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EXPERIMENT 1 - Adi

This document discusses sampling and digitization of analog signals. It explains that according to the Nyquist theorem, an analog signal must be sampled at twice its highest frequency to avoid aliasing. It then provides examples of sampling sinusoidal and square waves at different rates and analyzing the resulting digital Fourier transforms. It shows that sampling below the Nyquist rate leads to aliasing, while sampling above it allows for interpolation by upsampling and low-pass filtering to remove duplicates in the higher frequency bands.

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ARUOS Soura
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49 views9 pages

EXPERIMENT 1 - Adi

This document discusses sampling and digitization of analog signals. It explains that according to the Nyquist theorem, an analog signal must be sampled at twice its highest frequency to avoid aliasing. It then provides examples of sampling sinusoidal and square waves at different rates and analyzing the resulting digital Fourier transforms. It shows that sampling below the Nyquist rate leads to aliasing, while sampling above it allows for interpolation by upsampling and low-pass filtering to remove duplicates in the higher frequency bands.

Uploaded by

ARUOS Soura
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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Sampling of a Sinusoidal Waveform

AIM:
● To sample an Analog signal waveform above its Nyquist sampling rate.
● To obtain DFT of Analog waveform

MATLAB FUNCTION USED


​fftshift, fft, cos, subplot, stem, abs

THEORY
The Nyquist Theorem, also known as the sampling theorem, is a principle that engineers
follow in the digitization of analog signals. For analog-to-digital conversion (ADC) to result in a
faithful reproduction of the signal, slices, called ​samples,​ of the analog waveform must be
taken frequently. The number of samples per second is called the sampling rate or sampling
frequency. Suppose the highest frequency component, in hertz, for a given analog signal is
​ ax​. According to the Nyquist Theorem, the sampling rate must be at least 2​fm
fm ​ ax​, or twice the
highest analog frequency component. The sampling in an analog-to-digital converter is
actuated by a pulse generator (clock). If the sampling rate is less than 2​fm ​ ax​, some of the
highest frequency components in the analog input signal will not be correctly represented in
the digitized output. When such a digital signal is converted back to analog form by a
digital-to-analog converter, false frequency components appear that were not in the original
analog signal. This undesirable condition is a form of distortion called aliasing.

SOURCE CODE AND RESULTS


DISCUSSION
In these 2 cases the sampling frequencies are less than 2*(fmax) thus they do
not satisfy Nyquist Theorem. In this case we can clearly see the effects of
aliasing. In 8khz sampling we see a peak at 0 Hz in the DFT plot. This is
because when we are sampling at lower frequencies the higher frequency parts
of the signal are not correctly sampled which leads to incorrect frequency
components. More accurately this is due to the overlapping of the periodic
copies of the baseband signal. The effect was clearly visible for 4 KHz and 5
KHz.
Spectrum of a Square Wave

AIM:
● To observe the Spectrum of a square wave

MATLAB FUNCTION USED


fftshift, fft, square, subplot, stem, abs

THEORY
A square wave has theoretically has infinite bandwidth. For practical purposes,
the spectrum beyond 10​th ​harmonic can be neglected.

SOURCE CODE AND RESULTS


DISCUSSION
A square wave has theoretically infinite bandwidth so in this case we are
considering uptill the 10th harmonic. This can be done as we can see from the
spectrum that the amplitude of the higher harmonics keep on decreasing. Also
all the amplitudes of the even harmonics are zero. Only the odd harmonics
exists. Their amplitude can be represented as An = 2/( π *n). We observe that
higher frequency components are smaller in magnitude. But this smaller
components give the mean square error when we construct the signal from its
discrete Fourier transform.
Interpolation or Up sampling
AIM:
● To interpolate an Analog Signal.

MATLAB FUNCTION USED


fftshift, fft, interpol, subplot, stem, abs, upsample, lowpass

THEORY
If an analog signal is sampled at a frequency higher than the Nyquist rate it is
possible to interpolate the intermediate L-1 samples or in other words to obtain
the samples at Fs2=LFs1 frequency. This can be simply done by passing the
sampled signals through an ideal low pass filter of cut-off frequency Fmax and
sampling it again at a higher rate.

SOURCE CODE AND RESULTS


DISCUSSION
During up-sampling, we insert zero in between samples which causes
duplication of lower frequencies in higher band. We need filter to filter out
higher frequency range to have our original signal upsampled. Here, in the
case we have used brick wall filter for digital filtering.

Aadi Swadipto Mondal


17EC10065

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