D Y Patiol College of Engineering, Akurdi, Pune

Department of Electronics & Telecommunication

EXPERIMENT NO: 01

Aim: A) AM-Generation (DSB-FC): Calculation of modulation index by graphical method, Power

of AM Wave for different modulating signal.

Apparatus: AM Wave Generation (ST2201) Trainer kit, CRO probes & CRO, Connecting wires,
etc

Theory: The transmitter adopts principle of amplitude modulation in which the instantaneous
amplitude of modulated signal is proportional to the instantaneous amplitude of modulating signal.
The modulating signal is superimposed on the base side of the RF oscillator .Hence this is called
low level AM.

Block Diagram of AM Modulation and Demodulation

AM MODULATION:
A sinusoidal carrier signal is said to be amplitude modulated when its amplitude is varied in
accordance with (i.e. in proportion to) the instantaneous amplitude of the message (i.e. the
modulating) single. If the carrier is described by
Vc (t) = A cos Wct
And the modulating signal be X (t), then the amplitude modulated (A.M.) signal S(t) is
S (t) = A [1 + K x (t)] cos Wct (1)
Where K is some constant.

The spectrum of an AM signal consists of the carrier component (frequency W c) & two sidebands.
The portion of the AM signal spectrum that lies above the frequency W c (and below + Wc) is called
the upper sideband while that below Wc (and above – Wc) is called the lower sideband.

If the modulating signal x (t) is a sinusoidal signal i.e.

X (t) = Em cos Wmt

Then the AM signal S (t) becomes

S (t) = A (1 + K Em cos Wmt) cos Wct
The constant K is chosen such that K. E m does not exceed unity.
The product K Em (usually denoted as m) is called the depth of modulation (or modulation
index). As a result of amplitude modulation to a depth m, the modulated signal amplitude varies
between (1+m) A and (1-m) A. The spectrum of this signal consists of a component at Wc – Wm)
with amplitude mA/2, the carrier component at Wc with amplitude A and another component at (W c
+ Wm) with amplitude mA/2.
A square law modulator makes use of a square - law device, like a transistor amplifier
operating over its nonlinear portion of the characteristic. The input voltage Vi and output current i o
may be related as
io = K Vi + K Vi2
When the input signal Vi is arranged to be the sum of X (t) & Vc (t), the output current consists of
the carrier term at frequency fc (due to the linear term in eq-2) and the sidebands around this
fequency (due to the square term in eq-2.) The amplifier has a load tuned to the carrier frequency
for necessary filtering.

Procedure:

Ensure the following initial conditions exist on the board.

 Audio input select switch should be in INT position.
 Mode switch in DSB position
 Output amplifier’s gain potentiometer in full clockwise position.
 Speakers switch in OFF position.
 Turn on power to the ST2201 board.
 Turn the audio oscillator block’s amplitude pot to its full clockwise maximum.
 Turn the balance pot, in the balanced modulator and band pass filter circuit 1 block. to its fully
clockwise position. It is this block that we will use to perform double sideband (DSB) amplitude
 modulation.
 Monitor, in turn, the two inputs to the balanced modulator & band pass filter circuits 1 block, at
TP1 & TP9.
 The signal at TP1 is the audio frequency sine wave from the audio oscillator block. this the
modulating input to our DSB modulator.
 TP9 carries carrier which is a sine wave of 1 MHz frequency and amplitude of 120 mVpp
approximate. This is the carrier input to our DSB modulator.
 Next, examine the o/p of the balanced modulator & band pass filter circuit 1 block (at TP3),
together with the modulating signal at TP1 trigger the oscilloscope on the TP1 signal.
 Now apply the modulated waveform to the oscilloscope and the modulating signal to the X input.
 Press the XY switch, you will observe the waveform similar to the given below.
 In the waveform T3 measure B and V amplitudes on CRO. Then calculate modulation Index by
following formula;

____ Waveform (T3)

Some common trapezoidal patterns for different modulation indices can be used to calculate
modulation index as shown in the figure below.
TEST POINT WAVEFORMS

TP1. Modulating Sine wave signal: - O/P terminal of Audio Oscillator ( ST 2201)
+

TP9. Carrier Signal: - At O/P terminal of 1MHz Crystal Oscillator ( ST 2201)

TP3. AM modulated Signal: - At AM MOD O/P terminal of Balanced Modulator ( ST 2201)

Observation Table:

Sr. Vmax Vmin Modulation L1 L2 Modulation PT =

No. index= m index = m’ ( EC2/ 2) *
( 1 + (m2/2))

1
2
3
 Waveforms:

 Modulating signal
 Carrier Signal
 AM modulated Waveform ( for under modulation, perfect modulation and over modulation)
 Trapezoidal display (for under modulation, perfect modulation and over modulation)

Conclusion:
 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication
__

EXPERIMENT NO: 02

Aim:A) Frequency modulator using varactor diode and NE566 VCO, calculation of modulation index.
B) FM demodulator using such as IC 565(PLL based).

Apparatus:FM Modem kit,CRO/DSO, Probes, Connecting wires, Multimeter.

Theory:
In case of Frequency Modulation, frequency of high frequency carrier signal varies in
accordance to modulating signal. Frequency modulation has higher noise immunity.In our kit FM
Generator is obtain using function generator IC 8038. Frequency of IC8038 depends on voltage at pin
No. 8. This property of IC8038 is used. Modulating signal is given to pin No. 8 through decoupling
capacitor. O/p of IC 8038 is buffered.For FM demodulation phase lock loop IC 565 is used. Free
running frequency of PLL must be equal to carrier frequency. PLL tracks FM Generator O/p & O/p of
phase detector is demodulated signal. This O/p is filtered using second order low pass filter.

FREQ. ADJ.
+12V FM MODEM
BUFFER
I/P
O/P

IC 8038

DC BIAS

SIGNAL
GEN.
- 12 V AMPL. ADJ.
FREQ. ADJ.
FREQ. ADJ.

I/P O/P
I/P O/P
PLL FILTER
565
GND.

POWER ON M/s. KASHTRONICA

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 Procedure:

1. Switch on the power supply.

2. Adjust carrier i. e. O/p of IC 8038 to 50 Khz using frequency adjust pot.
3. Adjust free running frequency of PLL 565 to 50 Khz using frequency adjust pot.
4. Connect buffered O/p of IC 8038 to I/p of PLL.
5. Now slightly vary the frequency of 8038 & observe its effect on O/p of PLL & verify whether PLL
tracks the frequency of 8038.
6. Now connect O/p of signal generator i. e. modulating signal to I/p of 8038. Keep signal gen. Freq.
minimum & Amplitude 500mV p-p.
7. Observe FM O/p together with modulating signal on CRO. We cannot observe stable o/p, since freq
.of FM o/p changes at every instant. Use spectrum analyzer to study effect of amplitude of
modulating signal on FM o/p. From Bessel’s function calculate modulation index.
8. Observe O/p of PLL & Connect it to filter & observe filter O/p.
9. If you are not getting proper o/p slightly adjust PLL Freq. If there is over modulation reduce the
amplitude of modulating signal
10. Now increase Freqof modulating signal. At higher Freq. o/p of Receiver (After LPF) decreases.
In case of FM modulation index depends on freq. of modulating signal. As freq. increases index
decreases. So in practical systems Pr-emphasis& De-emphasis circuits are used.

Calculation of modulation index without using Spectrum Analyzer:

1. Observe the frequency of 8038 buffered O/p & adjust it to 50 Khz using frequency adjust pot &
measure the voltage at the variable point of potentiometer given on front panel. Say ‘V’ now
suppose we are going to apply modulating signal of amplitude 1 Vp-p & frequency 1 Khz. Due to
this maximum DC voltage at pin 8 of IC 8038 will be V+0.5 & minimum DC voltage will be V-0.5.

V+0.5
V
V-0.5

1. Adjust the DC voltage to V+0.5 using pot & measure the Freq. at the O/P of IC 8038 Say F1.
2. Adjust the DC voltage to V-0.5 using pot & measure the frequency at the O/P of IC 8038. Say F2.
3. Calculate (F2-F1) & maximum deviation from center frequency will be

F = (F2-F1)/2.

Mf= F

Fm

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Calculations:

Conclusion:

SE E&TC SEM Principles of Communication

II Systems
 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication

EXPERIMENT NO: 03

Aim:Verification of Sampling Theorem, PAM Techniques, (Flat top & Natural sampling),
reconstruction of original signal, Observe Aliasing Effect in frequency domain.

Apparatus:PAM kit,CRO, CRO Probes,Connecting wires.

Theory:
According to sampling theorem,sampling rate is rapid enough so that at least two samples are
taken during the course of the period correspondingto the highest frequency spectral i.e. if our
signal frequency is 1 kHz thensampling frequency must be 2 kHz minimum for proper
reproduction oforiginal signal from sampled signal.It is called as nyquist criteria.

There are two types of sampling:

1. Natural Sampling.
2. Flat top Sampling.

1. Natural Sampling: -
In natural sampling the pulse has a finite width.Natural sampling is sometimes called chopper
sampling because the waveform of sampled signalwaveform.Let us consider an analog continuous
time signal to be sampledat the rate of fs Hz and fs is higher than nyquist rate such that
samplingtheorem is satisfied sampled signal s (t) is obtained by multiplication ofsampling function
and input signal x (t).Sampling function c (t) is a trainof periodic pulse of width z and frequency
equal to fs Hz.Thus finally wecan say that the sampled signal consist of sequence of pulse varying
amplitudewhose tops are not flat but follow the instantaneous amplitude of input signal.

2. Flat Top Sampling: -

It is also a practically possible sampling method.Natural sampling is little complex,and it is very
to get flat top samples.The top of the samples remains flat and equal to instantaneous value of base
band signal n (t) at the start of sampling.The duration each sample is Ts and sampling rate is equal
to fs=1/Ts.Normally The width of the pulse in flat top sampling and natural sampling is increased
as far as possible to reduce the band width.Only starting edge of pulse representInstantaneous
value of the mathematically equivalent to the convolution ofInstantaneous samples and
pulse.Finally, it is concluded that a flat toppulse has constant amplitude established by the
sampled value of the signalat some point within the interval.We have arbitrarily sampled the
signal atthe beginning of the sample pulse. To generate flat top sample, the signal is held using a

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hold capacitor after sampling pulse.

Procedure:-
1. Natural Sampling:-
1. Connect crystal CLK output to CLK point of control block.
2. Connect point ‘A’ to sampling S-CLK of natural sampling block &also to hold CLK of
receiver
section.
3. Connect fixed sine wave i.e. O/P of frequency divider block to I/P of natural sampling block.
4.Observe O/P of buffer i.e. sample O/P(PAM O/P).
5.Connect PAM O/P to I/P of receiver.
6.Connect point ‘D’ to ‘C1’ point & observe O/P of buffer (you can observespikes in O/P
signal
waveform).
7. Connect point ‘D’ to ‘C2’ point & observe O/P of buffer (you can observethat O/P is not
properly tracking & some overshoots).
8.Connect O/P of receiver buffer to I/P of LPF & observe LPF O/P.
9.To verify sampling theorem & aliasing effect.Connect variable frequency sine wave to I/P of
natural sampling block & vary signalfrequency from minimum to maximum (aliasing means
conversion of high frequency signal to lower frequency due to sampling).
10.Now connect O/P of 555 CLK to ‘CLK’ I/P of control block andvary the sampling
frequency.

2. Flat top Sampling:-

1.Connect crystal CLK O/P to “CLK” point of control block.
2.Observe point ‘B’ & ‘C’ on dual scope CRO.For flat top sampling I/P waveform is sampled
using sampling pulse‘B’ & its value is hold using capacitor is damped usingpulse ‘C’

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 therefore between time period ‘B’ & ‘C’ we haveget flat top sample.
3.Now connect point ‘B’ to sampling CLK of flat top sampling block & hold ‘CLK’ of
receiver.
4.Also connect point ‘C’ to dump CLK of flat top sample block.
5.Connect fixed sine wave to I/P of flat top sample block & observeO/P of flat top sample
block.
6.Connect O/P of flat top sample block to I/P of receiver.
7.Connect point ‘D’ to ‘C1’ & observe buffer O/P.
8.Also repeat step(1) to (7) for 555 as I/P to CLK tocontrol block.

Waveforms: Draw all the waveforms observed at different test points.

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Conclusion:

SE E&TC SEM Principles of Communication

II Systems
 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication
__

EXPERIMENT NO: 04

Aim:To study Generation and Detection of PWM using IC 555

Apparatus:PWM modulation Demodulation Trainer, Function generator, CRO, Bread Board, Power supply

Theory:
Pulse Width Modulation is an important feature of today’s every microcontroller due to its requirement for
controlling many devices in every field of Electronics. PWM is widely used for motor controlling, lighting
controlling etc. IC 555 timer is used for generating PWM. 555 Timer IC is a very useful and general purpose IC
which can be used in many applications.
555 IC is wired in monostable mode of operation. In this mode the output is LOW (0V) when there is no
triggering, when it is triggered via 2nd pin the output goes HIGH (Vcc) for some time. This time period is
determined by the expression T=1.11 RC. Trigger is applied via a differentiator circuit to make sharp pulses. The
resistor of differentiator is connected to Vcc to generate negative trigger pulses and the diode avoids positive
spikes.
This output is modulated using the input voltage applied at the control pin of the IC. So whenever the trigger pin
pulses become low, the output of the IC switches to high and as a result the discharge transistor is disabled. So
C2 charges through R2.This capacitor keeps on charging until the voltage is above the input control voltage, at
which the IC changes its state. Now the output is low which makes the discharge transistor activated thereby
discharging the capacitor C2.
Hence the output pulse width is determined by the control voltage. This process continues and we get a
continuous stream of pulses which can be used for motor control, driving LED’s, transmitting servo signals for
remote control applications etc.

SE E&TC SEM II Principles of Communication Systems

Procedure:
1. Switch ON the trainer kit.
2. Connect the clock O/P to the I/P of clock terminal of PWM modulation.
3. Connect the AF O/P to the I/P of AF terminal of PWM modulation.
4. Observe the PWM output at pin 3 of 555 IC on CRO.
5. During the demodulation, connect the PWM O/P of PWM modulation to the PWM I/P of PWM demodulation
6. Observe the demodulation output at AF O/P of PWM demodulation on CRO.

Conclusion:

SE E&TC SEM II Principles of Communication Systems

 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication

EXPERIMENT NO: 05

Aim: To study PCM with uniform quantization

Apparatus: Experiment kit, DSO, DMM, Connecting Wires, Probes.

Theory:
Pulse code modulation is a digital transmission of samples of analog signal. In PCM generator we have
sampler, quantizer and encoder. In PCM receiver there is serial data, this serial data is converted into
parallel data and then digital to analog conversion is done. The o/p of DAC is fed into LPF and analog i/p is
reconstructed.PCM performance as an analog system depends on quantization noise introduced by ADC.

Block Diagram:

In PCM modem kit pot provided at the top left side for varying DC bias at the i/p of A/D. We can measure
DC bias voltage at the test point. 8 bit switch is provided for varying bit resolution of A/D. If switch is ON,

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it means bit is going to latch. If LSB bit is OFF it means ground is connected to respective pin of latch.

Procedure:
1. Switch on power supply.
2. Observe 1MHz clock o/p signal on panel and connect this clock to ADC clock i/p.
3. Connect o/p of divide by 2 n/w to i/p of divide by 8 n/w.
4. Connect the o/p of II divide by 8 n/w o/p to SC point.
5. Provide a required i/p signal from function generator to i/p of ADC block.
6. Observe the signal at this point and measure frequency.
7. Connect PCM o/p to i/p of receiver and measure the reconstructed signal at the receiver
8. Pass this signal to LPF and observe the signal at the o/p.
9. Set the biasing DC voltage to 2.5 v with the pot for Tx and Rx.
10. Now keep LSB bits in switch “OFF” one by one and measure the voltage
at i/p of ADC and O/P of DAC.
11. The difference of two voltages will give quantization noise with respect to bits in error.

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Observations:-
Draw the following waveforms:
1. I/p signal
2. Start of Conversion & End of Conversion
3. O/P enable & PCM output
4. LE signal
5. DAC O/P signal
6. LPF o/p.

Conclusion:-

SE E&TC Principles of Communication

SEM II Systems
 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication

EXPERIMENT NO: 06

Aim:- To study PCM with Companding (A law and µ law)

Apparatus:- Experiment kit, DSO, DMM, Connecting Wires, Probes.

Theory: -

In uniform quantization, the step size is fixed. The quantization noise power remains constant. But
signal power is proportional to the square of the amplitude. Hence signal power will be small for weak
signals, but the quantization noise power is constant. Therefore the signal to quantization noise ratio for the
weak signals is very poor. This will affect the quality of the signal. Practically it is difficult to implement the
non-uniform quantization because it is not known in advance about the changes in the signal level. In linear
PCM if Bit resolution is 8 Bits, then there are 2 8 =256 quantization levels. Also, if signal amplitude is
capable of swinging through all available quantization ranges, without extending beyond the outermost
ranges, the output signal to quantization noise ratio is 6N db.
If the signal is reduced in amplitude so that not all quantization ranges are used then S/Nq ratio is
reduced. That means effective number of quantization levels are also reduced. To avoid this problem a
process called COMPANDING is used. COMPANDING means compressing of a signal at transmitter &
expanding of a signal at receiver. Hence the weak signals are amplified & strong signals are attenuated
before applying them to a uniform quantizer. COMPANDING required to improve the signal to quantization
noise ratio of weak signals.
Compression produces signal distortion. To undo the distortion, at the receiver we pass the
recovered signal through an Expander Network. An Expander Network has an input-output characteristic
which is the inverse of the characteristic of the compressor. The inverse distortion of compressor &
expander generate a final O/p signal without distortion.
A] ‘’ law compander: In the U.S., Canada & Japan, ‘’ law is used.
y = +log ( 1 + ’’ IxI )
log (1 + ‘’)

For synthesis of law compander, we have used log, Antilog, Adder, Subtractor circuits using OPAMPs.

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B] ‘A’ law compander: Rest of the world used A law
y =+A [x] For IxI< 1/A
1+ log A
& y= +1+log a[x] For 1/A<IxI< 1
1 + log A
To observe A & law curve directly on CRO , connect sine wave &companded O/p either of A law or  law
to two channels of CRO & press ‘XY’ mode switch, you will observe curve on screen related to that law.
You will observe that near zero, slope of curve is very sharp than higher values which is desired. Also for A
law we can observe break point near zero value due to two equations.
At PCM receiver DAC gain control pot has to be perfectly adjusted to get proper wave shape after
expander block.

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SEM II Systems
Procedure:
1. Switch on the power supply
2. Connect O/p of function Gen. to i/p of A-law compressor block. Keep frequency of Function Gen.
Minimum.
3. Set the voltage to 1V at i/p of the A-law compressor and observe the waveform.
4. Also observe the waveform at the o/p of the compressor.
5. Connect the o/p of the compressor to the i/p of ADC 0809.
6. Observe the PCM o/p at the IC 74151.
7. Connect the o/p of the transmitter to the i/p of receiver and observe the reconstructed waveform at
the DAC.
8. Connect the o/p of the receiver to the i/p of the A-law expander and observe the waveform.
9. Also observe the waveform at the o/p of the expander.
10. Repeat the same process for μ-law.

Observations:-
Waveforms to be observed-
1. I/P signal of the compressor
2. O/P signal of the compressor
3. PCM O/P
4. Reconstructed signal at the DAC o/p.
5. O/p signal of the Expander.
6. O/p of LPF.
7. I/O Vs O/P graph for compressor & expander
Observation Table:

Conclusion:

SE E&TC Principles of Communication

SEM II Systems
 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication

EXPERIMENT NO: 07

Aim:- To study Delta Modulation &Demodulation, observe effect of slope overload & granular noise.
Apparatus:- Experiment kit, DSO, Connecting Wires, Probes.

Theory:-

Delta modulation (DM) is a DPCM scheme in which the difference signal ∆(t) is encoded into just a
single bit. The single bit provides for just two possibilities, is used to increase or decrease the estimate. The
baseband signal s(t) and its quantized approximation š(t)are applied as inputs to comparator. A comparator,
as its name suggests, simply makes a comparison between inputs. The comparator has one fixed o/p V(H)
when s(t)> š(t) and a different o/p V(L) when s(t)<š (t). When the comparator o/p is V(H), it will be
encoded as 1 else as 0.

Linear DM has little complexity compared to PCM yet it suffers from severe limitations on account
of which it finds almost no applications in real system. The major disadvantages of DM are slope overload
& granular noise.
DM will transmit speech without significant slope overload provided that the DM system is able to
transmit a sinusoid of frequency f=800 Hz whose amplitude A and frequency f has a maximum slope of
2πfA as it passes through zero. If slope overload is to be avoided then we require that the sampling rate fs
must satisfy the condition Sfs ≥ 2πfA or fs ≥ πf . 2A/S, where ‘S’ is the step size, ‘f’ in the signal frequency

SE E&TC SEM II Principles of Communication Systems

and ‘A’ is the amplitude of the signal.

Block Diagram:

SE E&TC SEM II Principles of Communication Systems

Procedure:-
1. Switch on the power supply.
2. Connect the o/p of fixed frequency generator to non inverting terminal of comparator.
3. Connect pt. A on the panel to pt. B.
4. Connect DAC o/p to INV terminal of comparator & to the LPF.
5. Observe the DM o/p for a given fixed freq. sine wave.
6. Observe fixed freq. sine wave & DAC o/p simultaneously.
7. Observe OE of both adder & subtractor section.
8. Observe the o/p of LPF w. r. to i/p.
9. To observe slope overload error, connect triangular i/p of variable function generator to non
inverting terminal & observe DAC o/p.
10. To observe granular noise, connect square wave i/p of variable function generator to non-inverting
terminal & observe DAC o/p.

Observations:
Draw following waveforms:
1. Input signal
2. Sample Clock
3. DAC output
4. DM output
5. OE of both the adder and subtractor block.
6. LPF output.
7. Output of DAC for triangular and square inputs for input frequency of 1KHz.
Waveforms:

SE E&TC SEM II Principles of Communication Systems

Conclusion:

SE E&TC SEM II Principles of Communication Systems

 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication

EXPERIMENT NO: 08

Aim:- To study adaptive delta modulation & demodulation

Apparatus:- Experiment kit, DSO, Connecting Wires, Probes.

Theory:-

ADM is an extension of delta modulation. By varying the step-size in accordance with the input
signal, the delta modulator is enabled to cope with changes in the input signal.
ADM tracks changes in the sinusoidal input signal much better than DM. This improvement in the
performance of ADM is due to adaptation of the step size in successive iterations of the algorithm. In
particular, the reduced step size of the ADM results in smaller quantization errors near the extremities of the
input signal than the DM. However, both modulation schemes produce comparable quantization errors in
regions of the input signal where the slope is moderately high.
The improved tracking performance of the ADM results in an output signal with a much lower bit
rate, on the average, than the DM.
Block Diagram:

SE E&TC SEM II Principles of Communication Systems

Procedure :-
1. Switch on the power supply.
2. Connect the o/p of fixed freq. gen. to non-inverting terminal of comparator.
3. Connect pt. A on the panel to pt. B and pt. C to pt. D.
4. Connect DAC o/p to inverting terminal of comparator & to the LPF I/p.
5. Observe the ADM o/p for a given fixed freq. sine wave.
6. Observe fixed freq. sine wave & DAC o/p simultaneously.
7. Observe pt. A & C simultaneously.
8. Observe RST of both the counters and CLK.
9. Observe OE of adder or subtractor section.
10. Observe LE of main latch 74373 along with OE of adder or subtractor.
11. Observe the o/p of LPF w. r. to i/p.
12. Connect high frequency signal i/p of variable function generator to non- inverting terminal &
observe DAC o/p.
13. Connect square wave i/p of variable function generator to non-inverting Terminal & observe
DAC o/p.

SE E&TC SEM II Principles of Communication Systems

Observations:
Draw following waveforms:
1. Input Signal
2. Sample Clock
3. DAC output
4. ADM o/p
5. Point A & C simultaneously.
6. OE of adder or subtractor section.
7. LE of main latch 74373 along with OE of adder or subtractor.
8. O/P of LPF with respect to input.
9. Observe the effect of variable step size on slope overload error

Conclusion:

SE E&TC SEM II Principles of Communication Systems

 D. Y. Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication

EXPERIMENT NO: 09

Aim:- To study different Line coding formats / Data formats.

Apparatus:- Experiment kit, DSO, Connecting Wires, Probes, etc.

Theory:-

In this kit we have provided two different bit patterns to study different data formats. [RZ, NRZ,
Polar RZ, Bipolar NRZ and Biphase or splitphase]
1. RZ :- In case of RZ i.e. return to zero formats, if bit is ‘1’ then logic high level is transmitted for first
half bit period and logic low level for remaining half bit period is transmitted.

2. NRZ :- In case of NRZ i.e. not return to zero if bit is ‘1’ then logic high level is transmitted for full bit
period and if bit is ‘0’ then logic low level is transmitted.

3. POLAR RZ :- If bit is ‘1’ then logic high level is transmitted for first half bit period and then low level
for remaining half bit period. If bit is ‘0’ then negative high level is transmitted for first half bit period
and then low level for remaining half bit period.

4. BIPOLAR NRZ :- If bit is ‘1’ then logic high level is transmitted for full bit period. If next bit (not
necessary consecutive) is also ‘1’ then negative high level is transmitted i.e. for every ‘1’ sign of high
level is altered. If bit is ‘0’ then logic low level is transmitted.

5. SPLITPHASE :- If bit is ‘1’ then logic high level is transmitted for first half bit period and then
negative high level for remaining half bit period. If bit is ‘0’ then negative high level is transmitted for
first half bit period and then high level for remaining half bit period.

SE E&TC SEM II Principles of Communication Systems

Line coding Waveforms:

PSD of line codes:

SE E&TC SEM II Principles of Communication Systems

SE E&TC SEM II Principles of Communication Systems
Procedure:-
1. Switch on the power supply.
2. Connect one of the bit patterns as I/p to data format section.
3. Observe bit pattern together with different data formats on DSO.
4. Repeat the procedure for other bit patterns.

Observations:
Draw all the input & output waveforms of all line codes along with its PSD.

Wavrforms:

Conclusion:

SE E&TC SEM II Principles of Communication Systems

 D Y Patil College of Engineering, Akurdi, Pune
Department of Electronics & Telecommunication

EXPERIMENT NO. 10

Aim:To write a MATLAB program to implement Pulse Code Modulation.

Software: OCTAVE/ MATLAB

Theory: Pulse code modulation is a digital transmission of samples of analog signal. In PCM generator we
have sampler, quantizer and encoder. In PCM receiver there is serial data, this serial data is converted into
parallel data and then digital to analog conversion is done. The o/p of DAC is fed into LPF and analog i/p is
reconstructed.PCM performance as an analog system depends on quantization noise introduced by ADC.

Block Diagram:

SE E&TC SEM I Principles of Communication Systems

SE E&TC SEM I Principles of Communication Systems
SE E&TC SEM I Principles of Communication Systems
Conclusion:

SE E&TC SEM I Principles of Communication Systems

 D Y Patil College of Engineering, Akurdi, Pune
Department of Electronics & Telecommunication

EXPERIMENT NO. 11

Aim:

Write a MATLAB program to verify the sampling theorem by sampling an analog signal with
various sampling frequenciesi.e Fs=2Fm, Fs ≥ 2Fm and Fs< 2Fm.

Software:
OCTAVE / MATLAB

Theory:

In Digital Signal Processing, the signal to be transmitted must be in a discrete time form. The
message signal can be in analog form e.g. speech or video signal. In such a case the analog signal
should be first converted into a discrete time signal. “Sampling Process” is used for this purpose.
While using this sampling process practically, the sampling rate should be properly selected.

The sampling process should satisfy the following requirement: -

 Sampled signal should represent the original signalfaithfully.

 It should be possible to reconstruct the original signal from its sampledversion.

Sampling Theorem:

In order to represent the original signal faithfully, it is necessary to take as many samples as
possible. Higher the no. of samples, closer is the representation. The no of samples depends on
the sampling rate & maximum frequency of the signal to be transmitted.

Nyquist Criteria:

A continuous time signal x (t) can be completely represented in its sampled form and recovered
back from its sampled form if the sampling frequency is greater than or equal to the input signal
frequency.i.e.

Fs ≥ Fm

Where, Fs= Sampling frequency

SE E&TC SEM I Principles of Communication Systems

Fm= Max Input frequency

Continuous timeanalogsignal sampled

signal Spectrum of sampled signal1

Wm=Inputfrequency
Ws =Sampling frequency

Aliasing or Fold Over Error:

If the analog signal to be sampled is not strictly band limited and if the sampling frequency is
less than twice the input frequency, an error called aliasing or foldover error is observed. This
distortion occurs due to the overlapping of the adjacent spectrums in the sampled signal. Due to
this, high frequencies are reflected into the low frequencies. This phenomenon is called as
aliasing. Due to aliasing some of the information contained in the original signal is lost in the
process of sampling.

SE E&TC SEM I Principles of Communication Systems

SE E&TC SEM I Principles of Communication Systems
Aliasing Can Be Prevented By:

 The analog signal is passed through the band limiting (anti-aliasing) filter,
beforesampling.

 The sampling frequency is always kept greater than twice the input frequency
(Oversampling). Due to this, even though the analog signal is not strictly band
limited, the spectrums will not overlap. Guard band is provided between the adjacent
spectrums as shown inthe fig.

Algorithm:

1. Define the input analog signal frequency(Fm).

2. Plot the continuous timesignal.

3. Take the sampling frequency less than (2*Fm) and plot the sampled signal & serve
the aliasing effect.

4. Take the sampling frequency equal to (2*Fm) and plot the sampledsignal.

5. Take the sampling frequency (fs) greater than (2*Fm) and sample the analog signal.
Plot the sampledsignal.

Functions used in MATLAB Program:

Conclusion:

SE E&TC SEM I Principles of Communication Systems

 D Y Patil College of Engineering, Akurdi.
Department of Electronics & Telecommunication

Experiment NO. 12

Aim: To write a MATLAB program todemonstrate Scrambling and descrambling operation.

Software: OCTAVE/ MATLAB

Theory: A scrambler also referred to as a randomizer is a device that manipulates a data stream before
transmitting. It replaces sequences into other sequences without removing undesirable sequences, and
as a result it changes the probability of occurrence of long string of 0 or 1. It is an algorithm that
converts an input string into a seemingly random output string of the same length, thus avoiding long
sequences of bits of the same value, in this context, a randomizer is also referred to as a scrambler.
Scrambler is used to enable accurate timing recovery on receiver equipment without resorting to
redundant line coding. It facilitates the work of a timing recovery circuit, an automatic gain control and
other adaptive circuits of the receiver eliminating long sequences consisting of '0' or '1' only.Scrambling
is widely used in satellite, radio relay communications and PSTN modems. It can be placed just before
a FEC coder, or it can be placed after the FEC, just before the modulation or line code. The
manipulations are reversed by a descrambler at the receiving side.

At the transmitter side,

m1=m3 xor m4
m2=m xor m1

At the receiver side

m1=m3 xor m4
m=m2xor m1

 Scrambling-Coding operation applied at Tx to randomize a bitstream

SE E&TC SEM II Principles of Communication
Systems
  Eliminates long strings of like bits that might impair Rx synchronization
 Some bit synchronizers require zero crossings in signal for their operation
 Avoids production of undesirable discrete frequency components and DC in the power spectrum
 Implemented using Tapped shift registers
 Exact operation depends upon SR configuration
 Unscrambler has the reverse structure of the scrambler
 Susceptible to error propagation.

Functions used in MATLAB Program:

Conclusion:

SE E&TC SEM II Principles of Communication

Systems