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Sampling Therom

The Sampling Theorem states that a band-limited signal can be uniquely reconstructed from its samples if the sampling frequency is at least twice the highest frequency present in the signal. Aliasing occurs when the sampling frequency is too low, leading to distortion and overlap in the frequency spectrum. To avoid aliasing, the signal should be band-limited and the sampling frequency must exceed twice the maximum frequency component of the signal.

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2 views

Sampling Therom

The Sampling Theorem states that a band-limited signal can be uniquely reconstructed from its samples if the sampling frequency is at least twice the highest frequency present in the signal. Aliasing occurs when the sampling frequency is too low, leading to distortion and overlap in the frequency spectrum. To avoid aliasing, the signal should be band-limited and the sampling frequency must exceed twice the maximum frequency component of the signal.

Uploaded by

1DA22EC156
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
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ROHINI COLLEGE OF ENGINEERING AND TECHNOLOGY

4.1 SAMPLING THEOREM

It is one of useful theorem that applies to digital communication systems.

Sampling theorem states that “A band limited signal 𝒙(𝒕) with 𝑿(ɷ) = 𝟎 for
|𝑚| ≥ ɷ𝒎 can be represented into and uniquely determined from its samples 𝒙(𝒏𝑻)
if the sampling frequency 𝒇𝒔 ≥ 𝟐𝒇𝒎, where 𝒇𝒎 is the frequency component present
in it”.

(i.e) for signal recovery, the sampling frequency must be at least twice the
highest frequency present in the signal.

Analog signal 𝑥(𝑡) is input signal as shown in Fig 4.1, 𝛿𝑇(𝑡) is the train of
impulse shown in Fig 4.2. Sampled signal 𝑥𝑠(𝑡) is the product of signal 𝑥(𝑡) and
impulse train 𝛿𝑇(𝑡) as shown in Fig 4.2

EC3354 SIGNALS AND SYSTEM


ROHINI COLLEGE OF ENGINEERING AND TECHNOLOGY

Applying Fourier transform on both sides

Spectrum of continuous time signal x(t) with maximum frequency 𝜔𝑚 is shown in


Fig 4.5

EC3354 SIGNALS AND SYSTEM


ROHINI COLLEGE OF ENGINEERING AND TECHNOLOGY

Fig 4.6 Spectrum of 𝑥𝑠 𝑡 when 𝝎𝒔 − 𝝎𝒎 > 𝝎𝒎

Fig 4.7 Spectrum of 𝑥𝑠 𝑡 when 𝝎𝒔 − 𝝎𝒎 = 𝝎𝒎

For 𝝎𝒔 > 2𝝎𝒎


The spectral replicates have a larger separation between them known as guard
band which makes process of filtering much easier and effective. Even a non-ideal
filter which does not have a sharp cut off can also be used.
For 𝝎𝒔 = 𝟐𝝎𝒎
There is no separation between the spectral replicates so no guard band exists and
𝑋(𝜔) can be obtained from 𝑋𝑠(𝜔) by using only an ideal low pass filter (LPF) with
sharp cutoff.

EC3354 SIGNALS AND SYSTEM


ROHINI COLLEGE OF ENGINEERING AND TECHNOLOGY

For 𝝎𝒔 < 2𝝎𝒎


The low frequency component in 𝑋𝑠 𝜔 overlap on high frequency components of 𝑋
𝜔 so that there is presence of distortion and 𝑋 𝜔 cannot be recovered from 𝑋𝑠 𝜔 by
using any filter. This distortion is called aliasing.
So we can conclude that the frequency spectrum of 𝑋𝑠(𝜔) is not overlapped
for 𝝎𝒔 − 𝝎𝒎 ≥ 𝝎𝒎, therefore the Original signal can be recovered from the sampled
signal.
For 𝝎𝒔 − 𝝎𝒎 < 𝝎𝒎, the frequency spectrum will overlap and hence the original
signal cannot be recovered from the sampled signal.
∴ For signal recovery,

i.e., Aliasing can be avoided if 𝑓𝑠 ≥ 2𝑓𝑚


Aliasing effect (or) fold over effect
It is defined as the phenomenon in which a high frequency component in the
frequency spectrum of signal takes identity of a lower frequency component in the
spectrum of the sampled signal.
When 𝑓𝑠 < 2𝑓𝑚 , (i.e) when signal is under sampled, the individual terms in
equation

get overlap. This process of spectral overlap is called frequency folding effect.

EC3354 SIGNALS AND SYSTEM


ROHINI COLLEGE OF ENGINEERING AND TECHNOLOGY

Occurrence of aliasing
Aliasing Occurs if
i) The signal is not band-Limited to a finite range.
ii) The sampling rate is too low.
To Avoid Aliasing
i) 𝑥(𝑡) should be strictly band limited.
It can be ensured by using anti-aliasing filter before the sampler.
ii) 𝑓𝑠 should be greater than 2𝑓𝑚 .
Nyquist Rate
It is the theoretical minimum sampling rate at which a signal can be sampled
and still be reconstructed from its samples without any distortion
𝑁𝑦𝑞𝑢𝑖𝑠𝑡 𝑟𝑎𝑡𝑒 𝑓𝑁 = 2𝑓𝑚 . 𝐻𝑧
Data Reconstruction or Interpolation
The process of obtaining analog signal 𝑥(𝑡) from the sampled signal 𝑥𝑠(𝑡) is
called data reconstruction or interpolation.

The reconstruction filter, which is assumed to be linear and time invariant, has unit
impulse response h(𝑡).
The reconstruction filter, output 𝑦(𝑡) is given by convolution of 𝑥𝑠(𝑡) and h(𝑡).

EC3354 SIGNALS AND SYSTEM


ROHINI COLLEGE OF ENGINEERING AND TECHNOLOGY

𝛿(𝜏 − 𝑛𝑇) 𝑒𝑥𝑖𝑠𝑡 𝑜𝑛𝑙𝑦 𝑎𝑡 𝜏 = 𝑛𝑇


𝛿 (𝜏 − 𝑛𝑇) = 1 𝑎𝑡 𝜏 = 𝑛𝑇
∞

𝑦(𝑡) = ∑ 𝑥 (𝑛𝑇)ℎ(𝑡 − 𝑛𝑇)


𝑛=−∞

Ideal Reconstruction filter


The sampled signal 𝑥𝑠(𝑡) is passed through an ideal LPF (Fig 4.9) with
bandwidth greater than 𝑓𝑚 and a pass band amplitude response of T, then the filter
output is 𝑥(𝑡).
Transfer function of ideal reconstruction filter is

The impulse response of ideal reconstruction filter is


𝑓𝑠
2
ℎ(𝑡) = ∫ 𝑇𝑒 𝑗𝜔𝑡 𝑑𝑓
−𝑓𝑠
2

EC3354 SIGNALS AND SYSTEM


ROHINI COLLEGE OF ENGINEERING AND TECHNOLOGY

Substitute equation 1 in above equation

EC3354 SIGNALS AND SYSTEM

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