DIGITAL COMMUNICATION 2017

EXPERIMENT NO. 1
Aim: Study of Sampling theorem and Reconstruction of signal. Verify Nyquist criteria

Apparatus: Model ST 2151 trainer kit, connection wires, DSO, Power supply.

Theory: 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.

In electronics, a sample and hold circuit is used to interface real world, changing analogue signals to a
subsequent system such as an analog-to-digital converter. The purpose of this circuit is to hold the
analogue value steady for a short time while the converter or other following system performs some
operation that takes a little time. In most circuits, a capacitor is used to store the analogue voltage and
an electronic switch or gate is used to alternately connect and disconnect the capacitor from the
analogue input. The rate at which this switch is operated is the sampling rate of the system.

NYQUIST CRITERION (SAMPLING THEOREM):

The Nyquist Criterion states that a continuous signal band limited to fmHz can be completely
represented by and reconstructed from the samples taken at a rate greater than or equal to 2fm
samples/second.
The minimum sampling frequency is calls as NYQUIST RATE i.e. for faithful reproduction

SAMPLE AND HOLD:

One way to maintain reasonable pulse energy is to hold the sample value until the next sample is
taken. This technique is formed as Sample and Hold technique. A buffered Sample and Hold circuit
consists of unity gain buffers preceding and succeeding the charging capacitor. The high input
impedance of the proceeding buffer prevents the loading of the message source and also ensures that
the capacitor charges by a constant rate irrespective of the source impedance.

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Procedure:

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. Connect BNC connector to the DSO and to the trainers output port.
4. Connect Sample Output to fourth order low pass filter Input and Sample and hold Output to second order
low pass filter Input. Observe the output wave form.
5. Switch On the trainer's power supply & Oscilloscope.
6. By pressing Sampling Frequency Selector Switch, change the sampling frequency from 2 KHz, 5 KHz, 10
KHz, 20 KHz up to 40 KHz.
7. Observe how Sample output and Sample and Hold Output changes in
8. each case.
9. Also observe output of second order low pass filter and fourth order low pass filter.

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Observation:
Input Signal(100Hz)

Sampled output Sample and Hold output

Reconstruction of Sampled Wave

2nd Order 4th Order

Conclusion: Hence , In this experiment we observe sampled output of Fm = 1K Hz sinusoidal wave

with different sampling frequencies and we also observe reconstructed signal from second and fourth
order low pass filter.

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EXPERIMENT NO. 2

AIM: To study the PAM (Pulse Amplitude Modulation), PWM (Pulse Width Modulation) &
PPM (Pulse Position Modulation) of analog signal.

APPARATUS: Connecting Probes, DSO, PAM-PWM-PPM Modulation Kit.

THEORY: The aim of pulse modulation methods is to transfer a narrowband analog signal, for
example a phone call over a wideband baseband channel or, in some of the schemes, as a bit stream
over another digital transmission system.

Pulse Amplitude Modulation (PAM):

In pulse amplitude modulation system the amplitude of the pulse is varied in accordance with the
instantaneous level of the modulating signal. Pulse-amplitude modulation (PAM), is a form of signal
modulation where the message information is encoded in the amplitude of a series of signal pulses. It
is an analog pulse modulation scheme in which the amplitudes of a train of carrier pulses are varied
according to the sample value of the message signal. Demodulation is performed by detecting the
amplitude level of the carrier at every symbol period.

Pulse Width Modulation (PWM):

Pulse-width modulation (PWM), or pulse-duration modulation (PDM), is a modulation technique that
controls the width of the pulse, formally the pulse duration, based on modulator signal information.
Although this modulation technique can be used to encode information for transmission, its main use
is to allow the control of the power supplied to electrical devices, especially to inertial loads such as
motors. In addition, PWM is one of the two principal algorithms used in photovoltaic solar battery
chargers.

Pulse Position Modulation (PPM):

In PPM System, the position of the pulse relative to the zero reference level is
varied in accordance with the instantaneous level of the modulating signal. Pulse Position
Modulation (PPM) differs from PWM in that the value of each instantaneous
sample of a modulating wave is caused to vary the position in time of a pulse, relative to its
non-modulated time of occurrence. Each pulse has identical shape independent of the
modulation depth. This is an attractive feature, since a uniform pulse is simple to reproduce
with a simple switching power stage. On the other hand, a limitation of PPM is the
requirements for pulse amplitude level if reasonable powers are required. The power supply
level of the switching power stage will have to be much higher than the required load voltage.
This leads to sub-optimal performance on several parameters as efficiency, complexity and
audio performance.

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Figure : Different types of Pulse Modulation

Figure : PAM, PPM and PWM Trainer Kit

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PROCEDURE:

1. Connect the power cord to the trainer. Keep the power switch in Off position.
2. Connect 2 KHz Sine wave to signal Input.
3. Connect BNC connector to the DSO and to the trainers output port.
4. Connect the high frequency pulses as carrier frequency to the pulse input of the kit.
The pulse trains are available in the range of 4KHz, 8KHz, 16 KHz and 64 KHz.
5. Switch On the trainer's power supply & Oscilloscope.
6. Observe the output waveform at the DSO and compare the change in waveform with
respect to the input sine wave.
7. Now connect the output of the pulse modulation to the input of Low Pass Filter.
8. Connect the output of the Low Pass Filter to the AC amplifier.
9. Observe the output of the AC amplifier to the DSO and compare the demodulated
signal with respect to the input sine wave.

OBSERVATION: Pulse Frequency = 64 KHz

Pulse Amplitude Modulation:

Sampled Output Sample and Hold Output

Flat-Top Output

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Pulse Position Modulation Pulse Width/Duration Modulation

Reconstruction

Low pass filter and AC amplifier

CONCLUSION:

Hence, we studied Pulse Amplitude Modulation, Pulse Width Modulation, Pulse Position
Modulation and reconstructed input sinusoidal signal using Low Pass Filter and AC
Amplifier. We observed amplitude of reconstructed signal is less than input signal when we
used LPF. Using AC amplifier, we can get correct input signal.

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EXPERIMENT NO. 3

AIM: To write a MATLAB program to sample the sinusoidal message signal at

different sampling rate and verify the Nyquist criteria. Also reconstruct the sampled
signal using low pass filter (Using FDA Tools).

APPARATUS: MATLAB Software

THEORY:

Sampling: Sampling is the process in which a continuous time signal is sampled by

measuring its amplitude at discrete instants.

Sampling Theorem: The Sampling Theorem states that a signal whose spectrum is band-
limited to Fm Hz can be reconstructed exactly (without error) from its samples taken
uniformly at a frequency Fs 2Fm (Samples per second).

In other words, the minimum sampling frequency is Fs = 2Fm.

The frequency 2 Fm is called the Nyquist sampling rate.

The corresponding sampling interval Ts = 1/(Fs) is called Nyquist interval.

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MATLAB Code:

clc
clear all
close all
x=-pi/2:.01:pi/2;
fm=input('fm');
y=sin(2*pi*fm*x);
subplot(4,2,1);
plot(x,y);
title('sine signal');
xlabel('time');
ylabel('amplitude');

fs1=2*fm;
t1=-pi/2:.01/fs1:pi/2;
z=sin(2*pi*fm*t1);
subplot(4,2,3);
stem(t1,z);
xlabel('time');
ylabel('amplitude');
title('sampled signal');

fs2=fm/20;
t2=-pi/2:.01/fs2:pi/2;
a=sin(2*pi*fm*t2);
subplot(4,2,5);
stem(t2,a);
xlabel('time');
ylabel('amplitude');
title('undersampled signal');

fs3=3*fm;
t3=-pi/2:.01/fs3:pi/2;
b=sin(2*pi*fm*t3);
subplot(4,2,7);
stem(t3,b);
xlabel('time');
ylabel('amplitude');
title('oversampled signal');

e=filter(fm,1,z);
subplot(4,2,4);

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plot(t1,e);
xlabel('time');
ylabel('amplitude');
title('reconstruction of sampled signal');

c=filter(fm,1,a);
subplot(4,2,6);
plot(t2,c);
xlabel('time');
ylabel('amplitude');
title('reconstruction of undersampled signal');

d=filter(fm,1,b);
subplot(4,2,8);
plot(t3,d);
xlabel('time');
ylabel('amplitude');
title('reconstruction of oversampled signal');

Output waveforms :

CONCLUSION : Thus we have sampled sinusoidal wave and verified Nyquist theorem .

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EXPERIMENT NO: 4

AIM: To study Pulse Code Modulation (PCM), its demodulation and its application in
Time Division Multiplexing (TDM).

APPARATUS: TDM Pulse code modulator & transmitter (ST2153), TDM Pulse code
Demodulator & receiver (ST2154), DSO, testing probes, connecting wires

THEORY:

Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals.
Analog voice data must be translated into a series of binary digits before they can be transmitted.
With PCM, the amplitude of the wave to be transmitted is sampled at regular intervals and
translated into a binary number.

Sampling and quantization of a signal for 4-bit PCM

Time Division Multiplexing (TDM) is a type of digital (or rarely analog) multiplexing in
which switching takes place between two or more signals (mostly PCM signals), serially in
time. In this the time domain is divided into several recurrent time slots of fixed length, one
for each sub-channel.

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Procedure:

1. Set the initial conditions on both the kits.

2. Make connections as shown in Fig. 1.1. as follows:
ON ST2153:
a) ~2 KHz Signal to CH.I Input.
b) ~4 KHz Signal to CH.II Input.

Between ST2153 & ST2154

ST 2153 ST 2154
Tx.Clock output Rx.Clock input
Tx. TO output Rx. TO input
PCM output PCM data input

3. Set the 7- bit pattern for A/D conversion. Observe that the same set of data should be
there for D/A conversion.
4. Vary DC signal (I) and note that the LED's on the A/D converter block on ST2153 & D/A
converter of ST2154 always carries the same code. If you desire to examine the timing of
data flow & control signal in detail, switch the transmitter & receiver into SLOW mode.
5. Observe the two output waveforms at TDM PCM Receiver's CH.I (TP47) & CH.II (TP50)
outputs are distortion less & also observe the LED's in the error check code detector block are
'OFF'.
6. The errors in the system can be introduced with the help of fault switches given on the
techbook.

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Figure 1.1 Connection diagram for PCM

OBSERVATION:
Case 1: DC + DC

Input 1: DC Signal Input 2: DC Signal

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TDM Output TDM PCM Output

Receiver Output:
Output 1: DC Signal Output 2: DC Signal

Case2: AC+DC

Input 1: Dc Signal Input 2 : 2KHz Sine wave

TDM Output TDM PCM Output

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Receiver output:
Output 1: Dc Signal Output 2 : 2KHz Sine wave

Case 3: AC+AC
Input 1: 2KHz Sine wave Input 2 : 4KHz Sine wave

TDM PCM Output:

Receiver output:
Output 1: 2KHz Sine wave Output 2 : 4KHz Sine wave

CONCLUSION: Hence we studied the working of TDM PCM Transmitter and Receiver.

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