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SUDHARSAN ENGINEERING COLLEGE

SATHIYAMANGALAM, PUDUKKOTTAI

DEPARTMENT OF ECE

LAB MANUAL

ACADEMIC YEAR(2013-2014)

SUBJECT CODE/NAME : EC2307-COMMUNICATION SYSTEM

LABORATORY

YEAR/SEM : III/V

PREPARED BY,

Mrs. M.SUDHA , AP/ECE

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EC2307-COMMUNICATION SYSTEMS LABORATORY

SYLLABUS

1. Amplitude modulation and demodulation

2. Frequency Modulation and Demodulation

3. Pulse Modulation-PAM/PPM/PWM

4. Pulse Code Modulation

5. Delta Modulation, Adaptive Delta Modulation

6. Digital modulation and Demodulation-ASK,FSK,QPSK,PSK

7. Designing ,Assembling and Testing of Pre-emphasis/De-emphasis

circuits

8. PLL and Frequency Synthesizer

9. Line Coding

10. Error Control Coding using MATLAB

11. Sampling and Time Division Multiplexing

12. Frequency Division Multiplexing

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INDEX

Ex.No

Date

Name of the Experiment

Page.No

Marks

Signature

1.

Amplitude Modulation and

Demodulation

2.

Frequency Modulation and

Demodulation

3.

Pulse Modulation-PAM/PPM/PWM

4.

Pulse Code Modulation

5.

Delta Modulation, Adaptive Delta

Modulation

6(A).

6(B).

Digital modulation and Demodulation-

ASK,FSK,QPSK,PSK

Modulation & Demodulation using

MATLAB

7.

Designing ,Assembling and Testing of

Pre-emphasis/De-emphasis circuits

8.

PLL and Frequency Synthesizer

9.

Line Coding

10.

Error Control Coding using MATLAB

11.

Sampling and Time Division

Multiplexing

12.

Frequency Division Multiplexing

using MATLAB

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EX.NO: 1 AMPLITUDE MODULATION AND DEMODULATION

DATE: TECHNIQUES

AIM:

To study the function of Amplitude modulation and demodulation techniques.

APPARATUS REQUIRED:

1. ST2201&2202 TRAINER KIT

2. 2mm BANNANA CABLE

3. CRO

THEORY:

Amplitude modulation (AM) is a technique used in electronic communication, most

commonly for transmitting information via a radio carrier wave. AM works by varying the strength of

the transmitted signal in relation to the information being sent. For example, changes in signal

strength may be used to specify the sounds to be reproduced by a loudspeaker, or the light intensity of

television pixels. Contrast this with frequency modulation, in which the frequency is varied, and phase

modulation, in which the phase is varied in accordance to the modulating signal.

Demodulation is the act of extracting the original information-bearing signal from a

modulated carrier wave. A demodulator is an electronic circuit (or computer program in a software-

defined radio) that is used to recover the information content from the modulated carrier wave

CIRCUIT DIAGRAM:

PROCEDURE:

1. Ensure that the following initial conditions exist on the ST2201 board.

a. Audio oscillator's amplitude pot in full clockwise position.

b. Audio input select switch in INT position.

c. Mode switch in SSB position.

d. Output amplifier's gain pot in full clockwise position.

e. TX output select switch in ANT position.

f. Audio amplifier's volume pot in full counter-clockwise position.

g. Speaker switch in ON position.

h. On board antenna in vertical position, and fully extended.

2. Ensure that the following initial conditions exist on the ST2202 board.

a. RX input select switch in ANT position.

b. R.F amplifier's tuned circuit select switch in INT position.

c. R.F amplifier's gain pot in full clockwise position.

d. AGC switch in out position.

e. Detector switch in product position.

f. Audio amplifier's volume pot in fully counter clockwise position.

g. Speaker switch in 'ON' position.

h. Beat frequency oscillator switch in 'ON' position.

i. On - board antenna in vertical position, and fully extended.

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4. Turn on power to the modules.

5. On the ST2201 module, examine the transmitter's output signal (TP13)

6.Turn ST2201's amplitude pot (in the audio oscillator block) to its full counter clockwise

(minimum amplitude) position and note that amplitude of the monitored output signal from

ST2201 (at TP13) drops to zero

7. On the ST2202 module, monitor the output of the IF amplifier 2 block (TP28) and turn the

tuning dial until the amplitude of the monitored signal is at its greatest

8. On the ST2202 module, monitor the output of the product detector block (at TP37),

together with the output of the audio amplifier block (TP39), triggering the scope with the

later signal.

9. Turn the frequency pot in ST2201's audio oscillator block, throughout its range, noting

that the frequency of the tone generated by ST2202 remains close to that generated by

ST2201 for all pot positions.

10. With the receiver's tuning dial adjusted for correct demodulation, the transmitted signal is

obtained.

TABULATION:

Signal Amplitude Time

Message signal

Carrier signal

AM signal

Demodulated signal

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MODEL GRAPH:

RESULT:

Amplitude modulater and demodulater are constructed and its waveforms are analysed.

Percentage Modulation = Vmax- Vmin/ Vmax+Vmin=__________________.

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EX.NO: 2 FREQUENCY MODULATION AND DEMODULATION

DATE: TECHNIQUES

AIM:

1. To generate frequency modulated wave .

2. To Demodulate the modulated wave using envelope detector.

APPARATUS REQUIRED:

1. ST 2203 Trainer Kit.

2. CRO

3. Patch chords

THEORY:

In an AM System, the demodulator is designed to respond to changes in amplitude of

the received signal but in a FM receiver the demodulator is only watching for changes in

frequency and therefore ignores any changes in amplitude. Electrical noise thus has little or

no effect on a FM communication system. The bandwidth of the FM signal is very wide

compared with an AM transmission. Typical broadcast bandwidths are in the order of 250

KHz. This allows a much better sound quality, so signals like music sound significantly better

if frequency modulation is being used. When an FM demodulator is receiving an FM signal,

it follows the variations in frequency of the incoming signal and is said to lock on to the

received at the same time. The receiver 'lock on' to the stronger of the two signals and ignores

the other. This is called the 'capture effect' and it means that we can listen to an FM station on

a radio without interference from other stations.

PROCEDURE:

1. Ensure that the following initial conditions exist on the ST2202 board.

a. All Switched Faults in ‘Off’ condition.

b. Amplitude potentiometer (in mixer amplifier block) in fully clockwise

position.

c. VCO switch (in phase locked loop detector block) in ‘Off’ position.

2. Make the connections as shown in figure 13.

3. Switch On the power.

4. Turn the audio oscillator block's amplitude potentiometer to its fully clockwise position,

and examine the block's output TP1 on an Oscilloscope. This is the audio frequency sine

wave, which will be used as our modulating signal. Note that the sine wave's frequency can

be adjusted from about 300Hz to approximately 3.4 KHz, by adjusting the audio oscillator's

frequency potentiometer.

5. Connect the output socket of the audio oscillator block to the audio input socket of the

modulator circuit’s block.

6. Set the reactance / varactor switch to the varactor position. This switch selects the varactor

modulator and also disables the reactance modulator to prevent any interference between the

two circuits.

7. The output signal from the varactor modulator block appears at TP24 before being

buffered and amplified by the mixer/amplifier block, any capacitive loading (e.g. due to

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Oscilloscope probe) may slightly affect the modulators output frequency. In order to avoid

this problem we monitor the buffered FM output signal the mixer / amplifier block at TP34.

8. Put the varactor modulator's carrier frequency potentiometer in its midway position, and

then examine TP34. Note that it is a sine wave of approximately 1.2 Vpp, centered on 0V.

This is our FM carrier, and it is un-modulated since the varactor modulators audio input

signal has zero amplitude.

9. The amplitude of the FM carrier (at TP34) is adjustable by means of the mixer/amplifier

block's amplitude potentiometer, from zero to its potentiometer level. Try turning this

potentiometer slowly anticlockwise, and note that the amplitude of the FM signal can be

reduced to zero. Return the amplitude potentiometer to its fully clockwise position.

10. Try varying the carrier frequency potentiometer and observe the effects.

11. Also, see the effects of varying the amplitude and frequency potentiometer in the audio

oscillator block.

12. Turn the carrier frequency potentiometer in the varactor modulator block slowly

clockwise and note that in addition to the carrier frequency increasing there is a decrease in

the amount of frequency deviation that is present.

13. Return the carrier frequency potentiometer to its midway position, and monitor the audio

input (at TP6) and the FM output (at TP34) triggering the Oscilloscope on the audio input

signal. Turn the audio oscillator's amplitude potentiometer throughout its range of

adjustment.

.

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TABULATION :

Signals Amplitude Time Frequency

Message signal

Carrier signal

FM signal

Demodulated signal

MODEL GRAPH:

RESULT:

Frequency modulater and demodulater are constructed and its waveforms are analysed.

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EX.NO: 3 PULSE MODULATION-PAM/PPM/PWM

DATE:

AIM:

Study of Pulse Amplitude, Pulse Position and Pulse Width Modulation & Demodulation

Technique.

APPARATUS REQUIRED:

1. ST2110 with power supply cord

2. CRO with connecting probe

3. Connecting cords

THEORY:

PAM:

Most digital modulation systems are based on pulse modulation. It involves variation

of a pulse parameter in accordance with the instantaneous value of the information signal.

This parameter can be amplitude, width, repetitive frequency etc.

Depending upon the nature of parameter varied, various modulation systems are used.

Pulse amplitude modulation, pulse width modulation, pulse code modulation are few

modulation systems cropping up from the pulse modulation technique. In pulse amplitude

modulation (PAM) the amplitude of the pulses are varied in accordance with the modulating

signal.

In true sense, pulse amplitude modulation is analog in nature but it forms the basis of

most digital communication and modulation systems. The pulse modulation systems require

analog information to be sampled at predetermined intervals of time. Sampling is a process of

taking the instantaneous value of the analog information at a predetermined time interval.

A sampled signal consists of a train of pulses, where each pulse corresponds to the

amplitude of the signal at the corresponding sampling time. The signal sent to line is

modulated in amplitude and hence the name Pulse Amplitude Modulation (PAM).

PPM:

The Amplitude and width of the pulses is kept constant in this system, while the

position of each pulse, in relation to the position of a recurrent reference pulse is varied by

each instantaneous sampled value of the modulating wave. As mentioned in connection with

pulse width modulation, pulse-position modulations has the advantage of requiring constant

transmitter power output, but the disadvantages of depending on transmitter receiver is

synchronization.

PWM:

In pulse width modulation of pulse amplitude modulation is also often called PDM

(pulse duration modulation) and less often, PLM (pulse length modulation). In this system,

we have fixed amplitude and starting time of each pulse, but the width of each pulse is made

proportional to the amplitude of the signal at that instant.

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PROCEDURE:

FIGURE:1 PULSE AMPLITUDE MODULATION & DEMODULATION :

1. Connect the circuit as shown in Figure 1.

Keep the gain pot in AC amplifier block in anti clock wise position.

2. Switch ‘On’ the power supply & oscilloscope.

3. Observe the outputs at TP (3 & 5) these are natural & flat top outputs

respectively.

4. Observe the difference between the two outputs.

5. Vary the amplitude potentiometer and frequency change over switch & observe

the effect on the two outputs.

6. Vary the frequency of pulse, by connecting the pulse input to the 4 frequencies

available i.e. 8, 16, 32, 64 kHz in Pulse output block.

7. Switch ‘On’ fault No. 1, 2, 3, 4 one by one & observe their effect on Pulse

Amplitude Modulation output and try to locate them.

8. Monitor the output of AC amplifier. It should be a pure sine wave similar to

input.

9. Vary the amplitude of input, the amplitude of output will vary.

10. Similarly connect the sample & hold & flat top outputs to low pass filter and see

the demodulated waveform at the output of AC amplifier.

11. Switch ‘On’ the switched faults No. 1, 2, 3, 4, 5 & 8 one by one and see their

effects on output.

12. Switch ‘Off’ the power supply.

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FIGURE 2: PULSE POSITION MODULATION

1. Connect the circuit as shown in Figure 2 and also described below for clarity.

a. Input of pulse position modulation blocks to sine wave output of FG block.

2. Switch ‘On’ the power supply & oscilloscope.

3. Keep the oscilloscope at 0.5mS / div, time base speed and in X-5 mode, and

observe the pulse position modulated waveform at the pulse position modulation

block output.

4. Vary the amplitude of sine wave and observe the pulse position modulation,

keep the amplitude preset in center. Here you can best observe the pulse

modulation.

5. Switch ‘On’ fault No. 1, 2, & 6 one by one & observe their effects in pulse

position modulation output and try to locate them.

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FIGURE 3: PULSE POSITION DEMODULATION:

1. Connect the circuit as shown in Figure 3 and also described below for clarity.

a. Sine wave of 1 KHz to input of PPM block.

b. Output PPM block to input of low pass filter.

c. Output of low pass filter to input of AC amplifier.

d. Keep the gain potentiometer in amplifier block at maximum position.

2. Switch ‘On’ the power supply & oscilloscope.

3. Observe the waveform at the TP12 output of low pass filter block.

4. Then observe the demodulated output at TP14 output of AC amplifier.

5. Switch ‘On’ fault No. 1, 2, 6 & 8 one by one & observes their effect on

demodulated waveform & tries to locate them.

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FIGURE 4: PULSE WIDTH MODULATION:

1. Connect the circuit as shown in Figure 4 and also described below for clarity.

a. 1 KHz sine wave output of function generator block to modulation input

of PWM block.

b. 64 KHz square wave output to pulse input of PWM block.

2. Switch ‘On’ the power supply & oscilloscope.

3. Observe the output of PWM block.

4. Vary the amplitude of sine wave and see its effect on pulse output.

5. Vary the sine wave frequency by switching the frequency selector switch to 2

KHz.

6. Also, change the frequency of the pulse by connecting the pulse input to

different pulse frequencies viz. 8 KHz, 16 KHz, 32 KHz and see the variations

in the PWM output.

7. Switch ‘On’ fault No. 1, 2, & 5 one by one & observes their effect on PWM

output and tries to locate them.

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FIGURE 5: PULSE WIDTH DEMODULATION:

1. Connect the circuit as shown in Figure 8.1 and also described below for clarity.

a. 1 KHz sine wave output of function generator block to modulation input of

PWM block.

b. 64 KHz square wave output to pulse input.

c. Output of PWM to input of low pass filter.

d. Output of low pass filter to input of AC Amplifier.

2. Switch ‘On’ the power supply & oscilloscope.

3. Observe the output of low pass filter and AC amplifier respectively to understand the

demodulation of pulse width demodulation waveform in detail.

4. Vary the amplitude and frequency of sine wave and observe its effect on the demodulated

waveform.

5. Now, connect the pulse input in the pulse width modulation block to the different

frequencies available on board viz. 8, 16, 32 KHz and observe their demodulated waveforms.

6. Try varying the amplitude of sine wave signal; you will observe that the output signal

varies similarly.

7. Switch ‘On’ fault no, 1, 2, 5 & 8 one by one at a time. Observe their effects on final output

and try to locate them.

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TABULATION :

S.NO TYPE OF

MODULATION

AMPLITUDE TIME

1. INPUT SIGNAL s(t)

2.

PAM

(MODULATED)

PAM

(DEMODULATED)

3.

PPM

(MODULATED)

PPM

(DEMODULATED)

4.

PWM

(MODULATED)

PWM

(DEMODULATED)

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MODEL GRAPH:

RESULT:

Thus the Pulse amplitude, Pulse Position and Pulse width Modulation and

Demodulation techniques have been determined and also graphs are plotted.

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EX.NO: 4 PULSE CODE MODULATION

DATE:

AIM:

To study the pulse code modulation & demodulation technique.

APPARATUS REQUIRED:

1. Power Supply

2. IC TL5501

3. DAC0805

4. Resistors – 1k,5.6k,2.4k,5.1k(2 nos),10k,510Ώ

5. Capacitors-10μF,0.1μF(3 Nos),1nF.

6. AFO

7. CRO

THEORY:

Pulse Code Modulation technique involves following steps:

(a) Sampling: The analog signal is sampled according to the nyquist criteria. The nyquist criteria

states that for faithful reproduction of a band limited signal, the sampling rate must be at least twice

the highest frequency component present in the signal. So sampling frequency 2 fm, where fm is

maximum frequency component present in the signal

Practically the sampling frequency is kept slightly more than the required rate.

(b) Allocation of binary codes: Each binary word defines a particular narrow range of amplitude

level. The sampled value is then approximated to the nearest amplitude level. The sample is then

assigned a code corresponding to the amplitude level, which is then transmitted.

This process is called quantization and it is generally carried out by the A/D Converter

PROCEDURE:

1. Connections are made as per the circuit diagram.

2. Give the message signal as an input to the circuit by function generator.

3. Modulated signal output of pulse code can be observed through p7 of TL5501.

4. Demodulated output of the signal can be determined from the connections.

5. Draw the graph for the observed signals.

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CIRCUIT DIAGRAM:

TABULATION :

S.NO SIGNAL AMPLITUDE TIME

1. MESSEGE SIGNAL

2. PCM SIGNAL

3. DEMODULATED SIGNAL

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MODEL GRAPH:

RESULT:

Thus the Pulse Code Modulation and Demodulation technique have been studied.

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EX.NO: 5 DELTA MODULATION , ADAPTIVE DELTA

DATE: MODULATION

AIM:

To study delta modulation and adaptive delta modulation techniques

APPARATUS REQUIRED:

1. ST2105 trainer kit

2. CRO

3. patch chords

THEORY:

DELTA MODULATION:

Delta modulation is a system of digital modulation developed after pulse code

modulation. In this system, at each sampling time, say the Kth sampling time, the difference

between the sample value at sampling time K and the sample value at the previous sampling

time (K-1) is encoded into just a single bit. i.e. at each sampling time we ask simple question.

ADAPTIVE DELTA MODULATION:

Delta modulation system is unable to chase the rapidly changing information of the

analog signal, which gives rise to distortion & hence poor quality reception. This is known as

slope overloading phenomenon. The problem can be overcome by increasing the integrator

gain (i.e. step-size). But using high step-size integrator would lead to a high quantization

noise.

PROCEDURE:

1. Connect the mains supply to the Trainer

2. Make connection on the board as shown in the figure 1

3. Ensure that the clock frequency selector block switches A & B are in A = 0 and

B = 0 position.

4. Ensure that integrator 1 block's switches are in following position:

a) Gain control switch in left-hand position (towards switch A & B).

b) Switches A & B in A=0 and B=0 positions.

5. Ensure that the switches in integrator 2 blocks are in following position:

a) Gain control switch in right-hand position (towards switch A & B)

b) Switches A & B are in A = 0 and B = 0 positions.

6. Switch 'ON' the trainer.

7. To ensure the correct operation, the I/P of Comparators’ (+) terminal is connect to DC

source of OV & (-) terminal is connector to Integrator 1 O/P. O/P of Comparator is fed to the

I/P of bistable CKT Transmission clock is connected to clock generator .

8. Connect Unipolar bipolar Connector O/P to the integrator.Insure the O/P at integrator 1 &

it should be ensure that O/P of integrator is Triangular waveform & if it is not triangular then

set the level control observe bistable also the O/P of Comparator & CKT on CRO & it should

be square wave. The output from the transmitter's bistable circuit (TP14) will now be a

stream of alternate '1' and '0', ‘s' this is also the output of the delta modulator itself. The

delta modulator is now said to be 'balanced' for correct operation.

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9. Examine the signal at the output of integrator 2 (TP47) at the receiver. This should be a

triangle wave, with step size equal to that of integrator 1, and ideally centre around 0 Volts. If

there is any DC bias at the output of integrator 2, remove it by adjusting the receiver's level

adjust preset (in the bistable & level changer circuit 2 block). This preset adjusts the relative

amplitudes if the positive and negative output levels from the receiver's level changer circuit

only when these levels are balanced will there be no offset at the output of integrator2.

10. Display the data of the transmitter's bistable (at TP14), together with the analog input at

TP9 (again trigger on this signal), and note that the 250 Hz sine wave has effectively been

encoded into a stream of data bits at the bistable's output, ready for transmission to the

receiver.

11. For a full understanding of how the delta modulator is working, examine the output of the

voltage comparator (TP11), the bistable's clock input (TP13), and the level changer's bipolar

output (TP15)

12. Display the output of integrator 1 (TP17) and that of integrator 2 (TP47) on the scope.

Note that the two signals are very similar in appearance, showing that the demodulator is

working as expected.

13. Display the output of integrator 2 (TP47) together with the output of the receiver's low

pass filter block (TP51).

14. The current system clock frequency is 32 KHz. This is set by the A, B switches in the

clock frequency selector block, which are currently in the A= 0, B= 0 positions. While

monitoring the same signals, increase the system clock frequency to 64 KHz, by putting the

switches in the A = 0, B = 1 positions.

15. By changing the system clock frequency to first 128 KHz (clock frequency selector

switches in A=l, B=0 positions), and then to 256 KHz (switches in A=l,B=1 positions), note

the improvement in the low - pass filter's output signal (TP51). Once again, it may be

necessary to adjust slightly the transmitter's level adjust preset, in order to obtain a stable

oscilloscope trace.

16. Using a system clock frequency of 256 KHz (which gives a step size of approximately

60mV), compare the low pass filter's output. (TP51) with the original analog input (TP9).

There should now be no noticeable difference between them, other than a slight delay.

17. While continuing to monitor the transmitter's analog input (TP9) and the receiver's low-

pass filter output (TP51), disconnect the comparator's + input from the 250Hz sine wave

output, and connect it the 500Hz, 1 KHz and 2 KHz outputs in turn. Note that, as the

frequency of the analog signal increases, so the low pass filter's output becomes more

distorted and reduced in amplitude.

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25

1. Connect the mains supply.

2. Connect the board as per figure 2

3. Ensure that the clock frequency selector switches A & B are in A=0 & B=0

position.

4. Ensure that the switches in TX. Integrator gain control block are in following

positions.

a) Gain control switch at the L.H.S. position. (towards switches A & B)

b) Switches A & B in position A=0 & B=0.

5. Ensure that the switches in receiver's integrator gain control block are in

following positions:

a) Gain control switches at the R.H.S. position. (towards switches A & B)

b) Switches A & B in Position A=0 & B=0.

6. Turn all the potentiometers of function, generator block namely 250Hz to 2 KHz to their

fully clockwise positions.

7. Turn ON the supply.

8. As the gain control switch is towards A & B switches the gain setting is still manual,

connect the voltage comparator's +ve input to 0V & check whether the modulator &

demodulator are balanced for correct operation as in delta modulation experimentation.

Change the clock frequency selector switches to the A=1, B=1, positions (256 KHz Clock

Frequency) before continuing.

9. Disconnect the voltage comparators '+' input from 0V and reconnect it to the 2 KHz output

from the function generator block.

10. Monitor the 2 KHz analog input at TP9 and the output of integrator 1 at TP17.

11. At the transmitter, move the slider of the gain control switch in the integrator 1 block to

the right-hand position (towards the sockets labeled A, B). At the receiver, move the slider of

the gain control switch in the integrator 2 blocks to the left-hand position (again towards the

sockets labeled A, B). The gain of each integrator is now controlled by the outputs of the

counter connected to it.

12. Once again examine the 2 KHz analog input at TP9 and the output of integrator 1 at

TP17, noting that the" slope overloading problem has been eliminated, and that the

integrator's output once again follows the analog input signal. Again, it may be necessary to

adjust slightly the transmitter's level adjust preset, in order to obtain a stable trace of the

integrator's output signal.

13. Compare the output of integrator 1 (TP17) with that of integrator 2 (TP47); noting that, as

expected, both are identical in appearance.

14. Examine the output of the low pass filter (TP51) and the output of integrator 2 (TP47).

The filter has removed the high-frequency components from the integrator's output signal, to

leave goods, clean 2 KHz sine wave.

15. Compare the original 2 KHz analog input signal (at TP9) with the output signal from the

receiver's low pass filter at TP47)..

16. Disconnect the voltage comparator’s '+' input from the 2 KHz function generator output,

and reconnected it in turn to the 1 KHz, 500Hz and 250Hz outputs, noting in each case that

the demodulator’s output signal is identical to the modulator's input signal, but delayed in

time.

17. Examine also the test points in the adaptive control circuit 1 block (TP20-24), to ensure

you have a complete understanding of how the adaptive delta modulator is operating.

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27

TABULATION :

Amplitude time Frequency

Input signal

Integrator signal

Modulated signal

Demodulated signal

MODEL GRAPH:

RESULT:

Thus the delta modulation and demodulation and adaptive delta modulation and

demodulation is obtained and its corresponding graphs are drawn.

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EX.NO: 6(a) DIGITAL MODULATION AND DEMODULATION

DATE: TECHNIQUES

AIM:

To study the function of ASK,PSK,FSK and QPSK modulation and demodulation

APPARATAUS REQUIRED:

1. ST2156 and ST2157 Trainers.

2. 2 mm Banana cable

3. Oscilloscope & Probes

THEORY:

In digital modulation, an analog carrier signal is modulated by a digital bit stream.

Digital modulation methods can be considered as digital -to-analog conversion, and the

corresponding demodulation or detection as analog -to-digital conversion. To be able to transmit

the data over long distance, we have to modulate the signal that is varying phase, frequency

or amplitude according to the digital data. At the receiver separate the signal and the digital

information by the process of demodulation.

A modulating carrier with a data stream is to change the amplitude of the carrier wave

every time the data changes. This modulation technique is known as Amplitude Shift Keying.

CIRCUIT DIAGRAM:

FIGURE :1 AMPLITUDE SHIFT KEYING MODULATION & DEMODULATION TECHNIQUE

29

PROCEDURE:

1.Connect the power supplies of ST2156 and ST2157 but do not turn on the power

supplies until connections are made for this experiment.

2. Make the connections as shown in the Figure 1.

3. Switch 'ON' the power.

4. On ST2156, connect oscilloscope CH1 to ‘Clock In’ and CH2 to ‘Data In’ and

observe the waveforms.

5. On ST2156, connect oscilloscope CH1 to ‘NRZ (L)’ and CH2 to ‘Output’ of

modulator Circui t (l ) on ST2156 and observe the waveforms.

6. On ST2156, connect oscilloscope CH1 to ‘NRZ (L)’ and CH2 to ‘Output’ of

comparator on ST2157 and observe the waveforms.

MODEL WAVEFORM:

30

FREQUENCY SHIFT KEYING (FSK):

THEORY:

Frequency-shift keying (FSK) is a frequency modulation scheme in which digital

information is transmitted through discrete frequency changes of a carrier wave. The simplest

FSK is binary FSK (BFSK). BFSK uses a pair of discrete frequencies to transmit binary (0s

and 1s) information. With this scheme, the "1" is called the mark frequency and the "0" is

called the space frequency. The time domain of an FSK modulated carrier is illustrated in the

figures to the right.

CIRCUIT DIAGRAM:

FIGURE 2: FREQUENCY SHIFT KEYING MODULATION & DEMODULATION

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PROCEDURE

1.Connect the power supplies of ST2156 and ST2157 but do not turn on the power

supplies until connections are made for this experiment.

2. Make the connections as shown in the Figure 2.

3. Switch 'ON' the power.

4. On ST2156, connect oscilloscope CH1 to ‘Clock In’ and CH2 to ‘Data In’ and

observe the waveforms.

5. On ST2156, connect oscilloscope CH1 to ‘NRZ (L)’ and CH2 to ‘Output’ of

Summing Amplifier on ST2156 and observe the waveforms.

6.Adjust the amplitude of FSK waveform at Summing Amplifier’s output on

ST2156.

7. On ST2156, connect oscillscope CH1 to ‘NRZ (L)’ and CH2 to ‘Output’ of

comparator on ST2157 and observe the waveforms.

WAVEFORM:

32

PHASE SHIFT KEYING (PSK)

THEORY

PSK is a digital modulation scheme which is analogues to phase modulation. In

binary phase shift keying two output phases are possible for a single carrier frequency one

out of phase represent logic 1 and logic 0. As the input digital binary signal change state the

phase of output carrier shift two angles that are 180o out of phase. In a PSK modulator the

carrier input signal is multiplied by the digital data. The input carrier is multiplied by either a

positives or negatives consequently the output signal is either +1sinwt or 1sinwt. The first

represent a signal that is phase with the reference oscillator the latter a signal that is 180 out

of phase with the reference oscillator. Each time a change in input logic condition will change

the output phase consequently for PSK the output rate of change equal to the input rate range

and widest output bandwidth occurs when the input binary data are alternating 1/0 sequence.

The fundamental frequency of an alternate 1/0 bit sequence is equal to one half of the bit rate.

FIGURE :3 PHASE SHIFT KEYING MODULATION & DEMODULATION

PROCEDURE:

1. Connect the power supplies of ST2156 and ST2157 but do not turn on the power

supplies until connections are made for this experiment.

2. Make the connections as shown in the figure 3.

3. Switch 'ON' the power.

4. On ST215, connect oscilloscope CH1 to ‘Clock In’ and CH2 to ‘Data In’ and

observe the waveforms.

33

5. On ST2156, connect oscilloscope CH1 to ‘NRZ (L)’ and CH2 to ‘Output’ of

Modulator Circuit (l ) on ST2156 and observe the waveforms.

6. Adjust the ‘Gain’ potentiometer of the Modulator Circuit (l ) on ST2156 to

adjust the amplitude of PSK waveform at output of Modulator Circui t (l ) on

ST2156.

7. Now on ST2157 connect oscilloscope CH1 to ‘Input’ of PSK demodulator and

connect CH2 one by one to output of double squaring circuit, output of PLL,

output of Divide by four(÷ 2) observe the wave forms.

8. On ST2157 connect oscilloscope CH1 to output of Phase adjust and CH2 to

‘output’ of PSK demodulator and observe the waveforms.

9. Now connect oscilloscope CH1 to ‘PSK’ output of PSK demodulator on

ST2157 and connect CH2 ‘Output’ of Low Pass Filter on ST2157 and observe

the waveforms.

34

QUADRATURE PHASE SHIFT KEYING (QPSK)

THEORY:

QPSK is another form of angle-modulated, constant-amplitude digital modulation. It

is an M-ary encoding technique where M=4. with QPSK four output phases are possible for a

single carrier frequency. Two bits (a dibit) are clocked into the bit splitter. After both bits

have been serially inputted, they are simultaneously parallel outputted. One bit is directed to

the I channel and the other to the Q channel. The I bit modulates a carrier that is in phase with

the reference oscillator and the Q bit modulates a carrier that is 900 out of phase with the

reference carrier. QPSK modulator is two BPSK modulators combined in parallel.

The input QPSK signal is given to the I and Q product detectors and the carrier recovery

circuit. The carrier recovery circuit produces the original transmit carrier oscillator signal.

The recovered carrier must be frequency and phase coherent with the transmit reference

carrier. The QPSK signal is demodulated in the I and Q product detectors, which generate the

original I and Q data bits. The output of the product detectors are fed to the bit combining

circuit, where they are converted from parallel I and Q data channels to a single binary output

data stream.

35

FIGURE :4 QPSK MODULATION & DEMODULATION

PROCEDURE:

1.Connect the power supplies of ST2156 and ST2157 but do not turn on the power

supplies until connections are made for this experiment.

2. Make the connections as shown in the Figure 4.

3. Switch 'ON' the power.

4. On ST2156, connect oscilloscope CH1 to ‘Clock In’ and CH2 to ‘Data In’ and

observe the waveforms.

5. On ST2156, connect oscilloscope CH1 to ‘Clock Output’ and CH2 one by one

to ‘Sine’ and ‘Cosine’ output of 960 KHz and observe the waveforms.

6. On ST2156, connect oscilloscope CH1 to ‘Data In’ and connect CH2 one by one

to ‘I Data’ and ‘Q Data’ outputs and observe the waveforms.

7. Now connect oscilloscope CH1 to ‘I Data’ output on ST2156 and connect CH2

one by one to ‘Signal In’, ‘Carrier In’ and ‘Output’ of modulator circuit (l ) on

ST2156 and observe the waveforms.

8. Now connect oscilloscope CH1 to ‘Q Data’ output on ST2156 and connect CH2

one by one to ‘Signal In’, ‘Carrier In’ and ‘Output’ of modulator circuit (ll) on

ST2156 and observe the waveforms

9. Now connect oscilloscope CH1 to ‘Data Out’ on ST2156 and CH2 to ‘Output’

of Summing Amplifier on ST2156 and observe the waveforms.

10. Set ‘Carrier frequency’ selection switch to ‘960 KHz’ on ST2157.

36

11. Now on ST2157 connect oscilloscope CH1 to ‘Input’ of QPSK demodulator

and connect CH2 one by one to output of double squaring circuit, output of

PLL, output of Divide by four(÷ 4) observe the wave forms.

12. On ST2157, connect oscilloscope CH1 to ‘I’ output of QPSK demodulator and

CH2 to ‘Q’ output of QPSK demodulator and observe the waveforms. Set all

toggle switch to 0, now vary the phase adjust potentiometer and observe its

effects on the demodulated signal waveforms.

13. Connect oscilloscope CH1 to ‘I’ output of QPSK demodulator on ST2157 then

connect CH2 one by one to output of low pass filter, output of Comparator on

ST2157 and observe the waveforms.

37

Tabulation:

MODULATION

TECHNIQUES

CLOCK SIGNAL DATA SIGNAL MODULATED

OUTPUT

DEMODULATED

OUTPUT

Amplitude

(V)

Time

(sec)

Amplitude

(V)

Time

(sec)

Amplitude

(V)

Time

(sec)

Amplitude

(V)

Time

(sec)

ASK

FSK

PSK

QPSK

38

RESULT:

Thus the ASK, FSK, PSK and QPSK modulation and demodulation process is

obtained and its corresponding output is plotted.

39

EX.NO: 6(B) DIGITAL MODULATION AND DEMODULATION TECHNIQUES USING

DATE : MATLAB

AIM:

To study the function of ASK,PSK,FSK and QPSK modulation and demodulation

using MATLAB.

SOFTWARE REQUIRED:

1. MATLAB 7

PROCEDURE:

ASK USING MATLAB

ALGORITHM :

Initialization commands

ASK modulation

1. Generate carrier signal.

2. Start FOR loop

3. Generate binary data, message signal(on-off form)

4. Generate ASK modulated signal.

5. Plot message signal and ASK modulated signal.

6. End FOR loop.

7. Plot the binary data and carrier.

ASK demodulation

1. Start FOR loop

2. Perform correlation of ASK signal with carrier to get decision variable

3. Make decision to get demodulated binary data. If x>0, choose ‘1’ else choose ‘0’

4. Plot the demodulated binary data.

PROGRAM

%ASK Modulation

clc;

clear all;

close all;

%GENERATE CARRIER SIGNAL

Tb=1; fc=10;

t=0:Tb/100:1;

40

c=sqrt(2/Tb)*sin(2*pi*fc*t);

%generate message signal

N=8;

m=rand(1,N);

t1=0;t2=Tb

for i=1:N

t=[t1:.01:t2]

if m(i)>0.5

m(i)=1;

m_s=ones(1,length(t));

else

m(i)=0;

m_s=zeros(1,length(t));

end

message(i,:)=m_s;

%product of carrier and message

ask_sig(i,:)=c.*m_s;

t1=t1+(Tb+.01);

t2=t2+(Tb+.01);

%plot the message and ASK signal

subplot(5,1,2);axis([0 N -2 2]);plot(t,message(i,:),'r');

title('message signal');xlabel('t--->');ylabel('m(t)');grid on

hold on

subplot(5,1,4);plot(t,ask_sig(i,:));

title('ASK signal');xlabel('t--->');ylabel('s(t)');grid on

hold on

end

hold off

%Plot the carrier signal and input binary data

subplot(5,1,3);plot(t,c);

title('carrier signal');xlabel('t--->');ylabel('c(t)');grid on

subplot(5,1,1);stem(m);

title('binary data bits');xlabel('n--->');ylabel('b(n)');grid on

41

% ASK Demodulation

t1=0;t2=Tb

for i=1:N

t=[t1:Tb/100:t2]

%correlator

x=sum(c.*ask_sig(i,:));

%decision device

if x>0

demod(i)=1;

else

demod(i)=0;

end

t1=t1+(Tb+.01);

t2=t2+(Tb+.01);

end

%plot demodulated binary data bits

subplot(5,1,5);stem(demod);

title('ASK demodulated signal'); xlabel('n--->');ylabel('b(n)');grid on

PSK USING MATLAB

ALGORITHM

Initialization commands

PSK modulation

1. Generate carrier signal.

2. Start FOR loop

3. Generate binary data, message signal in polar form

4. Generate PSK modulated signal.

5. Plot message signal and PSK modulated signal.

6. End FOR loop.

7. Plot the binary data and carrier.

PSK demodulation

1. Start FOR loop

Perform correlation of PSK signal with carrier to get decision variable

42

2. Make decision to get demodulated binary data. If x>0, choose ‘1’ else choose ‘0’

3. Plot the demodulated binary data.

PROGRAM

% PSK modulation

clc;

clear all;

close all;

%GENERATE CARRIER SIGNAL

Tb=1;

t=0:Tb/100:Tb;

fc=2;

c=sqrt(2/Tb)*sin(2*pi*fc*t);

%generate message signal

N=8;

m=rand(1,N);

t1=0;t2=Tb

for i=1:N

t=[t1:.01:t2]

if m(i)>0.5

m(i)=1;

m_s=ones(1,length(t));

else

m(i)=0;

m_s=-1*ones(1,length(t));

end

message(i,:)=m_s;

%product of carrier and message signal

bpsk_sig(i,:)=c.*m_s;

%Plot the message and BPSK modulated signal

subplot(5,1,2);axis([0 N -2 2]);plot(t,message(i,:),'r');

title('message signal(POLAR form)');xlabel('t--->');ylabel('m(t)');

grid on; hold on;

subplot(5,1,4);plot(t,bpsk_sig(i,:));

title('BPSK signal');xlabel('t--->');ylabel('s(t)');

43

grid on; hold on;

t1=t1+1.01; t2=t2+1.01;

end

hold off

%plot the input binary data and carrier signal

subplot(5,1,1);stem(m);

title('binary data bits');xlabel('n--->');ylabel('b(n)');

grid on;

subplot(5,1,3);plot(t,c);

title('carrier signal');xlabel('t--->');ylabel('c(t)');

grid on;

% PSK Demodulation

t1=0;t2=Tb

for i=1:N

t=[t1:.01:t2]

%correlator

x=sum(c.*bpsk_sig(i,:));

%decision device

if x>0

demod(i)=1;

else

demod(i)=0;

end

t1=t1+1.01;

t2=t2+1.01;

end

%plot the demodulated data bits

subplot(5,1,5);stem(demod);

title('demodulated data');xlabel('n--->');ylabel('b(n)');

grid on

FSK USING MATLAB

ALGORITHM

Initialization commands

FSK modulation

44

1. Generate two carriers signal.

2. Start FOR loop

3. Generate binary data, message signal and inverted message signal

4. Multiply carrier 1 with message signal and carrier 2 with inverted message signal

5. Perform addition to get the FSK modulated signal

6. Plot message signal and FSK modulated signal.

7. End FOR loop.

8. Plot the binary data and carriers.

FSK demodulation

1. Start FOR loop

2. Perform correlation of FSK modulated signal with carrier 1 and carrier 2 to get two

decision

variables x1 and x2.

3. Make decisionon x = x1-x2 to get demodulated binary data. If x>0, choose ‘1’ else choose

‘0’.

4. Plot the demodulated binary data.

PROGRAM

% FSK Modulation

clc;

clear all;

close all;

%GENERATE CARRIER SIGNAL

Tb=1; fc1=2;fc2=5;

t=0:(Tb/100):Tb;

c1=sqrt(2/Tb)*sin(2*pi*fc1*t);

c2=sqrt(2/Tb)*sin(2*pi*fc2*t);

%generate message signal

N=8;

m=rand(1,N);

t1=0;t2=Tb

for i=1:N

t=[t1:(Tb/100):t2]

if m(i)>0.5

m(i)=1;

45

m_s=ones(1,length(t));

invm_s=zeros(1,length(t));

else

m(i)=0;

m_s=zeros(1,length(t));

invm_s=ones(1,length(t));

end

message(i,:)=m_s;

%Multiplier

fsk_sig1(i,:)=c1.*m_s;

fsk_sig2(i,:)=c2.*invm_s;

fsk=fsk_sig1+fsk_sig2;

%plotting the message signal and the modulated signal

subplot(3,2,2);axis([0 N -2 2]);plot(t,message(i,:),'r');

title('message signal');xlabel('t---->');ylabel('m(t)');grid on;hold on;

subplot(3,2,5);plot(t,fsk(i,:));

title('FSK signal');xlabel('t---->');ylabel('s(t)');grid on;hold on;

t1=t1+(Tb+.01); t2=t2+(Tb+.01);

end

hold off

%Plotting binary data bits and carrier signal

subplot(3,2,1);stem(m);

title('binary data');xlabel('n---->'); ylabel('b(n)');grid on;

subplot(3,2,3);plot(t,c1);

title('carrier signal-1');xlabel('t---->');ylabel('c1(t)');grid on;

subplot(3,2,4);plot(t,c2);

title('carrier signal-2');xlabel('t---->');ylabel('c2(t)');grid on;

% FSK Demodulation

t1=0;t2=Tb

for i=1:N

t=[t1:(Tb/100):t2]

%correlator

x1=sum(c1.*fsk_sig1(i,:));

x2=sum(c2.*fsk_sig2(i,:));

46

x=x1-x2;

%decision device

if x>0

demod(i)=1;

else

demod(i)=0;

end

t1=t1+(Tb+.01);

t2=t2+(Tb+.01);

end

%Plotting the demodulated data bits

subplot(3,2,6);stem(demod);

title(' demodulated data');xlabel('n---->');ylabel('b(n)'); grid on;

QPSK USING MATLAB

ALGORITHM

Initialization commands

QPSK modulation

1. Generate quadrature carriers.

2. Start FOR loop

3. Generate binary data, message signal(bipolar form)

4. Multiply carrier 1 with odd bits of message signal and carrier 2 with even bits of message

signal

5. Perform addition of odd and even modulated signals to get the QPSK modulated signal

6. Plot QPSK modulated signal.

7. End FOR loop.

8. Plot the binary data and carriers.

QPSK demodulation

1. Start FOR loop

2. Perform correlation of QPSK modulated signal with quadrature carriers to get two decision

variables x1 and x2.

3. Make decision on x1 and x2 and multiplex to get demodulated binary data.

If x1>0and x2>0, choose ‘11’. If x1>0and x2<0, choose ‘10’. If x1<0and x2>0, choose ‘01.

If

47

x1<0and x2<0, choose ‘00’.

4. End FOR loop

5. Plot demodulated data

PROGRAM

% QPSK Modulation

clc;

clear all;

close all;

%GENERATE QUADRATURE CARRIER SIGNAL

Tb=1;t=0:(Tb/100):Tb;fc=1;

c1=sqrt(2/Tb)*cos(2*pi*fc*t);

c2=sqrt(2/Tb)*sin(2*pi*fc*t);

%generate message signal

N=8;m=rand(1,N);

t1=0;t2=Tb

for i=1:2:(N-1)

t=[t1:(Tb/100):t2]

if m(i)>0.5

m(i)=1;

m_s=ones(1,length(t));

else

m(i)=0;

m_s=-1*ones(1,length(t));

end

%odd bits modulated signal

odd_sig(i,:)=c1.*m_s;

if m(i+1)>0.5

m(i+1)=1;

m_s=ones(1,length(t));

else

m(i+1)=0;

m_s=-1*ones(1,length(t));

end

%even bits modulated signal

48

even_sig(i,:)=c2.*m_s;

%qpsk signal

qpsk=odd_sig+even_sig;

%Plot the QPSK modulated signal

subplot(3,2,4);plot(t,qpsk(i,:));

title('QPSK signal');xlabel('t---->');ylabel('s(t)');grid on; hold on;

t1=t1+(Tb+.01); t2=t2+(Tb+.01);

end

hold off

%Plot the binary data bits and carrier signal

subplot(3,2,1);stem(m);

title('binary data bits');xlabel('n---->');ylabel('b(n)');grid on;

subplot(3,2,2);plot(t,c1);

title('carrier signal-1');xlabel('t---->');ylabel('c1(t)');grid on;

subplot(3,2,3);plot(t,c2);

title('carrier signal-2');xlabel('t---->');ylabel('c2(t)');grid on;

% QPSK Demodulation

t1=0;t2=Tb

for i=1:N-1

t=[t1:(Tb/100):t2]

%correlator

x1=sum(c1.*qpsk(i,:));

x2=sum(c2.*qpsk(i,:));

%decision device

if (x1>0&&x2>0)

demod(i)=1;

demod(i+1)=1;

elseif (x1>0&&x2<0)

demod(i)=1;

demod(i+1)=0;

elseif (x1<0&&x2<0)

demod(i)=0;

demod(i+1)=0;

elseif (x1<0&&x2>0)

49

demod(i)=0;

demod(i+1)=1;

end

t1=t1+(Tb+.01); t2=t2+(Tb+.01);

end

subplot(3,2,5);stem(demod);

title('qpsk demodulated bits');xlabel('n---->');ylabel('b(n)');grid on;

MODEL GRAPHS:

ASK

50

PSK:

FSK:

51

QPSK:

RESULT:

The program for ASK, FSK, PSK and QPSK modulation and demodulation has been

simulated in MATLAB and necessary graphs are plotted.

52

EX.NO: 7 DESIGNING, ASSEMBLING AND TESTING OF PRE-EMPHASIS/

DATE : DE-EMPHASIS CIRCUITS

AIM

i) To observe the effects of pre-emphasis on given input signal.

ii) To observe the effects of De-emphasis on given input signal.

APPARATUS REQUIRED

NAME OF THE

COMPONENT/EQUIPMENT

SPECIFICATIONS/RANGE QUANTITY

Transistor (BC 107) f T = 300 MHz

P = 1W

Ic(max) = 100 mA

1

Resistors 10 KΩ, 7.5 KΩ, 6.8 KΩ

1 each

Capacitors

10 nF

0.1 µF

1

2

CRO

20MHZ 1

Function Generator 1MHZ 1

1

Regulated Power Supply 0-30V, 1A 1

THEORY

In telecommunication, a pre-emphasis circuit is inserted in a system in order to

increase the magnitude of one range of frequencies with respect to another. Pre-emphasis is

usually employed in FM or phase modulation transmitters to equalize the modulating signal

drive power in terms of deviation ratio. In high speed digital transmission, pre-emphasis is

used to improve signal quality at the output of a data transmission. In transmitting signals at

high data rates, the transmission medium may introduce distortions, so pre-emphasis is used

to distort the transmitted signal to correct for this distortion. When done properly this

produces a received signal which more closely resembles the original or desired signal,

allowing the use of higher frequencies or producing fewer bit errors. In telecommunication,

de-emphasis is the complement of pre-emphasis. It is designed to decrease, (within a band of

53

frequencies), the magnitude of some (usually higher) frequencies with respect to the

magnitude of other (usually lower) frequencies in order to improve the overall signal-to-noise

ratio by minimizing the adverse effects of such phenomena as attenuation differences.

PROCEDURE

1. Connect the circuit as per circuit diagram as shown in Fig.1.

2. Apply the sinusoidal signal of amplitude 20mV as input signal to pre emphasis circuit.

3. Then by increasing the input signal frequency from 500Hz to 20KHz, observe the output

voltage

(vo) and calculate gain (20 log (vo/v).

4. Plot the graph between gain Vs frequency.

5. Repeat above steps 2 to 4 for de-emphasis circuit (shown in Fig.2). by applying the

sinusoidal

signal of 5V as input signal

CIRCUIT DIAGRAM

Pre-emphasis circuit:

De-emphasis circuit

54

TABLUATION

Pre-emphasis Vi=20mV

Frequency(KHz) Vo(mV) Gain in dB(20 log Vo/Vi)

De-emphasis Vi=5mV

Frequency(KHz) Vo(mV) Gain in dB(20 log Vo/Vi)

MODEL GRAPH:

55

RESULT:

The characteristics of Pre –emphasis and De-emphasis circuits were studied

56

EX.NO: 8 LINE CODING TECHNIQUES

DATE :

AIM:

To study the different line coding techniques.

APPARATUS REQUIRED:

1. ST2156 &ST2157 trainer kit.

2. CRO

3. 2mm banana cable.

THEORY:

Line coding consists of representing the digital signal to be transported, by an

amplitude- and time-discrete signal that is optimally tuned for the specific properties of the

physical channel (and of the receiving equipment). The waveform pattern of voltage or

current used to represent the 1s and 0s of a digital signal on a transmission link is called line

encoding. The common types of line encoding are unipolar, polar, bipolar and Manchester

encoding. Line codes are used commonly in computer communication networks over short

distances. Each of the various line formats has a particular advantage and disadvantage. It is

not possible to select one, which will meet all needs. The format may be selected to meet

one or more of the following criteria:

Minimize transmission hardware

Facilitate synchronization

Ease error detection and correction

Minimize spectral content

Eliminate a dc component

The Manchester code is quite popular. It is known as a self-clocking code because there is

always a transition during the bit interval. Consequently, long strings of zeros or ones do not

cause clocking problems.

PROCEDURE:

Non return to zero- level (NRZ-L):

Representation : +5V for data bit 1 and 0V for data bit 0.

Bandwidth : Low bandwidth.

DC Level : High DC component.

Timing Information : No timing information (For long stream of 1s

and 0s)

Waveforms of NRZ-L

57

Non return to zero- level (NRZ-M):

Representation : Level transition for bit 1 and unchanged level for bit 0.

Bandwidth : Low bandwidth.

DC Level : High DC component.

Timing Information : No timing information (For long stream of 0s)

Waveforms of NRZ-M

Return to zero (RZ):

Representation :0V for bit 0 and for bit 1, for half bit duration +5V and the rest of the bit

duration is represented as 0V.

Bandwidth : Twice as that required for the NRZ.

DC Level : High DC component.

Timing Information : No timing information (For long stream of 0s)

Waveforms of RZ-L

Biphase (Manchester):

Representation : For bit 1, +5V for first half bit time and 0V during the second half and for

bit 0, 0V for first half bit time and +5V during the second half.

Bandwidth : Twice as that required for the NRZ.

DC Level : No DC component.

Timing Information : Good clock recovery.

Waveforms of Manchester

Biphase (Mark):

58

Representation : For any bit either 1 or 0, first half bit duration +5V or 0V and invert of first

half during next half bit duration. Bit 0 Bit Pattern remains the same. Bit 1 Phase Reversal.

Bandwidth : Twice as that required for the NRZ.

DC Level : No DC component.

Timing Information : Good clock recovery.

Waveforms of Mark

Return to Bias (RB):

Representation : During the first half a period, positive level for bit 1 and a negative level

for bit 0 and during the second half bit time, both returns to the bias level.

Bandwidth : Twice as that required for the NRZ.

DC Level : The DC component depends on the string of 1’s and 0’s.

Timing Information : Good clock recovery (Self clocking system).

Waveforms of RB

Alternate Mark Inversion (AMI):

Representation : Like RB encoding, the AMI always returns to the bias level during second

half of the bit time interval and during the first half the transmitted level can be a positive, a

negative or bias level, as for a bit 0 bias level and for a bit 1 either a positive level or negative

level, the level being chose opposite to what it was used to represent the previous bit 1.

Bandwidth : Twice as that required for the NRZ.

DC Level : No DC component.

Timing Information : No timing information (For long sequence of 0’s).

59

Waveforms of AMI

TABULATION:

S.NO SIGNALS AMPLITUDE TIME PERIOD

1. CLOCK SIGNAL

2. NRZ(L)

3. NRZ(M)

4. RZ

5. BIPHASE(MANCHESTER)

6. BIPHASE (MARK)

7. RB

8. AMI

60

ODEL GRAPH:

RESULT:

Thus the different coding techniques were studied and observed for a given binary

data, and their corresponding waveforms plotted.

61

EX.NO: 9 PLL AND FREQUENCY SYNTHESIZER

DATE :

AIM:

To study phase lock loop and its capture range, lock range and free running VCO

APPARATUS REQUIRED:

1. Power Supply

2. LM565

3. Resistors -10K,680Ώ (2 nos)

4. Capacitors -1μF,0.1μF,0.01μF

5. CRO

6. AFO

THEORY:

A phase-locked loop or phase lock loop (PLL) is a control system that generates an

output signal whose phase is related to the phase of an input "reference" signal. It is an

electronic circuit consisting of a variable frequency oscillator and a phase detector. This

circuit compares the phase of the input signal with the phase of the signal derived from its

output oscillator and adjusts the frequency of its oscillator to keep the phases matched. The

signal from the phase detector is used to control the oscillator in a feedback loop.

Frequency is the time derivative of phase. Keeping the input and output phase in lock

step implies keeping the input and output frequencies in lock step. Consequently, a phase-

locked loop can track an input frequency, or it can generate a frequency that is a multiple of

the input frequency. The former property is used for demodulation, and the latter property is

used for indirect frequency synthesis.

Phase-locked loops are widely employed in radio, telecommunications, computers

and other electronic applications. They can be used to recover a signal from a noisy

communication channel, generate stable frequencies at a multiple of an input frequency

(frequency synthesis), or distribute clock timing pulses in digital logic designs such as

microprocessors. Since a single integrated circuit can provide a complete phase-locked-loop

building block, the technique is widely used in modern electronic devices, with output

frequencies from a fraction of a hertz up to many gigahertz

The LM565 and LM565C are general purpose phase locked loops containing a stable,

highly linear voltage controlled oscillator for low distortion FM demodulation, and a double

balanced phase detector with good carrier suppression. The VCO frequency is set with an

external resistor and capacitor, and a tuning range of 10:1 can be obtained with the same

capacitor. The characteristics of the closed loop system—bandwidth, response speed, capture

and pull in range—may be adjusted over a wide range with an external resistor and capacitor.

The loop may be broken between the VCO and the phase detector for insertion of a digital

frequency divider to obtain frequency multiplication.

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FEATURES:

1. 200 ppm /°C Frequency Stability of the VCO

2. Power Supply Range of ±5 to ±12 Volts with 100 ppm/% Typical

3. 0.2% Linearity of Demodulated Output

4. Linear Triangle Wave with in Phase Zero Crossings Available

5. TTL and DTL Compatible Phase Detector Input and Square Wave Output

6. Adjustable Hold in Range from ±1% to > ±60%

APPLICATIONS:

1. Data and Tape synchronization

2. Modems

3. FSK Demodulation

4. FM Demodulation

5. Frequency Synthesizer

6. Tone Decoding

7. Frequency Multiplication and Division

8. SCA Demodulators

9. Telemetry Receivers

10. Signal Regeneration

11. Coherent Demodulators

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PROCEDURE:

1. Connect + 5V to pin 10 of LM 565.

2. Connect -5V to pin 1.

3. Connect 10k resistor from pin 8 to + 5V

4. Connect 0.01µf capacitor from pin 9 to – 5V

5. Short pin 4 to pin 5.

6. Without giving input measure(f O) free running frequency.

7. Connect pin 2 to oscillator or function generator through a 1µf capacitor, adjust the

amplitude aroung 2Vpp.

8. Connect 0.1 µf capacitor between pin 7 and + 5V (C2)

9. Connect output to the second channel is of CRO.

10. Connect output to the second channel of the CRO.

11. By varying the frequency in different steps observe that of one frequency the wave

form will be phase locked.

12. Change R-C components to shift VCO center frequency and see how lock range of the

input

TABULATION:

S.NO SIGNALS AMPLITUDE TIME

1. INPUT SIGNAL

2. DEMODULATED OUTPUT

3. VCO

RESULT:

Thus the Phase Locked Loop have been determined.

64

EX.NO: 10 ERROR CONTROL CODING USING MATLAB

DATE :

AIM :

To implement the error control linear and cyclic block codes and using MATLAB

program.

APPARATUS REQUIRED :

PC with MATLAB software.

THEORY :

It’s a sub class of linear block codes. Advantage of cyclic codes is that they are easy to

encode. A binary code is to be cyclic code it exhibits two fundamental properties

Linearity property

Cyclic property

In coding theory, a linear code is an error-correcting code for which any linear combination

of code words is also a codeword. Linear codes are traditionally partitioned into block codes

and convolutional codes, although Turbo codes can be seen as a hybrid of these two types.

Linear codes allow for more efficient encoding and decoding algorithms than other codes.

Linear codes are used in forward error correction and are applied in methods for transmitting

symbols (e.g., bits) on a communications channel so that, if errors occur in the

communication, some errors can be corrected or detected by the recipient of a message block.

The code words in a linear block code are blocks of symbols which are encoded using more

symbols than the original value to be sent. A linear code of length n transmits blocks

containing n symbols. For example, the [7,4,3] Hamming code is a linear binary code which

represents 4-bit messages using 7-bit code words. Two distinct code words differ in at least

three bits. As a consequence, up to two errors per codeword can be detected and a single error

can be corrected. This code contains 24=16 code words.

PROCEDURE :

1. Use the communication block set.

2. Perform the coding technique for the message that is generated randomly.

3. Similarly generate a noisy code signal randomly.

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4. Perform the decoding operation.

5. Analyze the result and bit error rate is calculated.

PROGRAM :

LINEAR BLOCK CODES

clc;

clear all;

close all;

%Input Generator Matrix

g=input('Enter The Generator Matrix');

disp('The order of Linear block code for given generator matrix is:');

[n,k]=size(transpose(g))

fori=1:2^k

for j=k:-1:1

if rem(i-1,2^(-j+k+1))>=2^(-j+k)

u(i,j)=1;

else

u(i,j)=0;

end

end

end

u

disp('The possible codewords are:')

c=rem(u*g,2)

disp('The minimum hamming distance dmin for given block code is=')

d_min=min(sum((c(2:2^k,:))'))

disp('The error correction capability is= ')

ec = (d_min-1)/2

%Code Word

r=input('Enter the received code word:')

p=[g(:,n-k+2:n)];

h=[transpose(p),eye(n-k)];

disp('Hamming code')

ht=transpose(h)

disp('syndrome decoding table');

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sundromematrix = ht

errorpattern = eye(n)

disp('Syndrome of a given codeword is:')

s=rem(r*ht,2)

fori=1:1:size(ht)

if(ht(i,1:3)==s)

r(i)=1-r(i);

break;

end

end

disp('The error is in bit:')

i

disp('The corrected codeword is:')

r

disp(' actual message bit is:')

m=[r(1:k)]

CYCLIC BLOCK CODES

clc

close all;

clear all;

n=6;

k=4;

data=randint(5,k,[0 1]);

disp('data');

disp(data);

code=encode(data,n,k,'%cyclic,binary');

disp('code');

disp(code);

e=randerr(5,n,[0 1;.5 .5]);

disp('randerr');

disp(e);

noise=rem([code+e],2);

disp('noise');

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disp(noise);

newdata=decode(noise,n,k,'%cyclic,binary');

disp('newdata');

disp(newdata);

[numerr,ratio]=biterr(newdata,data);

disp('The bit error ratio is');

disp(ratio);

RESULT :

Thus the program for error control cyclic and linear block code is implemented and

the outputs are verified.

68

EX.NO: 11 SAMPLING AND TIME DIVISION MULTIPLEXING

DATE :

AIM:

To Study of sampling and reconstruction of signal and Time Division Multiplexing.

APPARATUS REQUIRED:

1. ST 2101 & ST2153 Trainer kit

2. CRO

3. Patch Chords

THEORY:

SAMPLING:

The signals we use in the real world, such as our voice, are called "analog" signals.

To process these signals for digital communication, we need to convert analog signals to

"digital" form. While an analog signal is continuous in both time and amplitude, a digital

signal is discrete in both time and amplitude. To convert continuous time signal to discrete

time signal, a process is used called as sampling. The value of the signal is measured at

certain intervals in time. Each measurement is referred to as a sample.

TIME DIVISION MULTIPLEXING:

Time division multiplexing is a technique of transmitting more than one information

on the same channel. This means that several information signals can be transmitted over a

single channel by sending samples from different information sources at different moments in

time. This technique is known as time division multiplexing or TDM. TDM is widely used in

digital communication systems to increase the efficiency of the transmitting medium.TDM

can be achieved by electronically switching the samples such that they inter leave

sequentially at correct instant in time without mutual interference.

PROCEDURE:

Sampling & Reconstruction:

Initial set up of trainer:

Duty cycle selector switch position : Position 5

Sampling selector switch : Internal position

1. Connect the power cord to the trainer. Keep the power switch in ‘Off’ position.

2. Connect 1 KHz Sine wave to signal Input.

3. Switch ‘On’ the trainer's power supply & Oscilloscope.

4. Connect BNC connector to the CRO and to the trainer’s output port.

5. Select 320 KHz (Sampling frequency is 1/10th of the frequency indicated by the

illuminated LED) sampling rate with the help of sampling frequency selector switch.

6. Observe 1 KHz sine wave (TP12) and Sample Output (TP37) on Oscilloscope. The display

shows 1 KHz Sine wave being sampled at 32 KHz, so there are 32 samples for every cycle of

the sine wave. (figure 1)

7. Connect the Sample output to Input of Fourth Order low pass Filter & observe

reconstructed output on (TP46) with help of oscilloscope. The display shows the

reconstructed original 1 KHz sine wave. (figure 2)

8. By successive presses of sampling Frequency Selector switch, change the sampling

frequency to 2KHz, 4KHz, 8KHz, 16KHz and back to 32KHz (Sampling frequency is 1/10th

69

of the frequency indicated by the illuminated LED). Observe how SAMPLE output changes

in each cases and how the lower sampling frequencies introduce distortion into the filter’s

output waveform. This is due to the fact that the filter does not attenuate the unwanted

frequency component significantly. Use of higher order filter would improve the output

waveform.

9. So far, we have used sampling frequencies greater than twice the maximum input

frequency.

FIGURE 1: Signal Sampling

70

FIGURE 2: Signal Reconstruction

Time Division Multiplexing:

1. Set up the following initial conditions on ST2153:

a) Mode Switch in 320 KHz (FAST mode) position

b) DC signal (I) & DC signal (II) Controls in function generator block fully clockwise.

c) ~ 2 KHz and ~4 KHz control levels set to give 10Vpp.

d) Pseudo - random sync code generator on/off switch in OFF Position.

e) Error check code generator switch A & B in A=0 & B=0 position (OFF Mode)

f) All switched faults off.

2. First, connect only the 2 KHz output to CH.I

71

3. Turn ON the power. Check that the PAM output of 2 KHz sine wave is available at TP17

of the ST2153.

4. Connect channel 1 of the oscilloscope to TP15 & channel 2 of the oscilloscope to TP17.

Observe the timing & phase relation between the sampling signal TP15 & the sampled

waveform at TP17.

5. Turn OFF the power supply. Now connect also the 4 KHz supply to CH.II.

6. Connect channel 1 of the oscilloscope to TP16 & channel 2 of the oscilloscope to TP17.

7. Observe & explain the timing relation between the signals at TP15, 7, 9, 16 & 17.

MODEL GRAPH:

Waveform at TP15

Waveform at TP16

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Waveform at TP 18when only one input signal is present

Waveform at TP17 when only one input is connected

73

Waveform at TP17 when only one input is connected

TABULATION: (SAMPLING)

S.NO SIGNAL AMPLITUDE TIME

1. INPUT SIGNAL

2. SAMPLED SIGNAL

3. RECONSTRUCTED SIGNAL

TABULATION: (TIME DIVISION MULTIPLEXING)

S.NO SIGNAL AMPLITUDE TIME

1. AT TP 15

2. AT TP16

3. AT TP 17

4. AT TP18

74

RESULT:

Thus the signal have been sampled and reconstructed and also Time Division

Multiplexing was studied and the graphs are plotted.

75

EX.NO: 12 FREQUENCY DIVISION MULTIPLEXING

DATE :

AIM:

To determine the Frequency Division Multiplexing using MATLAB

APPARATUS REQUIRED:

1. MATLAB Software/Simulink

THEORY:

Frequency-division multiplexing (FDM) is a technique by which the total bandwidth

available in a communication medium is divided into a series of non-overlapping frequency

sub-bands, each of which is used to carry a separate signal. This allows a single transmission

medium such as the radio spectrum, a cable or optical fiber to be shared by many signals. The

most natural example of frequency-division multiplexing is radio and television broadcasting,

in which multiple radio signals at different frequencies pass through the air at the same time.

At the source end, for each frequency channel, an electronic oscillator generates a

carrier signal, a steady oscillating waveform at a single frequency such as a sine wave, that

serves to "carry" information. The carrier is much higher in frequency than the data signal.

The carrier signal and the incoming data signal (called the baseband signal) are applied to a

modulator circuit. The modulator alters some aspect of the carrier signal, such as its

amplitude, frequency, or phase, with the data signal, "piggybacking" the data on the carrier.

Multiple modulated carriers at different frequencies are sent through the transmission

medium, such as a cable or optical fiber. Each modulated carrier consists of a narrow band of

frequencies, centered on the carrier frequency. The information from the data signal is carried

in sidebands on either side of the carrier frequency. This band of frequencies is called the

passband for the channel. As long as the carrier frequencies of separate channels are spaced

far enough apart so that their passbands do not overlap, the separate signals will not interfere

with one another. Thus the available bandwidth is divided into "slots" or channels, each of

which can carry a data signal.

At the destination end of the cable or fiber, for each channel, an electronic filter

extracts the channel's signal from all the other channels. A local oscillator generates a signal

at the channel's carrier frequency. The incoming signal and the local oscillator signal are

applied to a demodulator circuit. This translates the data signal in the sidebands back to its

original baseband frequency. An electronic filter removes the carrier frequency, and the data

signal is output for use. Modern FDM systems often use sophisticated modulation methods

that allow several data signals to be transmitted through each frequency channel.

PROCEDURE:

1. Get the blocks from the MATLAB\Simulink tool

2. Determine the sine wave and the bandpass filters .

3. Obtain the output of FDM from the spectrum scope.

76

BLOCK DIAGRAM;

RESULT:

Thus the Frequency Division Multiplexing was determined using MATLAB