Ec2307 Cs Lab Manual

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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.

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:
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 circuits 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.
.
9

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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.
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:
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.
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:
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.
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.
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,510
5. Capacitors-10F,0.1F(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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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 comparators '+' 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 demodulators 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.

26

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.
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

31

PROCEDURE
Summing Amplifier on ST2156 and observe the waveforms.
6.Adjust the amplitude of FSK waveform at Summing Amplifiers 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 180
o
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
2. Make the connections as shown in the figure 3.
33

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 90
0
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:
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
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
PROGRAM
% PSK modulation
clc;
clear all;
close all;
Tb=1;
t=0:Tb/100:Tb;
fc=2;
c=sqrt(2/Tb)*sin(2*pi*fc*t);
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;
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
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
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
title('demodulated data');xlabel('n--->');ylabel('b(n)');
grid on
FSK USING MATLAB
ALGORITHM
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.
PROGRAM
% FSK Modulation
clc;
clear all;
close all;
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);
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

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

QPSK USING MATLAB
ALGORITHM
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);
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;
else
m(i)=0;
end
%odd bits modulated signal
odd_sig(i,:)=c1.*m_s;
if m(i+1)>0.5
m(i+1)=1;
else
m(i+1)=0;
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
title('binary data bits');xlabel('n---->');ylabel('b(n)');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
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 K, 7.5 K, 6.8 K

1 each
Capacitors

10 nF
0.1 F

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

De-emphasis V
i
=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.
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.
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.
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.
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.
DC Level : The DC component depends on the string of 1s and 0s.
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.
Timing Information : No timing information (For long sequence of 0s).
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,680 (2 nos)
4. Capacitors -1F,0.1F,0.01F
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 systembandwidth, response speed, capture
and pull in rangemay 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.
62

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

63

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 0.01f 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 1f capacitor, adjust the
amplitude aroung 2Vpp.
8. Connect 0.1 f 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 :
Its 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 2
4
=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.
65

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');
66

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');
67

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 trainers 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 filters
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

72

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)
1. INPUT SIGNAL

2. SAMPLED SIGNAL

3. RECONSTRUCTED SIGNAL

TABULATION: (TIME DIVISION MULTIPLEXING)
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

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