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Signals Sampling Theorem

The Signals Sampling Theorem states that a continuous time signal can be accurately represented and recovered from its samples if the sampling frequency is at least twice the highest frequency component of the signal. The document provides a mathematical proof of this theorem, detailing the process of sampling and the implications of aliasing when the sampling frequency is insufficient. To avoid aliasing, it is essential to ensure that the sampling frequency exceeds twice the maximum frequency of the signal and to use anti-aliasing filters.

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5 views3 pages

Signals Sampling Theorem

The Signals Sampling Theorem states that a continuous time signal can be accurately represented and recovered from its samples if the sampling frequency is at least twice the highest frequency component of the signal. The document provides a mathematical proof of this theorem, detailing the process of sampling and the implications of aliasing when the sampling frequency is insufficient. To avoid aliasing, it is essential to ensure that the sampling frequency exceeds twice the maximum frequency of the signal and to use anti-aliasing filters.

Uploaded by

georgesocialmedia64
Copyright
© All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

Signals Sampling Theorem

Statement: A continuous time signal can be represented in its samples and can be recovered
back when sampling frequency fs is greater than or equal to the twice the highest frequency
component of message signal. i. e.

f s ≥ 2f m .

Proof: Consider a continuous time signal x(t). The spectrum of x(t) is a band limited to fm Hz
i.e. the spectrum of x(t) is zero for |ω|>ωm.

Sampling of input signal x(t) can be obtained by multiplying x(t) with an impulse train δ(t) of
period Ts. The output of multiplier is a discrete signal called sampled signal which is
represented with y(t) in the following diagrams:

Here, you can observe that the sampled signal takes the period of impulse. The process of
sampling can be explained by the following mathematical expression:

Sampled signal y(t) = x(t). δ(t) . . . . . . (1)

The trigonometric Fourier series representation of δ (t) is given by


∞
δ(t) = a 0 + Σ (a n cos nω s t + b n sin nω s t) . . . . . . (2)
n=1

Where
1 2 1 1
a0 = ∫ −T
δ(t)dt = δ(0) =
Ts Ts Ts
2
T

2 2 2 2
an = ∫ −T
δ(t) cos nω s dt = δ(0) cos nω s 0 =
Ts T2 T
2

2 2 2
bn = ∫ −T
δ(t) sin nω s t dt = δ(0) sin nω s 0 = 0
Ts Ts
2

Substitute above values in equation 2.

1 ∞ 2
∴ δ(t) = + Σ ( cos nω s t + 0)
Ts n=1 Ts

Substitute δ(t) in equation 1.

→ y(t) = x(t). δ(t)

1 ∞ 2
= x(t)[ + Σ ( cos nω s t)]
Ts n=1 Ts

1 ∞
= [x(t) + 2Σ (cos nω s t)x(t)]
Ts n=1

1
y(t) = [x(t) + 2 cos ω s t. x(t) + 2 cos 2ω s t. x(t) + 2 cos 3ω s t. x(t) . . . . . .
Ts

Take Fourier transform on both sides.

1
Y (ω) = [X(ω) + X(ω − ω s ) + X(ω + ω s ) + X(ω − 2ω s ) + X(ω + 2ω s )+ .
Ts

..]

1 ∞
∴ Y (ω) = Σ n=−∞ X(ω − nω s ) where n = 0, ±1, ±2, . . .
Ts

To reconstruct x(t), you must recover input signal spectrum X(ω) from sampled signal
spectrum Y(ω), which is possible when there is no overlapping between the cycles of Y(ω).

Possibility of sampled frequency spectrum with different conditions is given by the following
diagrams:
Aliasing Effect
The overlapped region in case of under sampling represents aliasing effect, which can be
removed by

considering fs >2fm

By using anti aliasing filters.

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