EXPERIMENT NO.

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 fm Hz 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 called 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.
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 trainer’s 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 each case.
8. Also observe output of second order low pass filter and fourth order low pass filter.

OBSERVATION:

Input Signal Carrier Frequency Sampled Output Sampled and Hold LPF 2nd order LPF 4th Order
Output
5.02Vpp 1KHz 4.1Vpp 2KHz 4.4Vpp 4.5Vpp 5.12Vpp 4.99Vpp
5.02Vpp 1KHz 4.0 Vpp 40KHz 4.45Vpp 5.08Vpp 4.56Vpp 4.88Vpp

Input wave. Sampling wave

Some of the sampled and sampled and hold output we observed.

Some of the 4th order and 2nd order LPF outputs.

CONCLUSION:

In this experiment we successfully sampled and reconstructed the given input signals and hence
compared it with the original signal also and also observed the output of various stages like sample,
sample and hold and 2nd 4th order LPF.
 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.
Figure: Different types of Pulse Modulation

Figure: PAM, PPM and PWM Trainer Kit

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 trainer’s 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 4 KHz, 8 KHz, 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.

OBSERVATIONS:
CONCLUSION:
In this experiment we studied the PAM (Pulse Amplitude Modulation), PWM (Pulse Width Modulation)
& PPM (Pulse Position Modulation) of analog signal using PAM-PWM-PPM Modulation Kit.
 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 = 2·Fm.
The frequency 2· Fm is called the Nyquist sampling rate.
The corresponding sampling interval Ts = 1/ (Fs) is called Nyquist interval.
MATLAB CODE:

clc

clear all

close all

fm=1

t=1:0.1:10

x=sin(2*pi*fm*t);

subplot(331)

plot(t,x)

title('x(t)')

xlabel('time')

ylabel('amplitude')

%undersampled

fs1=0.5*fm;

t1=0:0.1/fs1:10;

A=sin(2*pi*fm*t1);

subplot(332)

stem(t1,A)

title('undersampled')

xlabel('time')

ylabel('amplitude')

A1=filter(fm,1,A)

subplot(333)

plot(t1,A1)
title('reconstructed undersampled')

xlabel('time')

ylabel('amplitude')

%critically sampled

fs2=fm*2

t2=0:0.1/fs2:10

B=sin(2*pi*fm*t2)

subplot(334)

stem(t2,B)

title('critically sampled')

xlabel('time')

ylabel('amplitude')

B1=filter(fm,1,B)

subplot(335)

plot(t2,B1)

title('reconstructed critically sampled')

xlabel('time')

ylabel('amplitude')

%oversampled

fs3=fm*6

t3=0:0.1/fs3:10

C=sin(2*pi*fm*t3)

subplot(336)
stem(t3,C)

title('over sampled')

xlabel('time')

ylabel('amplitude')

C1=filter(fm,1,C)

subplot(337)

plot(t3,C1)

title('reconstructed oversampled')

xlabel('time')

ylabel('amplitude')

OUTPUT WAVEFORM:
CONCLUSION:
In this experiment we successfully sampled a given signal according to various frequencies to verify
aliasing criteria, Nyquist rate and over sampled conditions and then reconstructed the signal by
passing it through LPF and comparing it with original input wave and observed that it was not a
perfect sine wave in case of aliasing condition and some information was lost and in the rest cases a
good analogous wave was obtained after reconstruction.
 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.
PROCEDURE:

1. Set the initial conditions on both the kits.

2. Make connections as shown in Figure 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 tech book.

Figure 1.1 Connection diagram for PCM

OBSERVATIONS:

Input Signals Multiplexed Output.

Reconstructed Signals

CONCLUSION:
In this experiment we successfully observed the Pulse Code Modulation Technique and observed
various stages of the modulation right from converting the analog signal to digital by sampling than
quantizing it and multiplexing two inputs and converting parallel data to serial and then at the
demodulator side same process executed in a reversed order and reconstructing them and compared
them with the original input waves.
 EXPERIMENT NO. 5

AIM: To study the Modulation and Demodulation of a signal by Pulse Code Modulation using MATLAB
software.

APPARATUS: MATLAB Software

MATLAB CODE:

clc
clear all
close all

A=10
fm=2
t=0:0.01:1;
x = 10+ A*sin(2*pi*fm*t);
subplot(411)
plot(t,x)
title('Message signal')
xlabel('t')
ylabel('amp')

%sampled signal
fs=2*fm;
t1=0:(0.01/fs):1;
y=A+A*sin(2*pi*fm*t1);
subplot(412)
stem(t1,y)
title('Sampled signal')
xlabel('t')
ylabel('amp')

y1=floor(2*y)
subplot(413)
stem(t1,y1)
title('Quantized signal')
xlabel('t')
ylabel('amp')

%y2=floor(y1/3) %will be used for encoding and decoding

y2=dec2bin(y1)

y3=bin2dec(y2) %decoding

y4=filter(2,4,y3) %filter(fm,order of the filter, y4)

subplot(414)
plot(t1,y4)
title('Reconstructed signal')
xlabel('t')
ylabel('amp')
OUTPUT WAVEFORM:

CONCLUSION:
In this experiment we studied the modulation and demodulation of a given sinusoidal message signal
using Pulse Code Modulation and also reconstructed the sampled signal
 DIGITAL COMMUNICATION

S. V. N. I. T. SURAT 22