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8-The Sampling Theorem

The sampling theorem states that a continuous-time signal can be reconstructed from its discrete samples if: 1) The signal is bandlimited, meaning it contains no frequency components above a maximum frequency fm. 2) The signal is sampled at a rate Fs that is at least twice the maximum frequency fm (Fs ≥ 2fm), known as the Nyquist rate. 3) The sampled signal is passed through a low-pass filter with a cutoff frequency equal to fm to reconstruct the original continuous-time signal from the discrete samples.

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71 views8 pages

8-The Sampling Theorem

The sampling theorem states that a continuous-time signal can be reconstructed from its discrete samples if: 1) The signal is bandlimited, meaning it contains no frequency components above a maximum frequency fm. 2) The signal is sampled at a rate Fs that is at least twice the maximum frequency fm (Fs ≥ 2fm), known as the Nyquist rate. 3) The sampled signal is passed through a low-pass filter with a cutoff frequency equal to fm to reconstruct the original continuous-time signal from the discrete samples.

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هاشم عبدالكريم
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The Sampling Theorem

The Sampling Theorem

• Signal sampling is the process of converting the form of a signal from continuous-time to discrete-
time.

• Mathematically, the sampled signal may be represented as the product of the original continuous-
time signal and an impulse train, as shown below. This representation is commonly termed impulse
sampling.

x (t ) x [n ]

s (t ) =   (t − n )
n
x (t ) x (w )

0 w
−Wm Wm

s (t ) S (w )

t w
−2Ws −W s 0 Ws 2Ws
Ts

x [n ] X n (w )

w
−1 0 1 2 3 4 5 6 7 8
nT s −2Ws −W s 0 Ws 2Ws
or
n Figure S1
• Ts is the sampling time (the time spacing between any two successive samples).

• Fs is the sampling frequency

1
fs = (Hz) and Ws+2 fs (rad/s)
Ts

• The FT of the sampled signal is given by an infinite sum of shifted versions of the original signal’s FT.
The shifted versions are offset by integer multiples of Ws, as shown in Fig.S1.
• The sampling theorem is stated as follows:

let x(t) ⎯⎯


FT
→ x (w )
Represent a band limited signal so that x(w)=0 for |w|>wm.
2
If Ws>2Wm, where ws =
Ts
Is the sampling frequency, then x(t) is uniquely determined by its samples x [nTs ], n=0,  1,  2,......

• The minimum sampling frequency, 2wm, is often termed the Nyquist sampling rate or Nyquist
rate.
Where fm is the highest frequency present in the original signal x(t).

• The actual sampling frequency, ws, is commonly referred to as the Nyquist frequency.

• The original signal x(t) can be correctly reconstructed from its sampled version, x[n], if

1. The signal x(t) is band limited


2. fs  2fm Where fm is the highest frequency present in the original signal x(t).

• If one or both of these conditions are violated, then the shifted version of x(w) will overlap
and cause what is called aliasing.
X n (w )

w Ws  2Wm
−2Ws −W s 0 Wm Ws 2W s

X n (w )

−3W s −2W s −Ws


w Ws = 2Wm
0 Wm Ws 2Ws 3W s

X n (w )

Ws  2Wm
−3W s
w
−2W s −Ws 0 Wm Ws 2Ws 3W s 4Ws

aliasing
Ws
• Therefor, x(t) must be bandlimited(filtered) before it is sampled. This filter is usually a LPF and it is
known as the anti-aliasing filter.
Reconstruction
• A replica of x(t) can be reconstructed from x[n] using a LPF with fc=fm, provided that x(t) is
correctly sampled 

X[n] x (t)

LPF
fc = fm

w
n
−1 0 1 2 3 4 5 6 7

X n (w ) 

x (w)
LPF

w w
−W s 0 Wm Ws
= −Wm Wm
Wc

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