DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

ANALOG COMMUNICATION SYSTEMS

Prepared By:- Amandeep Singh Sandhu (Lab Supdt.)

Syllabus
1. To obtain Amplitude modulated Envelop and determine depth of modulation 2. To study envelop detector for demodulation of AM signal and observe diagonal peak clipping effect. 3. Frequency modulation using voltage controlled oscillator. 4. Generation of DSB-SC signal using balanced modulator. 5. Generation of single side band signal 6. To generate a FM Signal and measure Depth of modulation. 7. Detection of FM Signal using PLL. 8. To Study Super heterodyne AM receiver and measurement of receiver parameters viz. sensitivity, selectivity & fidelity. 9. Familiarization of PLL, measurement of lock and capture range, frequency demodulation, frequency multiplier using PLL. 10. Sampling Theorem & Reconstruction of Signal from its samples using Natural Sampling, Flat Top Sampling & Sample & Hold Circuits. 11. To study the circuit of PAM modulator & Demodulator 12. To study the circuit of PWM modulator & Demodulator 13. To study the circuit of PPM modulator & Demodulator

EXPERIMENT NO- 1
AIM: - To obtain Amplitude modulated Envelop and determine depth of modulation To study envelop detector for demodulation of AM signal and observe diagonal peak clipping effect. APPARATUS: - Cathode Ray Oscilloscope (CRO), Trainer Kit, connecting wires etc

Modulating wave signal

Input of AM Modulator (fig 1)

EXPERIMENT NO- 1
AIM: - To obtain Amplitude modulated Envelop and determine depth of modulation To study envelop detector for demodulation of AM signal and observe diagonal peak clipping effect. APPARATUS: - Cathode Ray Oscilloscope (CRO), Trainer Kit, connecting wires etc

THEORY: In Amplitude modulation, the amplitude of a carrier signal is varied by the modulating voltage whose frequency is invariably lower than (hat of the carrier frequency In practice, the carrier frequency may be highfrequency (HF). while the modulating frequency is audio frequency Formally AM is defined as a system of modulation in which the amplitude of the carrier signal is made proportional to the instantaneous amplitude of the modulating voltage Let the carrier voltage and the modulating voltage, Vc and Vm respectively be represented by Vc (t) = Vc Sin ct Vm (t) = Vm Sin mt Note that phase angle has been ignored in both expressions since it is unchanged by the amplitude modulation process. Its inclusion here would merely complicate the proceedings without affecting the result However, it will certainly not be possible to ignore phase angle when we deal with frequency and phase modulation From the definition of AM it follows that the maximum amplitude Vc of the un modulated carrier w have to be made proportional to the instantaneous modulating voltage Vm Sin mt when the carrier is amplitude-modulated.

Modulation index It can be defined as the measure of extent of amplitude variation about an unmodulated maximum carrier. As with other modulation indices, in AM, this quantity, also called modulation depth, indicates by how much the modulated variable varies around its 'original' level. For AM, it relates to the variations in the carrier amplitude and is defined in next paragraph. FREQUENCY SPECTRUM OF THE AM WAVES We shall show mathematically that the frequencies present in the AM wave are the carrier frequency and the first pair of sideband frequencies, where a sideband frequency is defined as

fSB = fc nfm and in first pair n = 1

When a carrier signal is amplitude-modulated, the proportionality constant is made equal to unity and the instantaneous modulating voltage variations are superimposed onto the carrier amplitude Thus, when there is temporarily no modulation, the amplitude of the carrier is equal to its un modulated value When modulation is present the amplitude of the carrier is varied by instantaneous value of the modulating signal This situation is illustrated in Fig-1. which shows how the maximum amplitude of the amplitude-modulated voltage is made to vary in accordance with modulating voltage changes Flg-1 also shows that something unusual (distortion, as it happens) will occur if Vm is greater than Vc The fact that the ratio Vm / Vc often occurs leads to the following definition of the modulation index

m = Vm / Vc
and called the percentage modulation.

-------- (1)

The modulation index is a number lying between 0 and 1, and it is very often expressed as percentage

From fig 1 and equation 1 it is possible to write an equation for the amplitude of AM voltage, thus we have

AVm Vc

V c V m Sin m t

V c m V c Sin m t
4

V c ( 1 m Sin m t )

---------- (1)

Output At Modulator Fig 2

MODULATED ENVELOP

AM DEMODULATION
5

It will be recalled that Vm = m Vc and it now possible to use this relation to calculate the index (or depth) of modulation from waveform shown in fig 2 as follow

Vm

Vmax Vmin 2

and

Vc Vmax Vm
m

Vmax

Vmax Vmin 2

Vmax Vmin 2

Vm Vc

(Vmax Vmin ) / 2 (Vmax Vmin ) / 2

(Vmax Vmin ) (Vmax Vmin )

REPRESENTATION OF AM Amplitude modulation may be represented in any of three ways, depending on the point of view. Fig -2 shows the frequency spectrum and so illustrates equation- 3. AM is shown simply as consisting of three discrete frequencies, of these the central frequency, i.e. the carrier has the highest amplitude, and other two are disposed symmetrically about it, having amplitude which are equal to each other, but which can never exceed half the carrier amplitude. The appearance of the amplitude - modulated wave is of great interest, and it is shown in Fig-2 for one cycle of the modulating sine wave. It is derived from Fig-1, which showed the amplitude, or what may now be called the top envelope of the AM wave, given by the relation. A = Vc + Vm Sinmt). Similarly, the maximum negative amplitude, or bottom envelope, is given by -A = - Vc + Vm Sinmt). The modulated wave extends between these two limiting envelopes and has a repretion rate equal to the un modulated carrier frequency The four quadrant multipliers allow operation with AC signals and provide greater speed and Bandwidth The complete circuit diagram is shown in Fig-3 II is based upon an offset linearised two quadrant

multiplier cell. The operational amplifiers convert the output current l0 to output voltage e0.

l0

IR IX

RX . RY VX . VY

Vs R

e0 I0 R5
Therefore

(R1 R5 ) . (VX VY ) R X R Y - Vs

e0

K . VX VY

During the demodulation process, AM output is given to the diode detector circuit. This diode detector is provided with a diode, after that the combination of resistor and capacitor, to form a low-pass circuit This arrangement is useful to reject high frequency carrier signal to enter into the output 6

Diagonal peak clipping effect.

AM Demodulation:Demodulation is the process of decoding an analog signal into digital data. When data is transferred over phone lines, a modem modulates the data into audible tones "carried" on frequencies between 0 Hz and 4 KHz. Demodulation is the act of removing the modulation from an analog signal to get the original baseband signal back. Demodulating is necessary because the receiver system receives a modulated signal with specific characteristics and it needs to turn it to base-band. Once the data reaches its intended destination, another modem demodulates the signal back into digital data. Cable TV networks also use modulation techniques to transfer data. But instead of audible tones, cable has sophisticated digital modulation schemes to greatly increase the amount of data that can be sent. There are several ways of demodulation depending on what parameters of the base-band signal are transmitted in the carrier signal, such as amplitude, frequency or phase.An example of a demodulation system is a modem, which receives a telephone signal (electrical signal) and turns this signal from the wire net into a binary signal for the computer. Envelope Detector An envelope detector can be used to demodulate a previously modulated signal by removing all high frequency components of the signal. The capacitor and resistor form a low-pass filter to filter out the carrier frequency. Such a device is often used to demodulate AM radio signals because the envelope of the modulated signal is equivalent to the base band signal. For demodulation, the modulated wave is fed to a circuit. This circuit is called detector. The output of the detector must have the same variations as the signal that modulated the carrier. If the variation in the output of the detector is not the same, then we say that the signal has been distorted. An envelope detector is an electronic circuit that takes a high-frequency signal as input and provides an output which is the "envelope" of the original signal. The capacitor in the circuit stores up charge on the rising edge, and releases it slowly through the resistor when the signal falls. The diode in series rectifies the incoming signal, allowing current flow only when the positive input terminal is at a higher potential than the negative input terminal. Most practical envelope detectors use either half-wave or full-wave rectification of the signal to convert the AC audio input into a pulsed DC signal. Filtering is then used to smooth the final result. This filtering is rarely perfect and some "ripple" is likely to remain on the envelope follower output, particularly for low frequency inputs such as notes from a bass guitar. More filtering gives a smoother result, but decreases the responsiveness; thus, real-world designs must be optimized for the application Diagonal Clipping in Diode Detector As the modulating frequency is increased, the diode ac load impedance, Zm does not remain purely resistive II does have reactive component also At high modulation depths, the current changes so fast that the time constant of the load does not follow the changes Hence the current decays slowly as shown in Fig. The output voltage follows the discharge law of RC circuit. This introduces distortion in the detected signal and it is called diagonal peak clipping effect. for this effect. Variations of modulated signal with percentage modulation are shown below. In each image, the maximum amplitude is higher than in the previous image. Note that the scale changes from one image to the next.

Block Diagram of Amplitude Modulation Process (FIG 1.2)

Block Diagram of Amplitude Demodulation Process (FIG 1.3)

EXPERIMENTAL PROCEDURE 1. Connect the AC Adaptor to the mains and the other side to the Experimental Trainer Switch ON the power 2. Observe the carrier and modulating waveforms and note their frequencies Carrier frequency is around 100 KHz and amplitude is variable from 0 -8Vp-p Modulating signal is 1KHz (Approx.) 3. Connect the earner and modulating signals to the modulator circuit. 4. Observe the amplitude modulated wave in synchronization with the modulating signal on a dual

trace CRO following fig shows the connections. 5. Connect Carrier l/P to ground .and apply a 2V peak to peak AF Signal input to {modulating l/P) and adjust P, for extreme anti clock wise position to get minimum AC output 6. Connect modulating l/P to ground and apply a 3 V peak to peak carrier signal to earner l/P and adjust P2 for extreme clock wise position to get minimum AC Output 7. Connect modulating inputs Carrier input to ground and adjust P3 for zero DC output 8. Make Modulating I/p 2 Vpp and Carrier l/p 3 Vpp peak to peak and adjust P4 for maximum output 9. Calculate maximum and minimum points on the modulated envelope on a CRO and calculate depth of modulation from the diagram 10. Observe that varying the modulating voltage, the depth of modulation varies 11. During demodulation connect this AM output to the input of the demodulator 12. By adjusting the RC time constant (i e . cut off frequency) of the filter circuit we get minimum distorted output 13. Observe that this demodulated output is amplified has some phase delay because of RC components. 14. Also observe the effects by changing the carrier amplitudes 15. In all cases, calculate the modulation index with the help of the following table

10

EXPERIMENT NO- 3
AIM: - Frequency modulation using voltage controlled oscillator.
To study Frequency Modulation Process and calculate modulation index (mf) Detection of FM Signal using PLL.

APPARATUS: -Cathode Ray Oscilloscope (CRO), Trainer Kit, connecting wires etc.

11

EXPERIMENT NO- 3
AIM: - Frequency modulation using voltage controlled oscillator.
To study Frequency Modulation Process and calculate modulation index (mf) Detection of FM Signal using PLL.

APPARATUS: -Cathode Ray Oscilloscope (CRO), Trainer Kit, connecting wires etc. THEORY: Frequency Modulation is a system in which the amplitude of the modulated carrier is keptConstant, while its frequency is varied by the modulating signal The first practical system wasPut

forward in 1936 as an alternative to AM in an effort to make radio transmissions moreResistant to noise Phase Modulation is a similar system in which the phase of the carrierSignal is varied instead of its frequency; as in FM. the amplitude of the carrier signal remainsConstant. The general equation of an un modulated wave or carrier wave may be written as

x = A Sin (t +]
where x = instantaneous value of voltage or current of carrier A= (maximum) amplitude - angular velocity, radians per second (rad/s) = Phase angle, rad Note that we represent an angle in radians

If any one of these three parameters is varied in accordance with another signal, normally of a lower frequency, then the second signal is called the modulation and the first is said to be modulated by the second In the frequency modulation, frequency of the carrier is made to vary F for simplicity, it is again assumed that the modulation signal is sinusoidal. This signal has two important parameters which must be represented by the modulation process without distortion: namely, its amplitude and frequency It is assumed that the phase relations of a complex modulation signal will be preserved By the definition of frequency modulation, the amount by which the carrier frequency is varied from its un modulated value called the deviation, is made proportional to the instantaneous value of the modulating voltage. The rate at which this frequency variations or oscillations takes place is naturally equal to the modulating frequency. The situation is illustrated in Fig Which shows the modulating voltage and the resulting frequency - modulated wave. Fig also shows the frequency variation with time which is seen to be identical to the variation with time of the modulating voltage As an example of FM. all signals having the same amplitude will deviate the carrier frequency by the same amount, say 45 KHz, no matter what their frequencies are. Similarly all signals of the same frequency say 2 KHz. will deviate the carrier at the same rate of 2000 times per second, no matter what their individual amplitudes are. The 12

amplitude of the frequency modulated wave remains constant at all times. This is infect, the greatest single advantage of FM.

Voltage controlled oscillator.

13

MATHEMATICAL REPRESENTATION OF FM The instantaneous frequency of the frequency modulated wave is given by

f= fc(1+K VmCos m t)
where fc = unmodulated (or average) carrier frequency, K = proportionally constant

Vm Cos mt = instantaneous modulating voltage (cosine being preferred for simplicity in calculations) The maximum deviation for this particular signal will occur when the cosine term has its maximum value, that is +/- 1 Under these conditions, the instantaneous frequency will be f= fc ( 1 + K Vm.) so that the maximum deviation 0 will be given by = K Vm fc The instantaneous amplitude of the FM signal will be given by a formula of the form

V = A Sin { F (c, m)} = A Sin

Where F (c, m) is some function, as yet undetermined, of the carrier and modulating frequencies. This function represents an angle and will be called for convenience. The problem now is to determine the instantaneous value (i.e. formula) for this angle

is the angle traced out by the vector A in time t. lf A were rotating with a constant angular velocity, say P this angle would be given by Pt in radians In this instance. However, the angular velocity is anything
but constant. It is in fact, governed by the formula for obtained from the equation

f= fc(1+K VmCos m t)
that is = c(1+KVm, Cosmt) In order to find . must be integrated with respect to time. Thus

= dt = c(1+K Vm, Cos m t)dt = c (1+K Vm, Cos m t)dt

c [t + K Vm, Sin m t] m ct

ct

K Vm fc Sin m t fm

[ Sin m t] fm ------------------------------------- (1)

The deviation utilized, in turn the fact that c is constant, the formula {Cos nx dx=(Sin nx] / n} which had shown that K voltage, thus

Vm fc =

Eqn 1 may now be subdivided to give the instantaneous value of the FM

V ASin(ct

[ Sin m t] ) fm
14

The modulation index for FM. mf is defined as

mf

max. frequency deviation modulating frequency

fm

V= A Sin ( c t + mf Sin m t)
it is important to note that as the modulating frequency decreases as the modulating voltage amplitude i.e. remains constant, the modulation index increases This will be the basis for distinguishing frequency modulation from phase modulation Note also that mf Which is the Ratio of two frequencies is measured in radians CIRCUIT DESCRIPTION In this experiment Frequency modulation is generated by using IC 8038 The frequency of the waveform generator is a direct function of the DC voltage given at pin 8 (measured from V+) E- By altering this voltage, frequency modulation is performed. For small deviations (e.g. +/-10%) the modulating signal can be applied directly to pin 8. merely providing DC decoupling with capacitor as in Fig. An external resistor between pins 7 & 8 is not necessary, but it is used to increase input impedance from about 8K (pin 7 & 8 connected together) to about (R + 8 8K) For larger FM deviations or for frequency sweeping the modulating signal is applied between the positive supply voltage and pin 8. During the demodulation. FM output is given to a phase lock loop ( 565 IC) We have seen that, during lock, the average dc level of the phase comparator output is directly proportional to the frequency of the input signal. As the frequency shifts, it is this output which causes the VCO to shift and keep tracking In other words, the phase comparator output is an exact replica of the original modulating audio signal Fig 3 shows connections of 565 as FM demodulator The component values shown are for a carrier frequency of 70 KHz approx The demodulated output is followed by a three stage filter to remove RF component. A small capacitor of 0 01 f is connected between pins 7 & 8 to eliminate possible oscillations in the current source Voltage controlled oscillator. In this experiment Frequency modulation is generated by using IC 8038 The frequency of the waveform generator is a direct function of the DC voltage given at pin 8 (measured from V+). By altering this voltage frequency modulation is performed For small deviations (e.g +/-10%) the modulating signal can be applied directly to pin 8 merely providing DC decoupling with a capacitor. An external resistor between pins 7 & 8 is not necessary, but i t can be used to increase input impedance from about 8KQ (pin 7 & 3 connected together] to about (R + 8 KQ). For larger FM deviations or for frequency sweeping the modulating signal is applied between the positive supply voltage and pin 8 During the demodulation, FM output is given to a phase lock loop ( 565 IC] We have seen that during lock the average dc level of the phase comparator output is directly proportional to the frequency of the

15

FM DEMODULATION USING PLL LM565

16

input signal As the frequency shifts, it is this output which causes the VCO to shift and keep tracking In other words the phase comparator output us an exact replica of the original modulating audio. The demodulated output is followed by a three stage filter to remove RF component A small capacitor is connected between pins 7 & 8 to eliminate possible oscillations in the current source. Frequency demodulation Frequency demodulation is the process that enables us to recover the original modulating signal from a frequency modulated signal. Here we describe a direct method of frequency demodulation involving the use of a popular device known as frequency discriminator, whose instantaneous frequency of the input of FM signal the circuit of frequency discriminator is shown in fig 1phase discriminator is shown in fig 2 both the primary and secondary tuned circuits are tuned to the centre frequency of the carrier the operation of this circuit very much depend on the 90 phase shift between the primary and secondary voltage at resonance This immediately rules out slope detection as a means of recovering the information signal. Instead, phase and frequency response of the tuned circuit are used to obtain the demodulated output EXPERIMENTAL PROCEDURE 1. Connect the AC Adaptor to the mains and the other side to the Experimental Trainer 2. Observe carrier signal and modulating signals on a dual trace CRO. 3. Carrier signal Modulator output without any modulating input 160 KHz and Amplitude is 4Vp-p 5V p-p (Variable)

A) Carrier frequency is

B) Modulating signal: Frequency is 12.5 KHz Amplitude is

Connect modulating signal to the modulator input and observe modulating signal and FM output on a dual trace CRO. The Band Width required for FM signal as per careen's rule is Band Width B

2(( fm)
f fm

Bandwidth is twice the sum of frequency deviation & modulating frequency. The modulation Index For Example t = 5Sec When f = Frequency deviation & fm = Modulating Frequency

1 t

1 5 Sec

0.2Mhz

fm 7.1KHz

f fm

100 KHz 7.1 KHz

14.08

17

Ampl. Of Mod Signal

Modulated Frequency

Mod Freq. = fm= = t / req.

BW Requried 2(f+fm)

0 0.5 1 Vpp 1.5 Vpp 2 Vpp 2.5 Vpp

f0 f1 f2 f3 f4 f5

--f1 - fc f2 - fc f3 - fc f4 - fc f5 - fc

4. Trigger CRO w.r.t CH1. Adjust amplitude of the modulating signal until we get undistorted FM output. It is difficult to trigger FM on analog CRO that is why you adjust modulating signal amplitude until small distortion notified in FM output Note: - In this position modulating signal amplitude is approximately 6Vp-p. Put CRO In CH1 V/div in 2 and CH2 v/div in 2 and time/div in 20s position. 5. Calculate max frequency and min frequency from the FM output and calculate modulating index 6. During demodulation connect circuit as shown in block diagram 2 7. Decrease the amplitude of modulating signal generator until we get undistorted demodulated output. Note: - In this position maximum modulating signal generator output is 1V p-p is due to capture range restrictions of PLL in demodulator. 8. Adjust the potentiometer in demodulator section until we get demodulated output. Precautions: 1. All the circuits should be according to circuit diagram. 2. All the connections should be tight. 3. The trainer kit should be switched off until the complete connections are made. 4. Switch off the trainer kit when not in use.

18

EXPERIMENT NO- 4
AIM: -Generation of DSB-SC signal using balanced modulator. APPARATUS: -Balance Modulator kit, CRO. Function generator, Connection Probes etc.

BLOCK DIAGRAM REPRESENTATION OF BALANCED MODULATOR

19

EXPERIMENT NO- 4
AIM: -Generation of DSB-SC signal using balanced modulator. APPARATUS: -Balance Modulator kit, AM Kit, CRO. Function generator, Connection Probes THEORY: Double-sideband suppressed-carrier transmission (DSB-SC): DSB-SC transmission in which (a) frequencies produced by amplitude modulation are symmetrically spaced above and below the carrier frequency and (b) the carrier level is reduced to the lowest practical level, ideally completely suppressed. In the double-sideband suppressed-carrier transmission (DSB-SC) modulation, unlike AM, the wave carrier is not transmitted; thus, a great percentage of power that is dedicated to it is distributed between the sidebands, which imply an increase of the cover in DSB-SC, compared to AM, for the same power used DSB-SC transmission is a special case of Double-sideband reduced carrier transmission. This is used for RDS (Radio Data System) because it is difficult to decouple. Definition of a DSBSC Consider two sinusoids, or co sinusoids, cos (t) and cos (t). suppressed carrier signal, or DSBSC, is defined as their product, namely: A double sideband

DSBSC= Ecost .cost

Generally, and in the context of this experiment, it is understood that:

>>
Equation (3) can be expanded to give:

cost .cost = (E/2) cos( - )t + (E/2) cos ( +)t

Equation 3 shows that the product is represented by two new signals, one with a frequency that is the sum frequency (+) and the other with a frequency that the difference (-) see Figure 1. These two components were derived from a carrier term with a frequency of rad/s, and a message with a frequency of rad/s. Due to the absence of the carrier component in the product signal, this product signal is described as a Double Sideband Suppressed Carrier (DSBSC) signal.The term carrier comes from the context of double sideband amplitude modulation (commonly abbreviated to just AM). AM is introduced in a later experiment (although, historically, AM preceded DSBSC). appearance of a DSBSC (equation. 1) is generally as shown in Figure Balanced Modulator. Balanced modulator is used for generating DSB-SC signal. A balanced modulator consists of two standard amplitude modulators arranged in a balanced configuration so as to suppress the carrier wave. The two modulators are identical except the reversal of sign of the modulating signal applied to them. The time domain

20

21

Modulator: The IC MC 1496 is used as Modulator in this experiment. MC 1496 is a monolithic integrated circuit Balanced modulator/Demodulator, is versatile and can be used up to 200 MHz. These modulators are used in this experiment to produce DSB-SC signals Controls is provided to balance the output Multiplier: A balanced modulator is essentially a multiplier. The output of the MC 1496 balanced modulator is proportional to the product of the two input signals. If you apply the same sinusoidal signal to both inputs of a ballooned modulator. This circuit uses a tuned push pull amplifier in which the modulating voltage is applied in the push pull at the two transistors & the carrier voltage is applied in parallel to the two transistors. The carrier signal contains no information the information is contained in each of the two side bands. Accordingly the carrier may be cropped or eliminated without losing any information. If we suppress carried, the system becomes suppressed carrier double side band system. It can be used in the following types one is Using the transistor or FET's or by Using diodes. The circuit diagram shows that balanced modulator. In this ic is used.t balance modulator have ic no 1496. The output voltage is given by

V0=2K1 Vmcoswmt +2Ka2VmVc {cos(wc+wm)t + cos(wc-wm)t}

In the ckt, the RF signal is applied at the pin no 8 &10 through resistance the modulating signal is applied through variable resistance. If you use two sinusoidal signals with different frequencies at the two inputs of a balanced modulator (multiplier) you can produce AM-DSB/SC modulation This is generally accomplished using a highfrequency carrier" sinusoidal and a lower frequency "modulation" wave form (such as an audio signal from microphone) The figure 1.1 is a plot of a DSB - SC wave form, this figure is the graph of a 100KHz and a 5KHz sinusoidal multiplied together. Figure 1.2 shows the circuit that you will use for this experiment using MC 1496 balanced modulator/ demodulator.

EXPERIMENTAL PROCEDURE FREQUENCY MULTIPLIER 1. As the circuitry is already wired you Just have to trace the circuit according to the circuit diagram given above 2. Connect the AC Adaptor to the mains and the other side 'ON' the power. to the Experimental Trainer. Switch

3. Measure the output voltages of regulated power supply circuit. 4. Connect same 5KHz sinusoidal signal (note that inputs of 0.1 to 0.4 appear to be acceptable but if there is distortion reduce the amplitude) to both the carrier and modulation inputs. 5. Observe the output on an oscilloscope and adjust the carrier null potentiometer (the one provided in modulator block) until the output is a 10KHz sinusoidal. Note that this is a very sensitive adjustment because you are making the biasing at both inputs exactly the same to get 22

the multiplying effect of the device, check the input and output frequencies using a frequency counter and verify that you have exact multiplication by 2 in frequency. 6. Change the input frequency and verify that you have multiplication at 10OKHz and 500KHz. Note that there is a decrease in amplitude at the higher frequencies but multiplying action continues. AM-DSB/SC 1. Apply a 100KHZ, 0.1 peak sinusoidal to the carrier input and a 5KHz, 0.1 peak sinusoidal to the modulation input. 2. Adjust the carrier null potentiometer to obtain a waveform like the one in figure 1.1. If spectrum analyzer is available, observe and sketch the output in the frequency domain.

23

EXPERIMENT NO- 5
AIM: -Generation of single side band signal APPARATUS: -Trainer kit, CRO. Function generator, Connection Probes etc.

BLOCK DIAGRAM REPRESENTATION OF SSB MODULATION SYSTEM

WAVEFORMS:

24

EXPERIMENT NO- 5
AIM: -Generation of single side band signal APPARATUS: -Trainer kit, CRO. Function generator, Connection Probes etc. THEORY: common method of generating a single-sideband (SSB) signal is to split an incoming audio intelligence signal into two signals that are identical, except for a 90-degree phase difference, and to split a constant amplitude and frequency carrier signal into two carrier signals that also have a 90degree phase difference. The resulting four signals are applied to two balanced-modulators such that each balanced-modulator receives one audio-phase and one carrier-phase signal. The output of each balanced-modulator is a double-sideband signal. However, there is no phase difference between one of the sidebands in the two modulator outputs, but there is a 180-degree phase difference between the other sideband in the two outputs. The two balanced modulator outputs are summed, which cancels one sideband and doubles the strength of the other, resulting in a single-sideband signal. Single-sideband modulation (SSB) is a refinement of amplitude modulation that more efficiently uses electrical power and bandwidth. It is closely related to vestigial sideband modulation (VSB) mplitude modulation produces a modulated output signal that has twice the bandwidth of the original baseband signal. Single-sideband modulation avoids this bandwidth doubling, and the power wasted on a carrier, at the cost of somewhat increased device complexity. SSB was also used over long distance telephone lines, as part of a technique known as frequency-division multiplexing (FDM). FDM was pioneered by telephone companies in the 1930s. This enabled many voice channels to be sent down a single physical circuit, for example in L-carrier. SSB allowed channels to be spaced (usually) just 4,000 Hz apart, while offering a speech bandwidth of nominally 300-3,400 Hz. SSB and VSB can also be regarded mathematically as special cases of analog quadrature amplitude modulation. An SSB signal is produced by passing the DSB signal through a highly selective band pass filter. This filter selects either the upper or the lower sideband. Hence transmission bandwidth can be cut by half if one sideband is entirely suppressed. This leads to single side band modulation (SSB). In SSB modulation bandwidth saving is accompanied by a considerable increase in equipment complexity. Single Sideband Suppressed Carrier (SSB-SC) modulation was the basis for all long distance telephone communications up until the last decade. It was called "L carrier." It consisted of groups of telephone conversations modulated on upper and/or lower sidebands of contiguous suppressed carriers. The groupings and sideband orientations (USB, LSB) supported hundreds and thousands of individual telephone conversations. Due to the nature of-SSB, in order to properly recover the fidelity of the original audio, a pilot carrier was distributed to all locations (from a single very stable frequency source), such that,

25

26

the phase relationship of the demodulated (product detection) audio to the original modulated audio was maintained. Also, SSB was used by the U.S. Air force's Strategic Air Command (SAC) to insure reliable communications between their nuclear bombers and NORAD. In fact, before satellite communications SSB-was the only reliable form of communications with the bombers. The main reason-SSB-is superior to-AM,-and most other forms of modulation are: (1) since the carrier is not transmitted in SSB, there is a reduction by 50% of the transmitted power. In AM out of 100% modulation: 67% of the power is comprised of the carrier; with the remaining 33% power in both sidebands. (2) Because in SSB, only one sideband is transmitted, there is a further reduction by 50% in transmitted power. (3) Finally, because only one sideband is received, the receiver's needed bandwidth is reduced by one half--thus effectively reducing the required power by the transmitter another 50%. PROCEDURE: 1. Refer to the block diagram and Carry out the following connections. 2. Keep all the switch faults in OFF position. 3. Connect o/p of FUNCTION GENERATOR section OUT post to I/p of Balance Modulator1 SIGNAL IN post. 4. Connect o/p of VCO OUT post to the input of Balance modulator1 CARRIER IN post . 5. Connect power supply with proper polarity to the kit ACL-01 & ACL-02, while connecting this, and ensure that the power supply is OFF. 6. Switch on the power supply. 7. Refer to the block diagram and Carry out the following connections. 8. Keep all the switch faults in OFF position. 9. Connect o/p of FUNCTION GENERATOR section OUT post to I/p of Balance Modulator1 SIGNAL IN post. 10. Connect o/p of VCO OUT post to the input of Balance modulator1 CARRIER IN post 11. Connect power supply with proper polarity to the kit ACL-01 & ACL-02, while connecting this, and ensure that the power supply is OFF. 12. Switch on the power supply & Keep switch SW1 towards 1-10 KHz position. 13. Keep sine level about 1 Vpp, Freq. about 3 KHz & Keep switch SW2 towards 500 KHz position. 14. Keep LEVEL about 2Vpp; FREQ. about 452 KHz. 27

15. Keep Balanced Modulator 1, Carrier Null in central position, so that the modulator is balanced and obtain an AM signal across the output with suppressed carrier, OUT LEVEL in clockwise position 16. Connect OUT post of balanced modulator 1 to IN post of ceramic filter. 17. Observe the SSB signal at the OUT post of ceramic filter. You can observe that the filter extracts only one of the two components (sidebands) generated by balance modulator. 18. Measure the frequency fc of the carrier (post CAR.), fm of the modulating signal (post SIG.) and fssb of the SSB signal across the output of the filter (post OUT). 19. Repeat Check that: fssb = fc + fm, this means that the band extracted by the filter corresponds to the Upper Side Band. 20. the last measurements setting the frequency of the carrier to 458 KHz. You obtain: fssb = fc fm, this means that the band extracted by the filter corresponds to the Lower Side Band. 21. Increase the frequency of the modulating signal (SINEWAVE) and check that the SSB signal attenuates and tends to a null.

28

EXPERIMENT NO- 7 (& 9)

Aim: - Familiarization of PLL, measurement of lock and capture range, frequency demodulation,

frequency multiplier using PLL

Apparatus :- PLL kit Function generator CRO connecting probes etc

Fig1

29

EXPERIMENT NO- 9)
Aim: - Familiarization of PLL, measurement of lock and capture range, frequency demodulation,

frequency multiplier using PLL

Apparatus :- PLL kit Function generator CRO connecting probes etc Theory :The basic block diagram of PLL is shown in figure 1 & circuit diagram is shown in figure 2 It consist of 1. Phase detector 2. A low pass filter 3. An error amplifier 4. A voltage controlled oscillator (VCO) The VCO is the free running multi-vibrator The output of the phase detector contains low frequency component which reduces to dc when both signal are of the same frequency The low pass filter selects this component which is then amplified S fed back to the oscillator This bias alter the frequency of the oscillations in such a way as to reduce the output of the phase detector since the oscillator frequency is controlled by bias voltage it is termed as voltage controlled oscillator In practice the oscillator is often a multi-vibrator type circuit producing a square wave output for a sinusoidal input signal the circuit when properly adjusted will look the oscillation fundamental frequency to the input frequency & the dc output of the phase detector is then proportional to V cosG Where vi is the amplitude of the input and 8 is the phase angle between input & oscillator fundamental This dc bias automatically alters the frequency of the oscillator such that the bias itself reduce to zero a condition which is met when the phase angle 9 is 90 When applied to detection of am signal the VCO is locked exactly in frequency and phase to the earner is externally shifted by a further 90 it does not matter whether the total phase difference is 0 or 180 the fundamental component of the VCO output will be proportional to sin wt where wc is the angular frequency of the carrier The low pass filter controls the capture range If VCO frequency is far away the beat frequency will be too high to pass through the filter and PLL will not respond we say that signal is out of capture band Once locked filter VCO free running frequency is adjusted with RT& CT to be in center of input frequency range PLL is internally broken between VCO output and phase comparator input A short circuit between pins 4 and 5 connects VCO output to phase comparator so as to compare f0 with input signal fs The Phase lock loop circuit has many applications in communications. Some examples of its use include Local Oscillators, Clock recovery, FM generation, FM/PM demodulation, and QAM demodulation.

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FM DEMODULATION USING PLL LM565

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FM detection FM demodulation is the process that enables us to recover the original modulating signal from a frequency modulated signal. Here we describe a direct method of frequency demodulation involving the use of a popular device known as frequency discriminator, whose instantaneous frequency of the input of FM signal the circuit of frequency discriminator is shown in fig 1phase discriminator is shown in fig 2 both the primary and secondary tuned circuits are tuned to the centre frequency of the carrier the operation of this circuit very much depend on the 90 phase shift between the primary and secondary voltage at resonance This immediately rules out slope detection as a means of recovering the information signal. Instead, phase and frequency response of the tuned circuit are used to obtain the demodulated output Hold range The "Hold range" is defined as the range of frequencies that the loop will remain in lock after initially being locked. Attach scope probe A to the input signal and probe B to the VCO output (Pin 4-5, which are to be tied together). Trigger on channel A. Apply a sine wave input, vary the input frequency and observe the signals as they "lock" and lose "lock". Lock range The "Lock range" is the difference between the maximum and minimum frequencies at which the loop will acquire lock when initially unlocked.If the PLL is initially not locked-in to the input signal and the incoming frequency is gradually moved toward the VCO free running frequency, a point is reached where the Lock range VCO frequency is "captured" by the incoming frequency and the PLL becomes locked Frequency multiplier. The 565 can be used to multiply a given frequency by locking it to a harmonic of the signal frequency This system has a limitation that the lock range decreases as higher harmonics become weaker A better techniques is to insert a frequency divider between VCO and the phase comparator as shown in fig-1. The output of VCO is divided digitally by a number of times the multiplication is desired. The sub divided frequency is then given to the phase comparator. The PLL no keeps the VCO tracking at a frequency which is N times the input signal frequency. A Schematic diagram showing the use of 565 and a digital divider for frequency multiplication is shown in fog. Transistor BC 107 acts as a logic interface to drive the logic circuit. IC 74161 (or 74163) is shown wired for divide by 2, 4, 8 & 16.

PROCEDURE:I. Switch 'ON' the power supply by connecting the Power card to the A C mains II. Then check the VCO output at PIN 4 of IC565

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III. This is a square wave form the frequency of the wave form depends on Ct (0 01 f) and R5 (variable 10K pot). IV. Observe the divided by 2, 4. 8 and 16 outputs of 74161 IC V. Connect some variable frequency input from the Function Generator to the input terminals o 565Then show the O/P at pin 4 x pin 11 12. 13 and 14 of IC 74161 by shorting pin 4& of IC 565 VI. Now connect the 2 output of IC 74161 to pin 5 of IC 565 and observe the VCO output (pin ' of IC 565] on one input of the Dual Trace Oscilloscope VII. Observe input signal given to the input of the PLL ( from Signal Generator) on the second channel of the Oscilloscope VIII. Vary the input signal Frequency until we get at pin 4 of IC 565 is twice the input signal Frequency That means input Frequency is multiplied by 2 times IX. Now disconnect the -2 output and connect -4 output of IC 74161 to pin 5 of 565 X. Adjust the 10KQ potentiometer provided at pin 8 of IC 565 until we get 4 times the input signal Frequency at pin 4 of IC 565 Otherwise adjust the input signal Frequency until we get the same result

33

EXPERIMENT NO- 8
Aim: - To Study Super heterodyne AM receiver and measurement of receiver parameters viz.

sensitivity, selectivity & fidelity

Apparatus: - heterodyne AM receiver kit Function generator CRO connecting probes etc.

34

EXPERIMENT NO- 8
Aim: - To Study Super heterodyne AM receiver and measurement of receiver parameters viz.

sensitivity, selectivity & fidelity

Apparatus: - heterodyne AM receiver kit Function generator CRO connecting probes etc.

Thoery:The super heterodyne receiver was developed to overcome the disadvantages of earlier receivers. A block diagram of a representative super heterodyne receiver is shown in figure 1. Super heterodyne receivers may have more than one frequency-converting stage and as many

amplifiers as needed to attain the de- sired power output The super heterodyne receiver is a commonly implemented method of demodulating Frequency Division Multiplexed signals, by using the heterodyning or frequency translation principle. The super heterodyne receiver solves many of the problems associated with the original Tuned

Radiofrequency receivers, by optimizing the characteristics and circuitry to operate at a single intermediate frequency, thereby increasing stability, sensitivity and selectivity. The selectivity of a radio receiver is its ability to discriminate between the wanted information signal, and all other signals received by the aerial. Good selectivity characteristics are essential in any radio receiver, as they determine the ability to reject all other signals other than the tuned signal, thereby ensuring information clarity with minimum distortion. Operation of Superheterodyne Receiver:- The typical arrangement for a super heterodyne receiver is as shown in Figure 1. The received RF input signal is amplified and then passed on to a mixer. In the mixer the modulated RF carrier is mixed (multiplied) with a sinusoidal waveform generated by a local oscillator (LO). The heterodyning local- oscillator frequency fLO is selected to be above the RF; thus the system is referred to as a Super heterodyne System. The process of mixing generates sum and difference frequencies, and so the mixer output consists of a carrier of frequency fLO+fRF and a carrier fLO-fRF Each carrier is modulated by the baseband signal to the same extent as was the input RF carrier, and therefore, even although the signal has been shifted in frequency, it will still contain the same information in its sidebands. A filter rejects the sum frequency, leaving the remaining Intermediate Frequency (IF) fIF = fLO-fRF.. The modulated IF carrier is applied to a demodulator, and the information is passed on to an audio speaker.

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measuring the overall strength of the signal and automatically adjusting the gain of the receiver to maintain a constant level of output. When the signal is strong, the gain is reduced. Circuit for Super heterodyne Receiver Although super heterodyne radio receivers looks not very complicated but for practicable purposes there must be additional circuitry involved in the design. One of them is Automatic Gain Control (AGC).The AGC circuit keeps the receiver in its linear operating range by weak, the gain is increased, or allowed to reach its normal maximum. For simplicity of circuit, I will present a circuit without AGC. The complete circuit at next page appears complicated,. Local Oscillator Stage:- In most of AM receivers, local oscillator (LO) is designed with the help

of a special component, known as oscillator coil. They are not more than ordinary Transformer but with an additional capability, that their core is movable between the coils. The main purpose of having a moveable core is to tune the oscillator at desire band. The top side of Local Oscillator (LO) is colored white in order to distinguish it from intermediate frequency transformers. They come in metal housing and there are five pins plus two pins of metal housing Mixer Stage:- Multiplying the RF signal from the antenna with the frequency of LO is an

essential part of demodulation. Different methods are employed for this purpose, transistors, diodes, transformers or other electronic components may be used. But I prefer IC NE612 for this purpose in my circuit for many reasons. The main reason is that using IC instead of other component is that the need of RF stage amplifier is reduced very much, because NE612 takes very little power from input signal. Moreover, other important reason is that the quality of mixing is very good and output signal is very much close to the intermediate frequency (IF). Another good reason is that as we all know that for mixer circuit the supply voltage should be very constant, and NE612 has its own voltage regulator, that mean that we dont have to implement one by our self. And the biggest advantage is that its use is very simple, attach antenna to pin 1 or 2, ground pin no. 3 and 6 volt to pin no. 8. Then connect LO between pin 6 and 7, and get IF frequency out from pin 4 and 5. Coupling Capacitor:- As we know that in super heterodyne design our RF stage and LO should

oscillate in such a way that their difference is always 455 kHz (IF frequency). In order to get simultaneously tuning of both circuits, we use coupling capacitor. They are just pair of two capacitors connected parallel to each other. One is for main tuning and other is for fine-tuning. In the case of FM, there are four capacitors. Intermediate Frequency Transformer/Filter (IFT):- Intermediate frequency filter is made with

the help of transformer similar to the LO stage, so it is called IFT. They too came in metal housing as LO. The only difference is that they also have a capacitor built in them., the IFT is, in fact, a parallel oscillatory circuit with a leg on its coil. The coil body has a ferrite core (symbolically shown with single upward straight dashed line) that can be moved (with screwdriver), which allows for the setting of the resonance frequency of the circuit, in our case 455 kHz. The same body contains another coil, with fewer quirks in it. Together with the bigger one it comprises the HF transformer that takes the signal 37

from the oscillatory circuit into the next stage of the receiver. Both the coil and the capacitor C are placed in the square-shaped metal housing that measure 10x10x11 mm. From the bottom side of the housing you can see 5 pins emerging from the plastic stopper, that link the IFT to the PCB, being connected inside the IFT. Besides them, there are also two noses located on the bottom side, which are to be soldered and connected with the device ground. Japanese IFT's have the capacitor C placed in the cavity of the plastic stopper, as shown in figure. The part of the core that can be moved with the screwdriver can be seen through the eye on the top side of the housing, figure 10-d. This part is colored in order to distinguish the IFT's between themselves, since there are usually at least 3 of them in an AM receiver. The colours are white, yellow and black (the coil of the local oscillator is also being placed in such housing, but is being painted in red, to distinguish it from the IFT) Detector Stage:- The detector stage is implemented with the easiest method that is with envelop

detection. No description is necessary, only the circuit is given below. Please not that this method is known asynchronous detection. Audio Amplifier Stage:- In order to get good and loud voice from the speaker it is essential to

have an audio frequency (AF) amplifier or simply audio amplifier. For this purpose well-known audio amplifier IC LM386 is used. It is low priced and good quality IC. We can get 20 to 200 times amplification from it. Pin 5 gives the output, which in turn is connected with the loudspeaker. The speaker should be round about 10 rated to 1W. If speaker is not available just omit the LM386 and place a headphone just after the detector. RECEIVER CHARACTERISTICS:Sensitivity, noise, selectivity, and fidelity are important receiver characteristics. These characteristics will be useful to you when performing receiver tests. They can help you to determine whether a receiver is working or not or in comparing one receiver to another.

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Sensitivity :-The ability of a receiver to reproduce weak signals is a function of the sensitivity of

a receiver. The weaker a signal that can be applied to a receiver and still produce a certain value of signal output, the better the sensitivity rating. Sensitivity of a receiver is measured under standardized conditions. It is expressed in terms of the signal voltage, usually in the microvolts that must be applied to the antenna input terminals to give an established level of the output. The output may be an ac or dc voltage measured at the detector output or a power measurement (measured in decibels or watts) at the loudspeaker or headphone terminals. Noise All receivers generate a certain amount of noise, which you must take into account when

measuring sensitivity. Receiver noise may originate from the atmosphere (lightning) or from internal components (transistors, tubes). Noise is the limiting factor of sensitivity. You will find sensitivity is the value of input carrier voltage (in microvolts) that must be applied from the signal generator to the receiver input to develop a specified output power. Selectivity is the degree of distinction made by the receiver between the desired signal and

unwanted signals. You will find the better the ability of the receiver to reject unwanted signals, the better its selectivity. The degree of selection is determined by the sharpness of resonance to which the frequency-determining circuits have been engineered and tuned. You usually measure selectivity by taking a series of sensitivity readings. As you take the readings, you step the input signal along a band of frequencies above and below the circuit resonance of the receiver; for example, 100 kilohertz below to 100 kilohertz above the tuned frequency. As you approach the tuned frequency, the input level required to maintain a given output level will fall. As you pass the tuned frequency, the required input level will rise. Input voltage levels are then compared with frequency. They can be plotted on paper or you might view them on an oscilloscope. They would appear in the form of a response curve. The steepness of the response curve at the tuned frequency indicates the selectivity of the receiver. Fidelity The fidelity of a receiver is its ability to accurately reproduce, in its output, the signal that

appears at its input. You will usually find the broader the band passed by frequency selection circuits, the greater your fidelity. You may measure fidelity by modulating an input frequency with a series of audio frequencies; you then plot the output measurements at each step against the audio input frequencies. The resulting curve will show the limits of reproduction.

Procedure:Experiment Setup for Measurement of Selectivity:- For AM broadcast receiver, several characteristics tests have been standardized. Figure. G1 shows the standard setup for measurement of selectivity. The AM signal from the AM signal generator is applied to the receiver through a standard coupling network known as dummy antenna. The output power is measured using power meter. Here, the loudspeaker is replaced by a load resistance of equal value. Selectivity curve of a tuned circuit: - The output of AM signal generator is adjusted to frequency same as receiver tuned frequency and to get standard receiver power output (50 Mw). Now the frequency of the AM generator is varied to either side of the receiver tuned frequency. The output of the receiver naturally falls, since the input frequency is now incorrect. Thus, every time the AM generator 40

output voltage is adjusted and noted down to get a standard 50 mW receiver output power. The attenuation is calculated and plotted to get typical selectivity curve shown in Fig. G.2. Experimental Setup for Measurement of Sensitivity:- In practice sensitivity test is carried out before selectivity test. It uses same setup as for selectivity test, as shown in Fig. G.3. While measuring sensitivity practically, the radio source and load is standardized so that the variation in measurement conditions do not affect the results. Here, a standard AM signal with 30% modulation and 400 Hz modulating frequency from AM signal generator is applied to the receiver through a standard coupling network (dummy antenna). The input voltage required to obtain a standard output power of 50 mW is measured. During measurement the input is increased at full volume till the standard output power is dissipated the load resistance. The input in microvolts of carrier voltage is then a measure of sensitivity. Fig. G3 shows the typical sensitivity curve. As shown in the Fig. G.2 the sensitivity varies over the tuning band Experimental Setup for Measurement of Fidelity:- The experimental setup for measurement of fidelity is same as setup required for measurement of selectivity and sensitivity (Refer. Fig. G.2) Here, the output of standard AM generator is fed to the receiver via dummy antenna. Instead of loudspeaker, a equivalent resistance of value 850 O is connected across the output of receiver. The output power is measured with the help of power meter connected across the equivalent output resistor. Keeping the output of AM signal generator constant the output power is measured for all modulating frequencies. The attenuation in the output power is indicated by negative sign.

41

EXPERIMENT NO 9
Aim: - Sampling Theorem & Reconstruction of Signal from its samples is using Natural

Sampling, Flat Top Sampling & Sample & Hold Circuits.

Apparatus: - Sampling & Reconstruction Trainer, CRO frequency meter function generator connecting probes etc.

42

EXPERIMENT NO 9
Aim: - Sampling Theorem & Reconstruction of Signal from its samples uses Natural Sampling,

Flat Top Sampling & Sample & Hold Circuits.

Apparatus: - Sampling & Reconstruction Trainer, CRO frequency meter function generator Connecting probes etc. Theory: Sampling is the process of converting an analog signal into a sequence of discrete values. If done correctly, sampling does not introduce distortions into the system. The sampling theorem defines the conditions for such successful sampling. The minimum rate at which samples must be taken (the sampling frequency) is of particular interest. The main objective of this lab is to experimentally verify the sampling theorem. You will first sample an analog signal, and then reconstruct it using a lowpass filter. You will investigate two types of sampling. Sampling Principles: When an analog message is conveyed over an Analog Communication System, the full message is typically used at all times. To send the same analog signal over a Digital Communication System, only its samples are required to be transmitted at periodic intervals. The

receiver will receive only samples of the message. It must attempt to re-construct the original message at all times from its samples only. Anyone associated with Digital Communication System has to know the Principles of Sampling and Re-Construction. It may seem astonishing that samples of a message and not the

entirewaveform, which can adequately describe all the information in a signal. However, we shall find that under some reasonable conditions, a message can be recovered exactly from its samples, even at times in between samples. SAMPLING THEOREM FOR LOW PASS RANDOM SIGNALS: Let m (t) be a signal, which is bandlimited such that its highest frequency spectral component is fm. Let the values of m(t) be determined at regular intervals separated by times Ts < 1/2fm, i.e., the signal is periodically sampled every Ts second. Then these samples m (n Ts) where n is an integer, uniquely determine the signal and the signal may be re-constructed from these samples with no distortion. Ts are the sampling time. SAMPLING CIRCUITRY: There are thee commonly used sampling circuits; 1) Natural Sampling circuit 2) Sample & Hold circuit 3) Flat Top Sampling circuit. 1. The Natural Sampling section takes sine wave as analog input and samples the analog input at the rate equal to the sampling signal. 2. For Sample and Hold circuit, the output is taken across a capacitor, which holds the level of the samples until the next sample arrives. 3. For Flat top Sampling, an inverted sampling clock is used. Output of flat top sampling circuit is pulses with flat top and top corresponds to the level of analog signal at the instant of rising edge of the clock signal.

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A. NATURAL SAMPLING AND ITS RECONSTRUCTION. PROCEDURE: 1 2 Refer to the Block Diagram (Fig. 1.1) & carry out the following connections and switch settings. Connect power supply in proper polarity to the kit DCL-01 & switch it on.

3. Connect the 1KHz, 5Vpp Sine wave signal, generated onboard, to the BUF IN post of the BUFFER and BUF OUT post of the BUFFER to the IN post of the Natural Sampling block by means of the connecting chords provided, 4. Connect the sampling frequency clock in the internal mode INT CLK using switch (SW4). 5. Using clock selector switch (Sl) select 8 KHz sampling frequency. 6. Using switch SW2 select 50% duty cycle (see Fig. 1.1). 7. Connect the OUT post of the Natural sampling block to the input IN1 post of the 2nd Order Low Pass Butterworth Filter. 8. Chang the sampling frequency to 4 KHz, 16 KHz, 32 KHz, and 64 KHz using push button switch S1. 9. Set the sampling frequency to 16 K Hz using switch S1. Chang the duty cycle of the sampling clock from 50% to 10%, and 90% using dipswitch SW2 (Fig. 3.2). B. SAMPLE AND HOLD AND ITS RECONSTRUCTION. PROCEDURE: 1 Refer to the Block Diagram (Fig. 1.2) & carry out the following connections and switch settings. 2 Connect power supply in proper polarity to the kit DCL-01 & switch it on.

3. Connect the 1KHz, 5Vpp Sine wave signal, generated onboard, to the BUF IN post of the BUFFER and BUF OUT post of the BUFFER to the IN post of the Sample & Hold block by means of the connecting chords provided, 4. Connect the sampling frequency clock in the internal mode INT CLK using switch (SW4). 5. Using clock selector switch (Sl) select 8 KHz sampling frequency. 6. Using switch SW2 select 50% duty cycle (see Fig. 1.1). 7. Connect the OUT post of the Natural sampling block to the input IN1 post of the 2 nd Order Low Pass Butterworth Filter. 8. Chang the sampling frequency to 4 KHz, 16 KHz, 32 KHz, and 64 KHz using push button switch S1. 9. Set the sampling frequency to 16 K Hz using switch S1. Chang the duty cycle of the sampling clock from 50% to 10%, and 90% using dipswitch SW2 (Fig. 3.2).

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C. FLAT TOP SAMPLING AND ITS RECONSTRUCTION: PROCEDURE: 1 Refer to the Block Diagram (Fig. 1.3) & carry out the following connections and switch settings. 2 Connect power supply in proper polarity to the kit DCL-01 & switch it on.

3. Connect the 1KHz, 5Vpp Sine wave signal, generated onboard, to the BUF IN post of the BUFFER and BUF OUT post of the BUFFER to the IN post of the Flat Top Sampling block by means of the connecting chords provided, 4. Connect the sampling frequency clock in the internal mode INT CLK using switch (SW4). 5. Using clock selector switch (Sl) select 8 KHz sampling frequency. 6. Using switch SW2 select 50% duty cycle (see Fig. 1.1). 7. Connect the OUT post of the Natural sampling block to the input IN1 post of the 2 nd OrderLow Pass Butterworth Filter. 8. Chang the sampling frequency to 4 KHz, 16 KHz, 32 KHz, and 64 KHz using push button switch S1. 9. Set the sampling frequency to 16 K Hz using switch S1. Change the duty cycle of the sampling clock from 50% to 10%, and 90% using dipswitch SW2 (Fig. 3.2). CONCLUSION: Comparing the reconstructed output of 2 nd order Low Pass Butterworth Filter for all the three types of sampling, it is observed that the output of the sample and hold is the best as compared to the output of natural sampling and the output of the flat top sampling. From the above observations we conclude that as the sampling frequency is increased, the reconstructed output is less distorted and almost original signal is reconstructed. For a sampling frequency of 4KHz, only 4 samples of the 1KHz signal are taken, whereas that for a sampling frequency of 8KHz, 8 samples of 1 KHz signal is taken. Hence, as the number of samples taken of the signal increases, the distortion of the reconstructed signal decreases. As per the Nyquist Criterion at least two samples are required for the reconstruction of the signal. If the Nyquist Criterion is not satisfied, or if the signal is not band limited, then spectral overlap, called "aliasing" occurs, causing higher frequencies to show up at lower frequencies in the recovered signal, and specially in voice transmission intelligibility is seriously degraded Thus, universally for the voice band (300Hz to 3000Hz), the sampling frequency used is 8KHz, which satisfies the Nyquist Criterion. Moreover, we can conclude that as the duty cycle increases, the sampling time, i.e., the time period over which the signal information is obtained, is more. approaches that of the original signal. Hence the reconstructed Signal Amplitude

46

EXPERIMENT NO 10
Aim: - To study the circuit of PAM modulator & Demodulator Apparatus: - PAM Trainer, CRO frequency meter function generator Connecting probes etc.

47

EXPERIMENT NO 10
Aim: - To study the circuit of PAM modulator & Demodulator Apparatus: - PAM Trainer, CRO frequency meter function generator Connecting probes etc. Theory:- Time division multiplexing is technique of transmitting more than one information on the one channel. This no activity time interrupt can be used to include the sample from other channels as well. This means that several information signals can be transmitted over single channel by sending samples from different information sources at different movements at a particular time. This technique is known as Time division multiplexing (TDM). TDM is widely used in digital communication system to increase the efficiency of transmission medium. TDM can be achieved electronically switching the samples with such a way that they interleave sequentially at correct instance in a time without mutual interface. The concept of TDM is illustrated in diagram. The multiplexer here is a single pole rotating switch or commutator. It may be electronic switch. The commutator has two functions: to take narrow sample of each input message at a rate of fs. Which is higher than 2w and to sequentially interleave the sample inside the interval Ts=1/fs. The multiplexed signal at the output of commutator is applied to amplitude modulator, which converts the PAM pulses into form suitable for transmission over the communication channel. The input message signals are passed through low pass filter before applying them to commutator. These filters are actually the anti aliasing filters. The multiplexed PAM signals can be received properly only if transmitter & receiver commutators are synchronized to each other in terms of the speed & position. Here synchronization is most essential and critical part of TDM. At the receiver end the received signal is applied to a pulse amplitude demodulator which performs the reverse operation of pulse amplitude modulator. Here is one more commutator, which is used for demultiplexing. The low pass filters at the receiving end are used to reconstruction of original signals. Procedure:1. 2. 3. 4. 5. Connect signals from function generator to section CH-I of transmitter trainer kit. Connect CRO to CH-I at input side & CH-II output side of kit. Switch on the power supply. Observe the input and output wave shapes at CRO screen. For demodulation connect the output of modulator to demodulator and check output of demodulation of CRO screen.

Precautions: 1. All the circuits should be according to circuit diagram. 2. All the connections should be tight. 3. The trainer kit should be switched off until the complete connections are made. 4. Switch off the trainer kit when not in use.

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EXPERIMENT NO 11
Aim: - To study the circuit of PWM modulator & Demodulator Apparatus: - PWM Trainer, CRO frequency meter function generator Connecting probes etc

49

EXPERIMENT NO 11
Aim: - To study the circuit of PWM modulator & Demodulator Apparatus: - PWM Trainer, CRO frequency meter function generator Connecting probes etc Theory: Pulse Modulation is used to transmit analog information, such as continuous speech signal It is a system, in which the continuous wave forms are sampled at regular intervals Information regarding the signal is transmitted only at the sampling times, together with any synchronizing pulse that may be required. At the receiving end, the original waveform is reconstructed from the information regarding the samples if these are taken frequently enough. Despite the fact that information about the signal is not supplied continuously, as in Amplitude Modulation or Frequency Modulation, the resulting receiver output can have negligible distortion Pulse Modulation may be subdivided broadly into two categories Analog and Digital In the former, the sample Amplitude is varied continuously, while in the later case the sample Amplitude is varied to the nearest predetermined level Pulse width Modulation is an analog type of modulation. PULSE WIDTH MODULATION In Pulse Width Modulation, we have a fixed Amplitude and starting time of each pulse, but the width of each pulse is made proportional to the Amplitude of the Modulating Signal at that instant. ; In the experiment, Pulse Width Modulation is generated by a Monostable Multivibrator, using 555-IC connected in the Monostable mode as shown in the diagram . Initially, the Synchronous clock from the synchronous clock generator is given to the pin-2 of the 555-IC and the AF signal is given to the pin-5 of the same If we observe the output at pin-3, we see the Pulse Width Modulated signal. The width of each pulse is varied in accordance with instantaneous Amplitude of the AF signal. PULSE WIDTH DEMODULATION The Demodulation of the Pulse Width modulation is a simple process Pulse Width Modulated signal is fed to an integrating (RC) circuit (Low pass Filter) from which the modulating signal is recovered, whose Amplitude at any time is proportional to the Width of the Pulse An Amplifier circuit is used to amplify the recovered AF signal EXPERIMENTAL PROCEDURE Switch ON the demonstration board by connecting the power card to the AC mains Observe the Synchronous clock generator output and AF signal outputs Connect Synchronous clock generator output to the Synchronous clock input point of PWM modulator and observe the same clock on one channel of a dual trace CRO Trigger the CRO with respect to CH 1

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Apply a variable DC voltage of 8 to 12 volts from any external regulated power supply Observe the PWM output on CH 2 If we observe the PWM output, its width varies according to the modulating voltage A variable amplitude of AF signal is given to observe how the PWM signals are varying for AC modulating voltages

In this case we have to trigger the CRO w.r.t modulating voltage

52

EXPERIMENT NO 12
Aim: - To study the circuit of PPM modulator & Demodulator Apparatus: - PPM Trainer, CRO frequency meter function generator Connecting probes etc

53

EXPERIMENT NO 12
Aim: - To study the circuit of PPM modulator & Demodulator Apparatus: - PPM Trainer, CRO frequency meter function generator Connecting probes etc Theory: - Pulse position modulation (PPM) is a pulse modulation technique that uses pulses that are of uniform height and width but displaced in time from some base position according to the amplitude of the signal at the instant of sampling. Pulse position modulation is also sometimes known as pulse-phase modulation. Pulse position modulation has the advantage over pulse amplitude modulation (PAM) and pulse duration modulation (PDM) in that it has a higher noise immunity since all the receiver needs to do is detect the presence of a pulse at the correct time; the duration and amplitude of the pulse are not important. In Pulse Position Modulation, we have a fixed Amplitude and pulse width of each pulse, but the position of each pulse is made proportional to the Amplitude of the Modulating signal at that instant. In the PPM Trainer board, Pulse Position Modulation is generated ay two Monostable Multivibrators as shown in Panel diagram ( PPM Modulator ) In this the first Multivibrator generates the pulse width Modulation output and the second Multivibrator generates the pulse position modulation. Initially, the Synchronous clock from the trainer is given to the pin2 of the first 555(IC) ( which is connected in Monostable mode) and the AF signal is given to the pin-5 of the same 555 (IC). Now if we observe the output at pin-3 of the same first 555(IC) . we get Pulse Width Modulation signal. The width of each pulse is varied if we change the Amplitude of the AF signal which is applied at pin - 5 of 555(IC). The output of the 1st 555(IC) is connected to Pin2 of the 2nd 555 (IC) through a capacitor So the generated Pulse Width Modulation pulses are used to Trigger the second Monostable Multivibrator. The position of each pulse is varied in accordance with the already generated pulse width modulation. But the PWM depends on the input AF sigmil Therefore the generated pulse position modulation signal depends on the Amplitude of me AF signal If we change the Amplitude of the AF signal, the position of the each pulse is varied But the width of the each pulse is remaines constant, because the time constant of the second Monostable Multivibrator is constant (fixed).

PULSE POSITION DEMODULATION The demodulation of the pulse position modulation is quite a simple process Pulse Position Modulation is fed to an Integrating (RC) circuit (Low Pass Filter) from which a modulating signal emerges whose Amplitude at any time is proportional to the pulse position modulation at that time. Amplifier circuit is also provided to get the AF signal generator back with amplified signal.

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EXPERIMENTAL PROCEDURE Switch ON the experimental kit Observe the Synchronous clock generator output and AF signal Generator outputs. Connect the Synchronous clock generator output to the Synchronous clock input point of PPM modulator & observe the same clock on channel 1 of a dual trace CRO Trigger the CRO with respect to C H1 Apply a variable DC voltage of 8 to 12 volts from any external regulated power supply Observe the PPM output on CH2 By varying the AF signal voltage, PPM output clock position changed, but its width maintains constant. If we observe the PWM output, its width vanes according to the modulating voltage

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