C. K. Pithawalla College of Engg. & Tech.

, Surat INDEX
SR No. 1 2 3 4 5 6 7 8 9 10 11 Date Practical Name
Amplitude Modulation Frequency Modulation Sampling Theorem Pulse Width Modulation Pulse Amplitude Modulation Pulse Position Modulation Pulse Code Modulation Time Division Modulation Amplitude Shift keying Frequency Shift keying Phase Shift Keying

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PRACTICAL 1
AIM:- To study Amplitude Modulation & Amplitude Demodulation. APPARATUS:- Kit of Amplitude Modulator & Demodulator, C.R.O., wires. THEORY:AMPLITUDE MODULATION
If the amplitude of the carrier signal is changed in accordance with the change in amplitude of the modulating signal keeping frequency & phase of the carrier signal to be constant is called as Amplitude Modulation.(A.M.) In its basic form AM produces a signal with power concentrated at the carrier frequency & in two adjacent sidebands. Each sideband is equal in bandwidth to that of the modulating signal & is a mirror image of the other.Thus, most of the power output by an AM transmitter is effectively wasted: Half power is concentrated at the carrier frequency, which carries no useful information (beyond the fact that a signal is present); the remaining power is split between two identical sidebands. The spectrum of an AM consists of the carrier component & two sidebands. Thus, the portion of the AM signal spectrum that lies above the frequency W C(& below WC) is called as the upper sideband. While that lies below WC( & above WC)is called as the lower sideband. The frequency range of an AM is 20Hz to 20KHz AM is commonly used at radio frequencies & was the first method used to broadcast radio

AMPLITUDE DEMODULATION
The permissible depth of modulation decreases with increasing frequency of the modulating signal & also increasing the value of the constant. The amplitude modulation system consists of the following sections: - Input Audio Amplifier - Modulating Audio Signal Generator - R.F. Carrier Oscillator - Double Balanced Amplitude Modulation Section - Band Pass Filter Section - D.C. Voltage Generating Section 1. Input Audio Amplifier: It consists of pre-amplifier & output amplifier stage. This section amplifies low level audio signal coming from mike/loudspeaker & it is given to the AM modulator section for live AM modulation. 2. Modulating Audio Signal Generator: IC1 (ICL 8038) waveform generator I.C. is used to generate sine wave signal. Pot (22k) is used to vary its frequency.

3. R.F. Carrier Oscillator: Transistor T1 is used to generate R.F. sine wave signal. Pot(P1) is used to vary its frequency. The frequency range is 200kHz to 1MHz. The output amplitude varies from 0 to 10v (p-p). 4. Double Balanced Amplitude Modulation Section: I.C. 1496 is used as balanced modulator. This I.C. is has two inputs as it works as the balanced modulator. The amplitude demodulation system consists of following sections: - Diode Detector - Product Detector - Local Oscillator - Low Pass Filter Section - Output Audio Amplifier - Power Supply 1. Diode Detector: It consists of diode (0A79) & capacitor & resistor. It works as an envelope detector circuit. 2. Local Oscillator: The local oscillator is required at demodulator side for product detector circuit. 3. Low Pass Filter Section: This section passes only low frequency up to 3.4kHz & reduces other frequencies. This filter circuit is required to remove R.F. carrier signal components from demodulated signal. 4. Output Audio Amplifier: This section is same as input audio amplifier except pre-amplifier.

BLOCK DIAGRAM:-

CIRCUIT DIAGRAM:-

WAVEFORM:-

PROCEDURE:A) First connect R.F. terminal of balanced modulator section to sine output terminal of audio oscillator. Connect +2v connector link. Connect C.R.O. channel-1 at sine wave output signal of audio frequency generator. B) Adjust amplitude of sine wave 2v (p-p) & audio frequency to1kHz. Connect R.F. carrier terminal of balanced modulator section to output terminal of R.F. carrier oscillator. Set output of R.F. oscillator to high frequency & amplitude of 6v (p-p). C) Connect C.R.O. channel-2 at modulated output at balanced modulator section. Trigger C.R.O. by channel-1.The amplitude modulated wave can be observed. D) See the effect on the AM modulated signal by varying the audio oscillator frequency & the amplitude. E) Vary R.F. carrier oscillator frequency & amplitude & observe its effect on the AM modulated signal. F) Calculate modulation index by following formula: m = / v = (Vmax Vmin) / (Vmax + Vmin) G) Draw the waveforms of modulating, carrier, modulated & demodulated signals. H) Calculate modulation index for three different conditions: Under, critical & over. I) Observe the waveforms on C.R.O. in x-y mode. CONCLUSION:-

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PRACTICAL 2
AIM: To study Frequency Modulation and Demodulation APPARATUS: Frequency modulation and Demodulation Kit, C.R.O Connecting probs. THEORY:
A sinusoidal carrier signals said to be frequency modulated when its' instantaneous frequency varies in accordance with the instantaneous amplitude of the modulating signal. FM modulated signal is represented Mathematically as:Where Vm is modulating signal.

FM = Vc Cos c t + K F Vm dt

A ratio of the max frequency deviation i.e. f to the modulating signal freq. fm is known as the modulation index MF MF = f/fm The FM Modulation system consists of: 1) Input audio amplifier 2) Modulating audio signal generator 3) Frequency modulation section 4) Dc voltage generation section The frequency Demodulation System consists of: 1) Frequency demodulation by PLL or 2) FM demodulation by Foster-seeley discriminator.or 3) FM demodulation by ratio detector(any one of above three) 4) o/p Audio Amplifier 5) Power Supply Input Audio Amplifier:The i/p Audio amplifier consists of pre-amplifier stage.The Section is used to amplify low level audio signal and give it to FM modulator section for FM modulation Modulating Audio Signal Generator:IC 8038 waveform generator IC is used to generate sone wave signal,pot P2 22K is used to vary its frequency. Another pot is used to vary its amplitude. FM modulation Section: For generating a FM signal a voltage controlled oscillator (VCO) can be used, the instantaneous frequency of which is made proportional to the instantaneous modulation signal amplitude. Another technique of generating FM signal consists in varying the oscillating frequency of an oscillator Varying the effective capacitance or inductance varies the frequency of oscillation.The required variation is presented by an amplifier, which in turn is controlled by modulating signal. Demodulation of FM signal using PLL A phase locked loop (PLL) in a f/b system that attempts to track instantaneous phase and hence the frequency of an I/p signal 565 IC is used

BLOCK DIAGRAM:

Audio Signal Generator

I/p Audio Amplifi er

FM Modulator

PLL FM Demodulation

FM modulated output

R/F Carrier Generat or

O/p Audio Amplifie r

Audio Output

PROCEDURE:
1) Connect CRO at channel 1 to observe the sine wave o/p (modulating signal) from Audio Frequency Generator Section (IC 8038). Adjust its amplitude and frequency at Say 1 v p-p and 1Khz. 2) See the Carrier signal at CRO ch. 2 from Voltage Controlled Oscillator section, which comprises of IC 8038 and Amplifier IC 356. The frequency of this section can be varied between 10 kHz to 150 kHz. 3) Observe the Modulated o/p at Modulator section on CRO ch. 2. See the effect of change in frequency of modulated o/p with change in amplitude of modulating signal. 4) Finally observe the demodulated o/p at PLL section and see the wave form. Verify that with change in amplitude/frequency of modulating signal the amplitude/frequency of demodulated o/p is varied accordingly.

CONCLUSION:

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PRACTICAL 3
AIM: To Study the Different Sampling Processes to prove Sampling Theorem. APPARATUS: -Kit of Sampling, CRO, Connecting Probes and Wires. THEORY: The Sampling Theorem Kit consists of following stages/sections: 1) Input Audio Amplifier 2) Modulating Audio Signal Generator 3) Sampling Pulse generator 4) Sampling Section 5) Low pass filter Demodulator Section 6) O/p Audio amplifier 7) Power Supply 1) Input Audio Amplifier: The I/p Audio amplifier ckt consists of pre amplifier and o/p Amplifier stages. This section is used to amplify low-level audio signal coming from Mike/Loudspeaker, and given to PAM modulator Section. 2) Modulating Audio Signal Generator: It consist of IC1 (8038) Waveform generator IC is used to generate Sine wave Signal. Pot p2 is used to vary its frequency. Pot p2 is used to vary the amplitude of sine wave signal. The switch S1 is used for course variation of frequency. 3) Sampling Pulse Generator: This Section is used to provide Sampling signal to Sample and hold Ckt. This section is based on the VCO IC CD4046BThis control voltage is varied by frequency pot (10K pot) .The pulse width of this pulse signal is varied by using monostable IC CD 4528B. Thus Sampling signal of variable frequency and pulse width is generated by thus section. 4) Sample and hold Ckt: To generate PAM signal sample and hold circuit is used. Here multiplexer CD 4053 B is used as sample and hold ckt. To observe different types of sampling connect as shown in procedure. 5) Low pass filter Demodulation: It is based on low pass filter is passes only low frequency up to 3.4 kHz & reduces all other frequency Thus it removes high frequency quantization noise of PAM signals. 6) O/p Audio amplifier: It is same as I/p audio Amplifier except preamplifier stage. 7) Power Supply Section: It is used for Different Power Supply voltages. Power Supply requirements are 230 V AC 50 Hz.

BLOCK DIAGRAM: -

Modulating Signal Audio Signal Generator Sampling Section J1 J2 J3

Modulated O/p

Envelop Detector

Low pass filter LPF Demodulator

Sampling Pulse Generator

Demodulated O/p

PROCEDURE:
1) Connect CRO at channel 1 to observe the sine wave o/p from Audio Frequency Generator Section (IC 8038). Adjust its amplitude and frequency say 1 v p-p and 1Khz (Fm) respectively. 2) Keep Frequency port of Sampling Pulse Generator Section(IC 8038- w/f gen.) at its mid position. Part 1:3) Observe the o/p sampling pulses on CRO at channel 2 i.e. Fs. 4) The I/p to the Sample & Hold ckt are Sine wave (Fm) and Sampling Pulse (Fs).Now Keeping Fs < 2 Fm, Fs =2 Fm & Fs > 2 Fm for all these cases to prove the Sampling Theorem. 5) Observe the modulated o/p at CRO on channel 2 of Sampling Section. 6) Also see the Demodulated o/p sine wave from LPF Demodulator section. 7) Observe the recovered o/p (demodulation) for all the above cases thus it can be proved that The Band limited signal can faithfully be recovered from its samples if the sampling frequency Fs >= 2 Fm. Part 2: Jumper Setting for Different types of sampling processes type Natural Sampling Flat top Sampling Sample & Hold Jumper J1 close open open J2 open close close J3 Open Close open

Observations Table :Condition f < fC f = fC f > fC Modulating signal Freq. Modulating Amplitude Sampling Frequency Output Frequency Output Amplitude

CONCLUSION: -

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PRACTICAL 4
AIM: To study the Pulse Width Modulation /Demodulation scheme and to observe related
waveform

APPARATUS: PWM KIT No: COM107, POWER SUPPLY, C.R.O. etc THEORY:
1) AUDIO SIGNAL GENERATOR: IC 8038 it is wave generator IC is used to sine wave signal. The sine wave output signal is available at pin2 of IC 1 and it then amplified by audio amplifier ic. The amplied output is available at output terminal pot p2 is used to vary the amplitude of sine wave signal. The switch s1 is used for course variation of frequency. Output amplitude vary from 0-5 v p-p 2) SAMPLING PULSE GERNERATOR: To generate PWM signal comparator cut is used sampling saw tooth waveform is required to provide sampling signal to comparator cut This section is based on voltage control oscillator IC CD4046B pulse frequency is determined by vary control voltage of vco at pin 9 of this IC. This control voltage is vary red by pot of 10k 3) PWM MODULATION SECTION: to generate PWM signal comparator cut is used here IC lm311 omp is as comparator 10k pot is used to vary amplitude of it output. The modulating input signal given as one input of comparator. The sampling is given to second input of comparator the output of comparator produce PWM signal 4) PWM DEMODULATION SECTION: this section consists is based on comparator and low pass filter PWM signal one of input of comparator the 1 v dc reference signal is given to second input of comparator output of comparator is pulse width demodulated signal This pulse width demodulated signal is passing through low pass filters made of three 741 IC and low pass filters passes only low frequency upto 3.4 kHz and reduce all other frequency

BLOCK DIAGRAM:
Practical

Audio signal Generator

DC Generator

PWM MODULATOR

PWM
DEMODULATOR

Power supply SAMPLING PULSE GENERATOR

Theoritical

WAVEFORM:

PROCEDURE:
1) Connect CRO at channel 1 to observe the sine wave o/p from Audio Frequency Generator Section (IC 8038).Adjust its amplitude and frequency say 1 v p-p and 1 KHz Respectively. 2) Keep Frequency port of Sampling Pulse Generator Section(IC 4046- VCO) at its mid position. 3) Also keep pulse width port of the same section at its mid position to adjust its duty cycle. 4) Observe the o/p sampling pulse with its duty cycle at channel 2 of CRO. 5) The Sampling pulse is converted to equivalent saw tooth w/f using mono stable multi vibrator IC LF356. 6) Observe the Saw tooth o/p at CRO channel 2. 7) Observe the modulated o/p at Pulse Width Modulator Section which uses IC LM 311 as a comparator (Two i/ps of comparator are Modulating signal & saw tooth wave). Now vary the pulse frequency of sampling pulse and see the effect on the phase of PWM o/p. 8) Observe the change in the width of the modulated o/p with respect to the change in the amplitude of the modulating signal. 9) Observe the Demodulated o/p at channel 2 of CRO from Pulse Width Demodulator Section comprising of Comparator IC 741 and a Low Pass Filter. 10) Measure the amplitude and frequency of demodulated o/p with change in the amplitude and frequency of modulating signal respectively. 11) Finally it is seen that Error in the recovered o/p Increases with Increase in Signal Amplitude Error in the recovered o/p Increases with Increase in Signal Frequency Error in the recovered o/p Decreases with Increase in Sampling Pulse Frequency

CONCLUSION:

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PRACTICAL 5
AIM: -To study Pulse Amplitude Modulation and demodulation. APPRATUS: -Pulse modulator/demodulator kit, C.R.O. THEORY: - The pulse amplitude modulation and demodulation system consists of following
section. 1. 2. 3. 4. 5. 6. 7. 8. Input audio amplifier Sampling pulse generator section Modulating audio generator Pulse amplitude modulator DC voltage source Pulse amplitude demodulator O/p audio amplifier Power supply

Input audio amplifier section The input audio amplifier circuit consists of pre-amplifier and o/p amplifier stage. This section is used to amplify low-level audio signal and give it to A.M. section for live A.M. Modulating audio signal generator IC 8038 waveform generator IC is used to generate sine wave signal. The frequency range is 20Hz to 20KHz. O/P amplitude vary from 0 to 20V(p-p), Sampling pulse generator section To generate PAM signal and hold circuit is used. The sampling pulse generator is required to provide sampling signal. This section is based on voltage-controlled oscillator (Vco). The pulse frequency is determined by varying control voltage of Vco. Pulse amplitude modulator To generate PAM signal sample and hold circuit is used. Pulse amplitude demodulator section The section is based on low pass filter. It passes only low frequencies upto 3.4Khz and reduces all other frequencies. O/P audio amplifier section IC 810 performs the function of the audio amplifier, device and o/p stage.

BLOCK DIAGRAM: -

Audio signal generator

PAM Modulator IC-1496

PAM signal

Power supply

Sampling Pulse Generator

PAM MODULATION

PAM Signal

L. P. F.

Demodulator rectifier IC 741 4053

Demodulated Signal

PAM DEMODULATION

PROCEDURE:
Connect jumper PAM o/p from modulator section to PAM I/p jumper of demodulator section. Connect sine i/p terminal of sample and hold section to sine o/p terminal of audio frequency generator. Connect pulse o/p of sampling pulse generator to sampling pulse i/p of s/h circuit. Keep frequency course switches of audio frequency generator to 2khz. Connect C.R.O. at o/p signal of audio generator adjust amplitude of sine wave to 2v(p-p). Observe PAM signal. Connect CRO channel-2 at demodulated o/p of this section. Vary the amplitude of sine wave modulating signal. Vary frequency of that and see the effect. Observe demodulated signal also.

CONCLUSION: -

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PRACTICAL 6
AIM:-To study Pulse Position Modulation & demodulation. APPARATUS: - PPM kit no: COM108, probes , CRO, etc. THEORY: The PPM modulation & demodulation system consist of following sections.
1. Input audio amplifier section. 2. Modulating audio signal generator section 3. sampling pulse generator section 4. pulse position modulator section comparator & multiplexer 5. DC voltage source 6. Pulse position demodulator section 7. output audio amplifier section 8. Power supply section (1) Input audio amplifier section .:The input audio amplifier circuit consists of pre-amplifier & output amplifier stage .This section is used to amplify low level audio signal coming from mike/loudspeaker & give it to A.M. modulator section for live A.M. modulation. The pre-amplifier consist of one transistor Q1.The input signal from mike is connected to the base of Q1 through coupling capacitor .The amplified audio signal obtained at the collector of Q1 is given to the output driver amplifier consisting of IC 810 at pin 10 through volume control pot p1. The I.C. 810 perform the function of the audio amplifier, driver & the output stage .This is amplified by the I.C. & the output is available at the pin 16 of the I.C. The feedback is given from the output to the emitter .The gain of the amplifier depends on this feed-back. (2) Modulating audio signal generator section:I.C. 1 (ICL 8038) waveform generator I.C. is used generate sine wave signal. The frequency range is 20 Hz to 20khz.the sine wave output signal is available at pin2 of I.C. & it is then amplified by i.c.4 .The amplified output is available output terminals. Pot P2 is used to vary the amplified of sine wave signal. The output amplitude varies from 0 to 5vpp. (3) Sampling pulse generator section:-

(4)

(5)

(6)

(7)

To generate PPM signal comparator & multiplexer circuit is used .The sampling saw tooth waveform & two 90 degree out of phase pulse signals are required to provide sampling signal to comparator & multiplexer circuit. This section is based on voltage controlled oscillator I.C.CD4046B .The pulse frequency is determined by varying control voltage of VCO at PIN 9 of this I.C. The pulse shape of this changed to saw tooth waveform by using mono-stable I.C. LF356.The output is available at PIN6.The pulse frequency can be varied from 3KHZ to 40 kHz. Pulse position modulator section:To generate PPM signal comparator & multiplexer circuit is used .Here LM311 is used as comparator & I.C. CD4053B is used as multiplexer. The modulating input signal is given to one input of the comparator. The sampling saw tooth waveform is given to second input of the comparator .The output signal of comparator is given to I.C. 4013 & 4528 in sampling generator section to produce 2 signals. The output of the comparator is given to the multiplexer .The multiplexer produce PPM signal with the help of the 2 signals taken out through the attenuator. DC voltage source:To see the effect of dc voltages on PAM modulated -5v to +5v voltage is required .These voltages are obtained from +15v & -15v DC power supply by potential divider made of 100K presets & variable pot. By varying this pot -5v to +5v DC are available. Pulse position demodulator section:This section is based on comparator, divider & low pass filter.The PPM signal is given to one input of comparator. The 1 volt DC reference signal to second input of comparator. This 1 volt reference signals generated by IC 741 by setting 4K7 preset. The comparator generates saw tooth signal at its output .The frequency of this signal is than divided by two by IC 4013. Then the output of divider is given to low pass filter made of two 741 IC. The low pass filter passes only low frequencies up to 3.4khz & reduces all other frequencies .Thus this removed high frequency quantization noise of PPM signal .By removing high frequency we recover original modulating signal. Output audio amplifier section:The IC 810 performs the function of the audio amplifier , driver & output stage. The amplified signal obtained from the transistor given at audio input of the IC. The feedback is given from the output to the emitter of the pre amplifier transistor . The gain of amplifier depends on this feed back .

BLOCK DIAGRAM:

Audio Gener ator

D.C. Generator

PPM Modulator

Demodulator

Power Supply

Sampling pulse generator

PROCEDURE:
1) Connect CRO at channel 1 to observe the sine wave o/p from Audio Frequency Generator Section (IC 8038).Adjust its amplitude and frequency say 1 v p-p and 1Khz respectively. 2) Keep Frequency port of Sampling Pulse Generator Section(IC 4046- VCO) at its mid position. 3) Also keep pulse width port of the same section at its mid position to adjust its duty cycle. 4) Observe the o/p sampling pulse with its duty cycle at channel 2 of CRO. 5) The Sampling pulse is converted to equivalent saw tooth w/f using mono stable multi vibrator IC LF356. 6) Observe the Saw tooth o/p at CRO channel 2. 7) Observe the modulated o/p at Pulse Position Modulator Section which uses IC LM 311 as a comparator. Now vary the pulse width of sampling pulse and see the effect on the phase of PPM o/p. 8) Observe the change in the phase shift of the modulated o/p with respect to the change in the amplitude of the modulating signal. 9) Observe the Demodulated o/p at channel 2 of CRO from Pulse Position Demodulator Section comprising of Comparator IC 741, Driver and a Low Pass Filter. 10) Measure the amplitude and frequency of demodulated o/p with change in the amplitude and frequency of modulating signal respectively. 11) Finally it is seen that Error in the recovered o/p Increases with Increase in Signal Amplitude Error in the recovered o/p Increases with Increase in Signal Frequency Error in the recovered o/p Decreases with Increase in Sampling Pulse Frequency Error in the recovered o/p Increases with Decrease in Sampling Pulse Width

CONCLUSION:

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PRACTICAL 7
AIM: To Study Pulse Code Modulation and Demodulation. THEORY:
It is a process in which analog signals are converted into digital form. The analog signal is represented by a series of pulses and non pulses (1s and 0s). Need for PCM: The stream of pulses and non-pulses is not easily affected by interference and noise. Even in the presence of a noise, the presence of a pulse can be realized. Digital signals are easy to be processed by cheap standard techniques. This makes it practical to implement complicated communication systems such as telephone networks. The amplitude, duration, and timing of a series of pulses are controlled in PCM because a pulse coded signal being in digital form is simple. For example, if an audio analog signal is to be transmitted, it must first be converted into a PAM signal by regular sampling of its amplitude. BASIC STAGES IN THE GENERATION OF PCM:

Filtering Sampling Quantizing Encoding

Then after, at the demodulator side decoding and filtering processes are done 1. FILTERING:

The filtering stage removes frequency above the highest signal frequencies. These frequencies if not removed may cause problem when the signal is going through the stage of sampling. The analog signal m(t) is passed through an anti-aliasing filter.

2. SAMPLING: The low pass filtered signal is then converted into a time discrete signal by means of a Sampler. This sampled signal is called a pulse amplitude modulated signal (PAM). Sampling of a waveform means determining instantaneous amplitude of signal at fixed intervals. Sampling period (Ts): the periodic time of the pulse train is known as the sampling period. It is the period from the beginning of one sample to the next, not the pulse duration. The sampling frequency is Fs =1/Ts

Sampling theorem: It states that: The sampling frequency must be at least twice the highest frequency in the modulating signal. Fs >=2W If the sampling condition is not met, parts of the spectra overlap and once such an overlap occurs the spectra can no longer be separated by filtering.

For most digital converters, sampling is performed by a sample and hold (S/H) circuit. Note that if the input voltage to be digitized is varying, a sample-and-hold circuit is mandatory. A/D converters with more precision cannot give their advertised accuracy without a sample-and-hold. A pretty good S/H can be put together from discrete components, as shown in the fig. The LF412 is a dual FET-input op-amp with very small input bias currents. Bipolar op-amps are not suitable, because the input base currents are too large. The circuit is basically two unity-gain buffers, with a hold capacitor between them, and a switch to disconnect the input. The switch is made from a MOSFET, which does very well. Sampling is done in the positive half of the pulse (+V), and holding in the negative half cycle of the pulse. When the gate is connected to -V, the sampled voltage is held on a hold capacitor and the output voltage is frozen at that point, we note that it

droops slowly but steadily. When the gate is connected to +V, the voltage at the input appears at the output, and varies simultaneously. 3. QUANTIZATION:

The sampled analog signal must be converted from a voltage value to a binary number that the computer can read. The conversion from infinitely precise amplitude to a binary number is called quantization. It may also be defined as rounding off the values of the flat-top samples to certain predetermined levels. In the Quantization process, the total signal range is divided into a number of sub-ranges. Each sub-range has its mid value designated as standard for that range. Comparators are used to determine which sub-range a given pulse amplitude is in and code for that sub-range is generated. There may be 8, 12 or 16 sub-ranges. Quantization Error: There is always some error associated with the quantization of a continuous signal. The difference between quantized level and the analog input is called as Quantization error which appears as noise and is called as Quantization noise. Ideally, the maximum quantization error is 0.5 least significant bits (LSBs), and over the full input range, the average quantization error is zero. The Signal to Quantization noise can be determined by subtracting the recovered from the Original signal and putting them in ratio together. The ratio between the noise and the signal is called the Signal to Quantization noise. 4. ENCODING: The quantized sample values are coded. Any plan for representing each of this discrete set of values as s particular arrangement of discrete event in a code is called a code. One of the discrete events in a code is called a code element or symbol. In a binary code, each symbol may is either of tow distinct values or kinds such as the presence or absence of a pulse. The two symbols of a binary code results in the maximum advantage over the effects of noise in a transmission medium. Ti is also easy to regenerate.

With an n bit (binary digit) binary code, one can represent a total of 2n distinct numbers. There are several ways by which binary symbols 1 and 0 can be represented by electrical signals. These constitute the PCM signal. 3. DECODING:

At the receiver, the PCM pulse may be reshaped. The reshaped clean pulse are regrouped into code words in the receiver and decoded into a quantized PCM signal. The decoding process involves generating a pulse the amplitude of which us the linear sum of all the pulses in the code word, each pulse weighted by its place-value (2 for a binary code) in the code. 4. FILTERING: The final operation in the receiving is to recover the signal wave by passing the decoder output through a low-pass reconstruction filter whose cut off frequency is equal to the message bandwidth Wm Assuming that transmission path is error free, the recovered signal includes noise with the exception of the initial distortion introduced by quantization process. STUDY OF CIRCUITS OF PULSE CODE MODULATION /DEMODULATION The PCM Modulation and Demodulation System consist of following sections. 1. Input Audio Amplifier 2. DC source 3. Modulating Audio Signal Generator 4. Sampling Pulse Generator 5. PCM Modulator (Encoder) section 6. Pulse Code Demodulator (Decoder) section 7. Low pass filter 8. Output Audio amplifier

9. Power supply

(1) INPUT AUDIO AMPLIFIER: It consists of pre-determined and output amplifier section. This section is used to amplify low level audio signal coming from mike/loud speaker. The pre-amplifier consists of one transistor Q1. Transistor Q1 (BC 148B) is connected in C-E configuration. The input signal form mike is amplified audio signal obtained at the collector of Q1 is given to the output driver amplifier consisting of IC 810 at pin 10 through volume control pot P1. The IC 810 performs the functions of the audio amplifier, driver and the output stage. The amplified signal obtained from the transistor Q1 is given at the audio input (pin 10) of the IC. This is amplified by the IC and the output is available at the pin 16 of the IC. The feedback is given from the output to the emitter of preamplifier transistor in the IC (through a resistor 4K provided internally). The feedback voltages develops over R^(22E) , C12 (220/10) being used for d.c blocking. The gain of amplifier depends on this feedback. The amplified output is available at pin 16 and intern at output terminals. (2) DC SOURCE SECTION:To see the effect of DC voltages on PCM modulated signal 5V to +5V DC voltage is required. These voltages are obtained from +15V and 15V DC of Power supply by potential divider made of 47K presets and variable pot p 401 (100K pot). By varying this pot 5V to +5V DC are available at VARDC socket. (3) MODULATING AUDIO SIGNAL GENERATOR SECTION:IC 1 (ICL 8038) waveform generator IC is used generate sine wave signal. Pot P2 (22K) is used to vary its frequency. The frequency range is 20Hz to 20 KHz. Presets Pr1 and pr2 are adjusted for proper peaks of sine wave signal. Pr3 is used to adjust duty cycle. The sine wave output signal is available at pin 2 of IC1 and it is then amplified by IC4. The amplified output is available output terminals. Pot P2 is used to vary the amplitude of sine wave Signal. The switch S1 is used for course variation of frequency. The output amplitude varies from 0 to 15Vpp. (4) SAMPLING PULSE GENERATOR:The faction of this stage to generate sampling pulses. The basic clock is generated by NOT gates made from NAND gates (IC 4011). The clock frequency depends on R-C network selected by Switch and it can be varied by frequency pot. On FAST position the clock frequency available is 1.1 MHz while selected SLOW position 1 to 2 Hz counter IC (4040) in modulated section. (5) STUDY OF PCM MODULATOR (ENCODER) SECTION:The 1.28 MHz sampling clock from sampling pulse generator is applied to binary counter IC-4040. Assuming 3 bit/4 bit bit select switch in the 4 bit position , then the output signals of 4040 counter viz. Q2, Q3, Q4, Q5 (at pins 2, 3,5 and 6) are current weighted by resistors 10K, 22K, 39K and 82K and are summed in resistor 1K5 to produce Ramp waveform of 16 steps. The ramp is symmetrical about 0 volts. The modulating analog signal is connected at input connector

from audio sine wave generator section. This modulating signal is buffered by buffer IC741 and applied to one input of comparator IC 529 at pin 3 through 15K resistor. The ramp waveform is connected to other input of comparator 529 at pin 4. The comparator output is +6V as long as the input voltage exceeds the ramp voltage, but when this situation is reversed the output changes to 0V. These two levels are converted to those acceptable by the remainder of the circuit, which operates with thresholds of +/- 2.4 V, by zener (2.7V zener). The output of comparator is strobed by AND gate (IC 4081 pin 8, 9, 10) which derives its other input from And gate (IC 4081 pin 13, 12, 11). This occurs once for every step of the ramp. So long as the comparator output is high the strobe cause o/p at pin 10 (4081) to go high so that the data presented at inputs D0, D1, D2, D3 of latch (IC 4042 pin 4, 7, 13, 14) is latched to the outputs Q0, Q1, Q3 ( IC 4042 pin 2, 10, 11, 1). However, when the ramp voltage exceeds the input and the comparator goes low, the latch ceases to be clocked and the previous digital number is stored. When the counter is full, all inputs to NAND gate ( IC 4068) are high and the output goes low ( pin 13 of 4068), consequently the Enable input of Shift register ( IC 4021 pin 9) goes high and the parallel data present on inputs P0 P7 (IC 4021 pin 5, 10, 13, 14 ) are loaded. This data is PCM 4 bit data and are given to NOT gates (IC 4049) for display on LED as well as these data are clocked at pin 3in several form, the clock signal forms the PCM output and is available at PCM o/p socket. A synchronization output is provided to assist in viewing the PCM output with CRO. Operation of bit select switch selects 3-btis operation by providing a permanently low fourth bit. Here logic 1 is +6 Volt and logic 0 is 6 Volt. The PCM data streams consists of two guard bits (Low), four data bits (High or Low as per input signal), two guard bits (Low) and eight bit long Sync pulse (High). (6) STUDY OF PCM DEMODULATOR (DECODER) SECTION:The decoder also incorporates a sampling clock made of NAND gates ( IC 4011 ). This clock is gated by the sync detector consisting of IC 3140 (used as comparator ) and transistor Q1 ( 2N2907A ). When the PCM input is high Q1 is turned OFF and capacitor C1- 560pf charges through resistor R1- 33K, when the PCM input is low , Q1 conducts discharging C1-560pf. The time constant C1 and R1 is selected so that the voltage on the comparator ( IC 3140 ) reaches the threshold set by two 33K resistors at pin 2 of 3140 after six consecutive high pulses. When the threshold is reached ( during the sync pulse only ), the comparator output goes low and inhibits the clock. At the end of the sync pulse C1 is discharged, the comparator output returns to the high state, the clock starts and the counter 4040 is reset. The clock output is divided by two by the D flip-flop 4013. This divided clock is regated by AND gate to form relatively narrow clock pulses with their leading edge delayed from the PCM data transition by half a pulse width. This pulse is further gated by AND gate and is used to clock the shift resister (IC 4015). The counter 4040 and inverter (IC 4011 pin 8, 9, 10) allow eight clock pulses to occur before AND gate (IC 4081 pin 4, 5, 6) is inhibited. Thus the two sets of guard bits and four data bits are loaded into shift register (4015). The data bits Q0, Q1, Q2, and Q3 (at pin 3, 10, 12, 13, 11) on the ninth clock pulse. The output of the latch is connected to weighted resistors identical to those in the decoder which sum a current into resistor R9-1K5. The voltage developed across R9-1K5 is amplified to restore the original level. (7) LOW PASS FILTER:-

This section passes only low frequencies up to 3.4 KHz and reduces all other frequencies. This filter is required to remove high frequency quantization error noise from demodulated signal. This section is based on IC 741 and R-C circuits. (8) OUTPUT AUDIO AMPLIFIER SECTION:The IC 810 performs the functions of the audio amplifier, driver and the output stage. The amplified signal obtained from the transistor Q1 is given at the audio input (pin 10) of the IV. This is amplified by the IC and the output is available at the pin 16 of the IC. The feedback is given from the output to the emitter of preamplifier transistor in the IC. The feedback voltages develops over R6 (22E), C12 (220/10) being used for d.c.blocking. The gain of amplifier depends on this feed-back. The amplified output is available at pin 16 and intern at LS terminals. (9) POWER SUPPLY SECTION:The regulated power supply is used for different supply voltages. Following output D.C.Voltages are required to operate PCM Modulation demodulation system. +15V, 250mA -15V, 250mA +6V, 250mA -6V, 250mA Three terminal regulators are used for different output voltages i.e. IC 7805 for +5V, IC 7815 for +15V, IC 7905 for -5V, IC 7915 for -15V, These ICs are supplied different dc input voltages by two half wave rectifiers consisting of D1D4 and D5-D8 and C1, C2, C3, C4. The capacitors at each input and each output are for filtering purpose. SW 301 is main ON/OFF Switch.

PROCEDURE:
1) Connect CRO at channel 1 to observe the sine wave o/p from Audio Frequency Generator Section (IC 8038).Adjust its amplitude and frequency say 1 v p-p and 1 KHz respectively. 2) Keep Frequency port of Sampling Pulse Generator Section(IC 4011) at its mid position. 3) Connect CRO channel 2 and observe the Sampling Clock pulse from Sampling Pulse Generator Section whose clock frequency depends on R-C n/w selected by s/w on FAST position and the available clock frequency is 1.1 MHz to 1.4 MHz. While selecting SLOW position 1 to 2 Hz clock frequency is generated. 4) This sampling clock is given to Counter IC 4040 at PCM modulator section which produces stair case o/p of quantization levels 8 & 16 depending on switch position of 3 or 4 bit mode. See the ramp (stair case o/p) o/p at Ramp o/p socket. 5) Externally connect one jumper to interconnect o/p of modulating signal to the i/p of PCM modulator section. 6) Connect CRO channel 2 at PCM o/p from modulator section.

7) Observe the std. bit stream of PCM o/p at CRO with 8 quantization levels by selecting s/w in position of 3 bit. Follow the same step to obtain 16 quantization levels by selecting s/w in position of 4 bit. 8) See the effect on Bit stream of PCM o/p with change in s/w position from 3 bit to 4 bit.One thing is to be checked i.e. observe the change in bit stream of PCM o/p with the change in the amplitude of the modulating signal. 9) Also check that by keeping Sampling pulse generator section in SLOW mode and observe the change in bit pattern on LED (in terms of different quantization levels) as per bit s/w mode. 10) Observe the raw o/p (without filtering) at PCM Demodulator section on CRO channel 2. 11) Finally see the Demodulated o/p sine wave at LPF section of PCM Demodulator.

CONCLUSION:

Date:

PRACTICAL 8
AIM: - To study about TDM Modulation & Demodulation. APPARATUS: -TDM Kit, CRO, Probes, wires, etc. THEORY: - TDM is time division multiplexing. In which we use pulse modulation which
permits the simultaneous transmission of several signals on a time sharing basis. Because a pulse modulated signal occupies only a part of the channel time, we can transmit several pulse-modulated signals on the same channel by interweaving them. Fig shows the TDM of two PAM signals. In this manner we can several signals on the same channel on the same channel by reducing pulse width.

g1(t) t

g2(t)

Thus this method is suitable when a signal is in the form of a pulse train. The pulses are made narrower, and the spaces are left between pulses are used for pulses from other signals. Thus, in effect, the transmission time is shared by a number of signals by interleaving the pulse trains of various signals in specified order. At the receiver, the pulse trains corresponding to various signals are separated. In our kit of TDM there will be 5 main parts. a) Input signal generators (IC 8083): Here we use four types of signals for multiplexing. 1) Sine wave generator, 2) Square wave generator, 3) Triangular wave generator, 4) Audio wave generator. We can very the amplitude and frequency of these signals. b) Sampling pulse generator: To generate PAM-TDM signal sample & hold circuit is used. c) Time division multiplexing section: To generate TDM signal we use analog multiplexer IC 4052. The sampling pulse generator is required to provide multiplexing of 4 different signals. These 4 signals are giving to I/P of multiplexer IC 4053. Here we give sample signal as selection line which has very high frequency compare to the input signal frequency. So there are 4 selection states. 1) during 00 sine wave is transmit 2) during 01 square wave is transmit 3) during 10 triangular wave is transmit 4) during 11 audio wave is transmit d) Time division demultiplexing section: Here we use same IC 4053, which is also use for demultiplexing purpose. In which applied o/p signal of multiplexing section as a i/p signal of the demultiplexing. And also we applied same sampling pulses as selection line of this IC. By this type of arrangement at the o/p side we get 4 original signal but in the form of PAM. e) Low pass filter section: This section is use for get pure original signal from their PAM signal. So after low pass filter we get original signals. Thus we can generate TDM signal & from it we can also generate original signal.

PROCEDURE: 1) 2) 3) 4) 5) 6) As shown in circuit diagram we apply four different signals having different amplitude & frequencies of sine wave, square wave, triangular wave and audio wave at multiplexing circuit. Draw graph of all the input signals. Here we also apply the pulse at varies frequencies to the TDM. Then observe the o/p of multiplexing section & draw the graph of it. Also see the o/p of varies signal from demultiplexing section. Then observe all o/p & i/p signal at same time & compare both.

CONCLUSION: -

BLOCK DIAGRAM: -(TDM)

Sine Wave

TDM O/P

DemodulatedO/ P

Signal Generator

Multiplexer
TDM I/P Sampling Pulse Generator

Demultiplexer

Power supply

Input Audio Amplifier

O/P Audio Amplifier

Date:

PRACTICAL 9
AIM: To study theory of ASK Modulation and Demodulation APPRATUS: ASK kit ,CRO ,connecting wires. THRORY:
In order to communicate information over a distance some form of modulated carrier signal must be used. The carrier is often a high freq alternating signal Amplitude shift keying is the simplest way of shifting the frequency spectrum of a signal from base band to some other band of frequency spectrum. It uses an auxillary signal called carrier Amplitude modulation means alternating the amplitude of the carrier signal in according with the base band signal. When the signal switch between two distinct level this is called keying. Thus switching a carrier on for 1 and off for 0 is called amplitude shift keying. The simplest and most common form of ASK operates as a switch, using the presence of a carrier wave to indicate a binary one and its absence to indicate a binary zero. This type of modulation is called on-off keying, and is used at radio frequencies to transmit morex code (referred to as continuous wave operation).

More sophisticated encoding schemes have been developed which represent data in groups using additional amplitude levels. For instance, a four-level encoding scheme can represent two bits with each shift in amplitude; an eight-level scheme can represent three bits; and so on. These forms of amplitude-shift keying require a high signal-to-noise ratio for their recovery, as by their nature much of the signal is transmitted at reduced power.

Here is a diagram showing the ideal model for an transmission system using an ASK modulation: It can be divided into three blocks. The first one represents the transmitter, the second one is a linear model of the effects of the channel, the third one shows the structure of the receiver. The following notation is used:ht(t) is the carrier signal for the transmission

hc(t) is the impulse response of the channel n(t) is the noise introduced by the channel hr(t) is the filter at the receiver L is the number of levels that are used for transmission Ts is the time between the generation of two symbols

MODULATOR: There are two approaches to generating ASK waveform. One technique
starts with the base band signal and use this to amplitude modulate a sinusoidal carrier Since the base band signal consist of distinct waveform segment. AM wave also consists of distinct modulated segments. Another approaches is to generate the AM wave directly with out first forming the base band signal we need simply switch an oscillator on and off as shown in fig. 1
J1

R1

Key = Space

1kohm

V1
1V 1000Hz 0Deg

R2
1kohm

[modulator] As shown in fig. 2 here we use op-amp resistor etc to works as wein bridge oscillator to generate the carrier wave according to amplitude of digital signal. Here we use amplifier to amplify our input

5V

VCCR3

1kohm

R7
10kohm
7 3 VS+ BAL1 BAL2 2 VS-

R2
1kohm

V3
12V

U1
1 5 6

R1 V1
1kohm 100Hz 1V

Q1
2SC2120

C1
10nF

V2
-12V
4

741

R4
1kohm

R5
1kohm

MODULATOR

DEMODULATOR: Demodulator of digital-modulation signals is similar to that analogmodulated signals. ASK can demodulated coherently or non coherently the coherent detector use elaborate equipment and has superior performance , especially when the signal power is low. While envelop detector is used for higher signal noise ratio. In fig 1 we used the diode clamping circuit, filter, op-amp, to get the original bas signal.
VCC VCC
-5V

-5V
7 3 VS+ BAL1 BAL2 2 VS4

U1
1 5 6

D1
1N4001GP

741

R1
1kohm

C1
1.0uF

VCC
-5V

DEMODULATOR

GRAPH:

CIRCUIT DETAIL:
The block diagram of ASK modulation and demodulation consist of following section (1) MODULATING NRZ DATA SIGNAL GENERATOR SECTION: If signal IC is used to generate NFZ data. This section generate bit clock word and NRZ data (2) R.F CARRIER OSCILLATOR SECTION: Frequency range is 200 kHz to 1 MHz o/p amplitude vary from 0 to 10v pp (3) DOUBLED BALANCED AMPLITUDE MODULATOR SECTION: IC1 is used as balanced modulator. The modulating audio signal is connected at pin:1 through buffer. (4) ASK MODULATOR: (A) ASK DEMODULATION BY DIODE DETECTOR: This ckt consisting of diode 0A79 and capacitor c1,c2 and r1.It works as an envelop detector etc. (B) ASK MODULATION BY SQURE LOW DETECTOR: This section is simillar as ask balanced modulator section. The diff is only that both input pin1&8 are given from ask modulating signal.

(5) DATA SQURE CKT: The ckt consisting of comparetor. The o/p of comparator Is pure digital signal with sharp rising and falling edge similar to input NRZ data.

Digital data generator nrz

ck bk wk

ASK O/P

Modulating i/p

Balance Modulator RF I/P

Diode detector

READ DATA

Power supply

Product detector

RAW DATA O/P

Data squrer

RF I/P

R.F Carriear generator

BLOCK DIAGRAM

PROCEDURE:
1. Connect NRZ data signal from data generator to the modulating terminal of balance modulator 2. Observe the ask modulator o/p at o/p of balance modulator. see the offset on the ask modulator o/p by varying the bit in NRZ data by selector switches. 3. Connected the modulated o/p of balance modulator to i/p of envelop detector. 4. Observe the demodulated NRZ data wave at the diode detector o/p. 5. Vary the bits in NRZ data by selector switches and observe raw data.

CONSIDERATION:

NRZ base band data are generated from NRZ Data Signal Generator Section which uses Digital ICs. See the o/p base band signal pattern on LED display as well as it can also be seen on CRO channel 1. By varying the RF carrier frequency generator knob set its frequency at mid position and amplitude at 10 v p-p. Observe the ASK modulated o/p at Balanced Modulator section. Also observe the effect on the modulated signal by varying the i/p NRZ bit pattern. Now observe the demodulated raw NRZ data signal at the o/p of Diode Detector section. Now connect the o/p raw data to the i/p to Data Squarer ckt.( IC 311) Which produces the pure NRZ data stream as an o/p.

CONCLUSION:

BLOCK DIAGRAM:

Level shifter +

I/p Sequence

FSK signal

Inverter

Level shifter

Frequency shift keying

Date:

PRACTICAL 10
AIM: To study Frequency Shift Keying. THEORY: Frequency Shift Keying is the most common form of digital modulation in
high frequency radio spectrum, and it has important applications in telephone circuits. In this tutorial both modulation and demodulation schemes will be discussed. Binary FSK: -It is a modulation scheme used to send digital information between digital equipment such as tele printers and computers. -The data are transmitted by shifting the frequency of a continuous carrier in a binary manner to one or the other of two discrete frequencies. One frequency s designated as a mark frequency and the other as a space frequency. -The mark and space corresponds to binary one and zero, respectively. Mark corresponds to higher radio frequency. Fig1 shows a relationship between the data and the transmitted signal. Frequency measurement of the FSK signals are usually stated in terms of shift and center frequency. Shifts are usually in the range of 50 to 1000 Hz. The nominal frequency is the midway between mark and space frequencies.

-FSK can be transmitted coherently and noncoherently. Coherently implies that phase of each mark or space tone has a fixed phase relationship with respect to reference. -Non coherent FSK is used for majority of the FSK transmissions. There are two schemes to transmit data with the use of FSK. 1. 2. Synchronous transmission have mark-to-space and space-to-mark transition in a Asynchronous signals do not require a reference clock but instead rely on special synchronism with a reference block. bit patterns to control timing during decoding. A. FSK DEMODULATOR: A simplified diagram using only FSK generator and FSK demodulator is shown in fig given below. The FSK generator is formed by using a 555 as an astable multivibrator whose frequency is controlled by the state of transistor Q1.The output frequency of the FSK generator is depends on the logic state of data input.150 HZ is the standard frequency at which the data are commonly transmitted. When the input logic is 1,Q1 is off. Under these conditions 555 works in its normal mode as an astable multivibrator, that is capacitor charges through RA and RB and capacitor discharge through RB. The values of Ra and Rb and C are selected so that Fo presents a mark frequency. On the other hand, when the input signal is at logic 0,Q1 is on which in turn connects the resistance Rc across Ra. This action reduces the charging time of the capacitor and increases the output frequency. The difference between FSK signals of these two frequencies is called the frequency shift. The output of the FSK generator is given to the 565 FSK demodulator circuit

B. FSK DEMODULATOR: The FSK demodulation methods can be divided in to two major categories. 1. FM detector demodulators 2. FILTER-type demodulators Filter-type demodulators attempt to match the FSK parameters to the demodulator configuration to optimize demodulator error performance. The classic matched filter demodulator is optimal for coherent FSK in white gaussian noise interference. A block diagram of a simple, matched filter demodulator for coherent FSK is shown in a fig.1 In the demodulator, the output of the matched filters is compared, and if the output from the mark filter is larger than that from the space filter, a decision is made that a mark signal is transmitted.

Optimum demodulation of non-coherent FSK can be achieved by envelope detection of the signal filter outputs in a filter-type demodulator. A demodulator of this type is shown in fig2. The output of the mark and space filters are envelope-detected and then compared to determine which has greater magnitude. Here the phase information is not required. With the right filter shape, performance of this type of demodulator approaches the theoretical optimum for noncoherent FSK. Bandpass filter centered about the desired mark and space filters tone.

For the general case of noncoherent FSK with non-white noise interference, the problem of optimum filter design is more complex. Two common types of non-white noise interference are adjacent. 1. Channel interference 2. CW interference. Because of the unpredictable nature of this kind of interference, it is desirable to create a filter type which performs well with both white and non-white interference. To minimize the effect of of non-white interference, it is desirable to use a Bandpass filter with relatively steep attenuation skirts and without the side lobes that are characteristic of the sinc-function filter. It is also desirable that the filter perform well in white noise. For each filter shape there is an optimum bandwidth. In, general, if the filter bandwidth is too wide, excess noise energy will be included. If the filter band width is too narrow, consecutive signal elements will interfere with each other.This is called intersymbol interference. To specify filter characteristics, it is conventional to talk in terms of the filters 3db band width.The optimum 3db bandwidth varies directly with the signal keying speed and inversely with element length.

CIRCUIT DIAGRAM:
VCC
5V

Q1
BJT_NPN_VIRTUAL

R3
7.5kohm

U1

4 RST 7 6 DIS

VCC 3

R1
240ohm XFG1

OUT

XSC1 G A B T

R2
3.6kohm

2 5

THR TRI CON GND 1

555_VIRTUAL

C1
10nF

C2
10nF

PROCEDURE:
NRZ base band data are generated from NRZ Data Signal Generator Section which uses Digital ICs. See the o/p base band signal pattern on LED display as well as it can also be seen on CRO channel 1.By varying the RF carrier frequency generator knob set its frequency at mid position and amplitude at 10 v p-p. 1) 2) 3) 4) Observe the FSK modulated o/p at FSK Modulator section which uses IC 8038 wave Also observe the effect on the modulated signal by varying the i/p NRZ bit pattern. Now observe the demodulated raw NRZ data signal at the o/p of FSK Demodulator Now connect the o/p raw data to the i/p to Data Squarer ckt. ( IC 311) Which produces form generator.

section using resonant Phase Shift Detector IC 1496. the pure NRZ data stream as an o/p.

CONCLUSION:

Date:

PRACTICAL 11
AIM: To study the Phase Shift Keying. THEORY:
Phase shift keying (PSK): is a digital modulation scheme that convey data by changing, or modulating, the phase of a reference signal. The PSK modulation and demodulation consist of following sections. 1) INPUT DATA GENERATOR SECTION: To generate psk signal i/p digital modulating signal is required. This digital data signal is generated by this section. It consist of many digital ics. The carrier clock of 1 to 15 khz is generated by ic 741 by pusing different push switches. Different data word can be generated . This data word is called as nrz data. Then this nrz datais converted to produce in phase data by ic-4013. 2) CARRIER PHASE SPLLITER: The function of this section is to produce 90 out of phase sine wave signal referred as 1 and 2 are required to provide sampling signal from one carrier signal. The carrier signal is applied at the i/p of this section from data generator section. Its phase is shifted by phase splitter. 3) PSK MODULATION USING BALANCED MODULATOR:

Two balanced modulator are used to generator PSK signal. The modulating audio signal is connected at ring through transistor Q1, first balanced modulator is given two i/p signals as Q1 from carrier phase splitter and in phase signal from data generate second balance modulator is given two i/p signals as 2 and dc bias from carrier phase splitter.

4) PSK DEMODULATOR USING BALANCED MODULATOR: The balanced modulator is used to demodulate psk signal. The PSK modulated signal is given at first i/p. the carrier signal is given at second i/p from carrier phase splitter section. 5) DATA SQUARER SECTION: This section is based on comparator i.e .at accept raw data. From the demodulator section provide pure digital recovered data at its o/p. the bias is used to set threshold if zero level for digital data.

CIRCUIT DIAGRAM:

PROCEDURE: 1) NRZ base band data are generated from NRZ Data Signal Generator Section which uses Digital ICs. See the o/p base band signal pattern on LED display as well as it can also be seen on CRO channel 1. 2) Observe the RF carrier signal (High freq. sine wave) at RF Carrier Generator Section on CRO at ch. 2. 3) Observe the PSK modulated o/p at PSK Modulator section which uses Balanced Modulator IC 1496. 4) Also observe the effect on the modulated signal by varying the i/p NRZ bit pattern. 5) Now observe the demodulated raw NRZ data signal at the o/p of PSK Demodulator section using Balanced Demodulator IC 1496. (Carrier Recovery ckt. is required to recover the same carrier signal for demodulation purpose) 6) Now connect the o/p raw data to the i/p to Data Recovery ckt. ( IC 311) Which produces the pure NRZ data stream as an o/p. CONCLUSION: