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Sampling

The document describes an experiment to study sampling and reconstruction of signals. Objectives are to demonstrate time domain sampling of band limited signals, signal reconstruction, and sampling in the frequency domain. The experiment involves sampling an input signal using a sampling function and reconstructing the original signal from the sampled output using a filter. The Nyquist sampling theorem states a signal can be reconstructed if sampled at a rate greater than twice its highest frequency.

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arsh
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242 views

Sampling

The document describes an experiment to study sampling and reconstruction of signals. Objectives are to demonstrate time domain sampling of band limited signals, signal reconstruction, and sampling in the frequency domain. The experiment involves sampling an input signal using a sampling function and reconstructing the original signal from the sampled output using a filter. The Nyquist sampling theorem states a signal can be reconstructed if sampled at a rate greater than twice its highest frequency.

Uploaded by

arsh
Copyright
© © All Rights Reserved
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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AIM:

To study the sampling and reconstruction of a signal.

APPARATUS:
Sampling and reconstruction kit
Patch chords
Cathode ray oscilloscope
MATLAB

OBJECTIVES:
To demonstrate the time domain sampling of band limited signal.
To demonstrate the signal reconstruction.
To demonstrate the sampling in frequency domain.

THEORY:
SAMPLING:
The sampling process is a process of converting a continuous time signal to an equivalent
discrete time signal. An analog continuous time signal g(t) is applied at the input of
multiplier. The other input of multiplier is train of pulses. This signal is called as Sampling
Function and denoted by s(t).The output of multiplier is Sampled Signal.

Figure 6.1
RECONSTRUCTION:
A continuous time signal can be processed by processing its samples through a discrete time
system. But we have to reconstruct original signal from discrete signal. This process is known
as reconstruction. For reconstruction process, we use a filter. When sampled signal is passed
through the filter, input signal should be sampled at a sufficient rate that is determined by
sampling theorem.

NYQUIST SAMPLING THEOREM:


If a signal is band limited and its samples are taken at a sufficient rate than those samples
uniquely specify the signal and signal can be reconstructed from these samples. A real signal
whose spectrum is band limited to fb can be reconstructed from its samples taken uniformly
at a rate fs >2fb samples per second. Minimum sampling frequency is fs=2fb for ideal case.
For ideal case, filter is sharply cut off at the frequency fb, only then input signal is obtained
but practically it is not possible.
For non ideal case ,fs>2fb.If this condition is not satisfied,there must be some error in
obtained signal from input signal.
OVERSAMPLING
In signal processing, oversampling is the process of sampling a signal with a sampling
frequency significantly higher than the Nyquist rate. Theoretically, a bandwidth-limited
signal can be perfectly reconstructed if sampled above the Nyquist rate, which is twice the
highest frequency in the signal. Oversampling improves resolution, reduces noise and helps
avoid aliasing and phase distortion by relaxing anti-aliasing filter performance requirements.

UNDERSAMPLING

In signal processing, undersampling or bandpass sampling is a technique where


one samples a bandpass-filtered signal at a sample rate below its Nyquist rate (twice the
upper cutoff frequency), but is still able to reconstruct the signal.When one undersamples a
bandpass signal, the samples are indistinguishable from the samples of a low-
frequency alias of the high-frequency signal. Such sampling is also known as bandpass
sampling, harmonic sampling, IF sampling, and direct IF-to-digital conversion.

PROCEDURE:

Make the connections on the kit.

Firstly switch on the kit.


Select input wave(sine wave or square wave) and applied at the input terminal and set its
voltage and frequency.
Apply pulse signal with frequency satisfying SAMPLING THEOREM at the terminals as
shown in diagram. From CK1 and CK2,take output and apply at the base of the transistor.
Output is taken across the transistor. This is sampled signal of input signal. By this process,
sampling output is obtained.
This output can be seen at CRO screen.
This sampled output is passed through RC network and applied at inverting terminal of IC
741 and non inverting terminal is grounded.
Output is taken across output terminal.
Signal obtained is the original input signal.
This obtained signal can be seen at CRO screen.
By this procedure, signal is sampled into discrete signal and reconstructs the sampled signal
into original signal.
CIRCUIT DIAGRAM:

Figure 6.2

WIRED CIRCUIT DIAGRAM:

Figure 6.3
OBSERVATIONS:
Sampling rate must be greater than twice of the bandwidth of signal g(t) for proper
reconstruction of input signal without any error.
By the process of sampling, continuous time signal is converted into a discrete time domain
signal. And by the process of reconstruction, it is reconstructed back into continuous time
domain signal.

PRECAUTIONS:
Make all the connection properly.
Take CRO observation carefully.

MATLAB Code:
clc;
clear all;
t = 0:0.001:1;
fm = input('Enter the modulating signal frequency = ');
x = sawtooth(2*pi*fm*t);
subplot(4,2,1);
plot(t,x,'black');
xlabel('Time ----->');
ylabel('Amplitude ----->');
title('Message Signal');
fs1 = input('Enter Sampling Frequency < Modulating Signal Frequency = ');
fs2 = input('Enter Sampling Frequency = Modulating Signal Frequency = ');
fs3 = input('Enter Sampling Frequency > Modulating Signal Frequency = ');
%Sampling at fs<<2fm
n = 0:1/fs1:1;
x1 = sawtooth(2*pi*fm*n);
subplot(4,2,2);
stem(n,x1,'green');
xlabel('Time ----->');
ylabel('Amplitude ----->');
title('Undersampledfs<<2fm Signal');
subplot(4,2,3);
plot(n,x1,'green');
xlabel('Time ----->');
ylabel('Amplitude ----->');
title('Reconstructed Undersampledfs<<2fm Signal');
%Sampling at fs=2fm
n = 0:1/fs2:1;
x2 = sawtooth(2*pi*fm*n);
subplot(4,2,4);
stem(n,x2,'red');
xlabel('Time ----->');
ylabel('Amplitude ----->');
title('Sampled at Nyquist Rate fs=2fm Signal');
subplot(4,2,5);
plot(n,x2,'red');
xlabel('Time ----->');
ylabel('Amplitude ----->');
title('Reconstructed Nyquist Rate fs=2fm Signal');
%Sampling at fs>>2fm
n = 0:1/fs3:1;
x3 = sawtooth(2*pi*fm*n);
subplot(4,2,6);
stem(n,x3,'blue');
xlabel('Time ----->');
ylabel('Amplitude ----->');
title('Oversampled fs>>2fm Signal');
subplot(4,2,7);
plot(n,x3,'black');
xlabel('Time ----->');
ylabel('Amplitude ----->');
title('Reconstructed Oversampled fs>>2fm Signal');

Output:
Enter the modulating signal frequency = 15 kHz
Enter Sampling Frequency < Modulating Signal Frequency = 10 kHz
Enter Sampling Frequency = Modulating Signal Frequency = 30 kHz
Enter Sampling Frequency > Modulating Signal Frequency = 500 kHz
Output Waveforms:

Figure 6.4
RESULT:
Sampling and reconstruction of signal has been verified

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