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(17EC29) ACLab Exp 1

The experiment aims to verify the Sampling Theorem by generating a sinusoidal signal, sampling it at rates below and above the Nyquist rate, and reconstructing the signal through interpolation. The Sampling Theorem states that a signal must be sampled at least twice the maximum frequency of the original analog signal in order to reconstruct it accurately without aliasing. The results show that the signal can be reconstructed when the sampling rate satisfies the Nyquist criterion, but aliasing occurs when it is not met.

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74 views3 pages

(17EC29) ACLab Exp 1

The experiment aims to verify the Sampling Theorem by generating a sinusoidal signal, sampling it at rates below and above the Nyquist rate, and reconstructing the signal through interpolation. The Sampling Theorem states that a signal must be sampled at least twice the maximum frequency of the original analog signal in order to reconstruct it accurately without aliasing. The results show that the signal can be reconstructed when the sampling rate satisfies the Nyquist criterion, but aliasing occurs when it is not met.

Uploaded by

Kamlesh
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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KAMLESH (17EC29)

Experiment 1
Object:-
Generate a sinusoidal signal. Sample and reconstruct a signal through interpolation. Vary
the sampling rate below and above the Nyquist rate and hence verify the Sampling theorem.

Theory:-
The real life signals that we encounter in our day to day basis are mostly analog signals. These
signals are defined continuously in time and have infinite range of amplitude values. In order to process
these signals to obtain meaningful information, they need to be converted to a format which is easily
handled by computing resources like microprocessors, computers etc... The first step in this process is to
convert the real-time signal into discrete-time signals. Discrete-time signals are defined only at a
particular set of time instances. They can thus be represented as sequence of numbers with continuous
range of values. The process of converting an analog signal (denoted as x(t)) to a digital signal (denoted
as x(n)) is called the analog -to-digital conversion (referred to as digitization), usually performed by an
analog -to-digital converter (ADC). Here t is the continuous time variable and n is the sequence order. In
many applications after the processing of the digital signal is performed, x(n) needs to be converted
back to analog signal x(t) before it is applied to appropriate analog device. This reverse process is called
digital-to- analog conversion and is typically performed using a digital-to-analog converter (DAC).

The typical block diagram of an ADC is shown in Fig. 1 below.

The process of digitization consists of first sampling (digitization in time) and quantization (digitization in
amplitude). In this experiment we will study and understand the principle of sampling, while the
principle of quantization will be studied in the next experiment. The sampling process depicts an analog
signal as a sequence of values. The basic sampling function can be carried out with an ideal 'sample-and-
hold' circuit which maintains the sampled signal until next sample is taken. An ideal sampler can be
KAMLESH (17EC29)

considered as a switch that periodically opens and closes every T seconds. The sampling frequency (fs in
Hertz) is thus defined as

fs=1T....(1)

The sampled discrete time signal x(nT) , n=0,1,2,.... of the original continuous time signal

x(t) is shown in Fig. 2 below.

In order to represent an analog signal x(t) by a discrete-time signal x(nT) accurately, so ,that the analog
signal can be exactly reconstructed back from the discrete-time signal, the sampling frequency fs must
be at least twice the maximum frequency component (fM) of the original analog signal. Thus we have,

fs≥2fm....(2)

The minimum sampling rate is called the Nyquist rate and the above Sampling Theorem is called the
Shannon's Sampling Theorem. When an analog signal is sampled at fs , frequency components higher
than fs/2 fold back into the frequency range [0, fs/2]. This folded frequency components overlap with
the original frequency components in the same range and leads to an undesired effect known as
aliasing. In this case, the original analog signal cannot be recovered from the sample data.

Consider an analog signal of frequency 1Hz as shown in Fig. 3(a) below. The sampling frequency is 4Hz.
The sampled signal is shown in Fig. 3(b), Note that an exact reconstruction of the missing samples is
obtained so long as the Shannon's Sampling theorem is satisfied.
KAMLESH (17EC29)

Now let's consider, the analog signal of frequency 5Hz as shown in Fig. 4(a) below. The sampling
frequency is same as above, i.e. 4Hz. The sampled signal is shown in Fig. 4(b), Note that the
reconstruction of the original analog signal is not possible since the sampling frequency does not satisfy
Shannon's Sampling Theorem. In this case the reconstructed signal has a frequency of 1Hz. The signal of
5Hz is folded back as 1Hz, into the range determined by the sampling frequency leading to the problem
of aliasing.

Result:-we have successfully construct a sinusoidal signal And Sample and reconstruct a signal through
interpolation and also Vary the sampling rate below and above the Nyquist rate.

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