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Sampling

A tutorial and derivation of the sampling theorem
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0% found this document useful (0 votes)
4 views6 pages

Sampling

A tutorial and derivation of the sampling theorem
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

Sampling and Reconstruction

Introduction
This section introduces the relationship between continuous signals/systems and their representation as
discrete-time signals/systems that allows the direct use of digital computing technology. Sampling forms
the bridge between these two representations of the world.

Sampling
Analog Discrete-Time
System System
Reconstruction

A Discrete-Time system has as its input and output sequences of values. Here it is assumed that these
sequences represent samples of analog signals that are equally spaced in time (Ts seconds apart). Extensions
to applications involving functions of other variables (e.g. distance, force, pressure. velocity, pixels, …) will
be apparent.

x(n*Ts) y(n*Ts)
Discrete-Time
System
xn yn

In order to develop the analysis tools for such a system, we must form an equivalent analog system.
x(t) x(t, Ts)
y(t)
Sample Re-Sample Reconstruct
(Ts sec.) Analog (Ts sec.)
System
Where:
x n = x (n * Ts ) = x (t ) t = n*T
s

and
y n = y (n * Ts ) = y (t ) t =n*T
s

Analyzing this equivalent system using standard analog tools will establish a set of discrete-time tools
including the Discrete Fourier Transform (DFT) and the z-Transform.
Another reason for examining the sampling/reconstruction process is that they form the actual hardware
interface between analog and discrete-time systems. We will then be prepared to perform Digital Filtering
of analog signals.

J. N. Denenberg Copyright DTS September 27, 2001


Sampling Tutorial Page 2 of 6

Sampling
Pulse Amplitude Modulation
PAM is where the analog signal after sampling takes on the form of a sequence of pulses Ts seconds apart.
Each pulse carries information about the analog signal’s amplitude at the time the pulse was generated.
There are two variants of PAM. The first, currently the dominant approach, called Natural Sampling is
where the shape of each pulse is affected by the changing input during the pulse. The second is called
Uniform Sampling (mainly used in the older laboratory grade samplers) where all of the pulses have the
same shape, but different amplitudes. We will analyze both.
Natural Sampling in the Time Domain
Assume that we have a generalized, time-limited pulse centered at t = 0 as shown below.

p(t)

-2Ts -Ts 0 Ts 2Ts

The Fourier Transform of this pulse is to denoted P(jω). A periodic version of this pulse where the original
pulse now repeats every Ts seconds is then:

J. N. Denenberg Copyright DTS September 27, 2001


Sampling Tutorial Page 3 of 6

pT(t)

1.2
A
1
0.8
0.6
0.4
0.2
0
-30s
-2T -25 -20 -15
-Ts -10 -5 0 5 10 15
Ts 20 25 30s
2T

[ ]
Note: lim pTs (t ) = p (t )
Ts → ∞

pTs (t ) , the pulse train, is periodic with period Ts and we can express it as a sum of individual pulses.

pTs (t ) = ∑ p(t − n * T ) s
n = −∞

“Natural Sampling” is then done by multiplying x(t), the input signal, by pTs (t )

x(t) ∞
x (t , Ts ) = ∑ x (t )* p(t − n * T )
*
s
n = −∞

pTs (t )

J. N. Denenberg Copyright DTS September 27, 2001


Sampling Tutorial Page 4 of 6

Uniform Sampling in the Time Domain


The second type of sampling, Uniform Sampling, results in a train of pulses in which each pulse has the
same shape. Each pulse has a “strength” which is set by the value of the input at the sampling “instant”.
This method also goes by the name “Flat Top Sampling” when the transmitted pulses are rectangular. One
way to implement uniform sampling is to precede a natural sampler with a “Sample and Hold” circuit.

x(t) xsh(t) ∞
x (t , Ts ) = ¦ x (n * T )* p(t − n * T )
S&H
*
s s
n = −∞

pTs (t )

The Sample and Hold circuit has the following effect on an analog signal.

2.5

1.5
xsh(t)
x(t)
1

0.5

0
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
1
2
3
4
5
6
7
8
9

This is a non-linear distortion of the signal, but now the signal is constant during each pulse of the pulse
train and multiplying leaves all of the resulting pulses having the same shape. A delay of ½ Ts must be
introduced in the pulse train to center each pulse in a constant region of the modified signal. This has the
effect of delaying the signal itself by Ts/2.

J. N. Denenberg Copyright DTS September 27, 2001


Sampling Tutorial Page 5 of 6

Frequency Domain Analysis of Sampled Signals


Introduction
Both types of sampling are practical. One, Natural Sampling, is simpler in the frequency domain while the
other, Uniform Sampling, is simpler in the time domain. In the limiting case, as the pulse width approaches
zero while holding the pulse area constant, both systems yield the same result. This limiting case is called
Ideal (or Impulse) Sampling where each pulse is now the “Impulse” or “Dirac Delta” function, δ (t − n * Ts ) ,
and is the form of sampling described in many textbooks.
Natural Sampling in the Frequency Domain
Let P(jω) be the Fourier transform of p(t), the single sampling pulse of width ∆t < Ts. The Fourier series for
the periodic pTs (t ) is:


pTs (t ) = ∑ P *ε n
jnω s t
ωs = , the sampling rate in radians per second.
n = −∞ Ts
Where:
Ts
1 1 1  2π 
Pn = ∫2 p (t ) * ε − jnω s t dt = P ( jnω s ) = P  jn 
Ts  Ts 
T
Ts − s
2
Ts
Which results in:

 2π   2π 
{
ℑ pTs (t ) = } 1
Ts
∑ P jn T  * δ  ω −
Ts
 = PTs (ω )
n = −∞  s   

1.2

1
ω)| -------
|P(jω

0.8

0.6

0.4

0.2

0
1 2 3 4 5 6 7 8 9 10 11 12 13
-0.2

-0.4

Where the height of each “bar” denotes the strength of the delta function at that frequency.

J. N. Denenberg Copyright DTS September 27, 2001


Sampling Tutorial Page 6 of 6

Since the Fourier Transform of a product of two time functions is the convolution of their individual Fourier
Transforms. The Fourier Transform of the sampled version of the input, x(t), is:

1 ∞
 2π    2π 
X Ts ( jω ) = ∑ P  jn
 T   * X  j  ω −  = PTs ( jω )
Ts n = −∞  s    Ts 

1.2
ω)|
|X(jω

1
ω)| -------
|P(jω
0.8
ω-ω
|X(jω ωο)|

0.6

0.4
ω-2*ω
|X(jω ωο)|

0.2

0
1
5
9
13
17
21
25
29
33
37
41
45
49
53
57
61
-0.2

-0.4

Notes:
1. Each repetition of X(jω), the Fourier Transform of x(t), is identical. It only varies in magnitude
according to the magnitude of P(jω) at the center frequency of that repetition.
2. If ωs (or fs) increases (i.e. Ts decreases), the repetitions remain equally spaced in the frequency
domain, but move further apart.
Aliasing
TBD
Uniform Sampling in the Frequency Domain
TBD

J. N. Denenberg Copyright DTS September 27, 2001

Common questions

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In Natural Sampling, the shape of each pulse is influenced by the changing input during the generation of the pulse, resulting in pulses of varying shapes that reflect the input signal's amplitude changes. In contrast, Uniform Sampling involves pulses with identical shapes; however, the amplitude or 'strength' of each pulse reflects the input signal's value at the sampling instant .

Uniform Sampling offers practical advantages in the time domain by simplifying signal processing. It uses consistent pulse shapes, making implementation easier and reducing the complexity in signal recording and manipulation. Its use of the 'Sample and Hold' circuit simplifies the retention of amplitude values, albeit at the cost of adding non-linear distortion, as opposed to Natural Sampling which requires accommodating changing pulse shapes .

Convolution is critical in the Fourier Transform of sampled signals because it combines the spectral properties of the sampling pulse and the input signal, allowing analysis of how the signal's frequency content is distributed and repeated in the frequency domain. The convolution relates to the individual Fourier Transforms by distributing the input signal's spectrum across the frequency domain according to the sampling pulse's bandwidth, determining how the sampled signal is structured and affecting reconstruction quality .

Reconstructing a sampled signal back into a continuous-time signal involves interpolating the discrete values to form a smooth, continuous waveform. This process is significant because it allows for the original analog signal to be regained from its digital samples, vital in converting processed digital data back into a form usable by analog systems, such as speakers or displays. Correct reconstruction is contingent upon sampling at a rate above the Nyquist frequency to avoid information loss due to aliasing .

The sampling rate (ωs or fs) impacts the spacing of spectral repetitions in the frequency domain. As ωs increases or Ts decreases, the repetitions of the signal's spectrum move further apart, minimizing overlapping and aliasing issues. Insufficient sampling rates can lead to aliasing, where higher frequency components fold back into the lower frequency range, distorting the reconstructed signal .

The 'Sample and Hold' circuit in Uniform Sampling ensures that each pulse in the sample train has a uniform shape but modulates its amplitude based on the sampled signal's value. It introduces a non-linear distortion by making the signal constant during each pulse, and a delay of ½ Ts is added to the pulse train to align each pulse in a region where the sampled signal remains constant, effectively delaying the reconstructed signal by Ts/2 .

Sampling bridges the gap between continuous signals and discrete-time systems by allowing continuous analog signals to be represented as sequences of discrete values, which can then be processed using digital computing technology. This process is essential for digital filtering of analog signals and forms the hardware interface between analog and digital systems .

The periodic extension of sampling pulses contributes to analyzing sampled signals by allowing the representation of the sampled signal as a sum of individual pulses repeated every sampling period Ts. This helps to understand and model the sampling process mathematically, making use of the Fourier series expansion to express the periodic pulse train and analyze its impact on both time and frequency domains, enabling efficient processing and reconstruction of signals .

Ideal Sampling functions as a limiting case where the pulse width approaches zero while maintaining constant pulse area, resulting in both Natural and Uniform Sampling yielding similar outputs. It utilizes the Dirac Delta function (Impulse function), where each pulse in the signal is represented as an 'Impulse' .

The Fourier Transform of the sampled input signal is crucial because it enables the conversion between time and frequency domain representations. It involves convolving the Fourier Transform of the sampling pulse with the Fourier Transform of the input signal, repeating the spectrum at intervals of the sampling frequency. This repetition helps analyze how well the signal can be reconstructed and the effects of aliasing if the sampling rate is insufficient to cover the whole spectrum of the original signal .

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