536 = 95.42%
Page 156

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CHENNAI INSTITUTE OF TECHNOLOGY

Sarathy Nagar, Kundrathur, Pudupedu, Chennai 600 069. Chennai

Department of Electronics and Communication Engineering

Subject Name: Communication Theory UNIT I

Subject Code: EC 2252

AMPLITUDE MODULATION SYSTEMS

CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY

Phasor representation of AM with carrier:

CHENNAI INSTITUTE OF TECHNOLOGY

These modulators are used in modulation.

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CHENNAI INSTITUTE OF TECHNOLOGY

Narrow bandwidth Frequency assignment Reduce the equipments limitations

28. Compare AM with DSB-SC and SSB-SC.

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CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

[1+ ma 2 / 2] PC = PC [1+ ma 2 / 2] PC % Power Saving = 2 x 100 2+ ma 2 (2/3) x 100 = 66.7 %

If ma = 1, then power saving

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CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY

Considering the SSB-SC signal I/p SC

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CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY

24. Write short notes on Frequency translation.

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CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY

CHENNAI INSTITUTE OF TECHNOLOGY HENNAI

CHENNAI INSTITUTE OF TECHNOLOGY