Objectives:: The Sampling Theorem

ECE
EE 8443
3512 – PatternContinuous
– Signals: Recognition
and Discrete
LECTURE 18: THE SAMPLING THEOREM
• Objectives:
Representation Using Impulses
FT of a Sampled Signal
Signal Reconstruction
Signal Interpolation
Aliasing
Multirate Signal Processing
• Resources:
Wiki: Nyquist Sampling Theorem
CNX: The Sampling Theorem
CNX: Downsampling
URL:
Representation of a CT Signal Using Impulse Functions
• The goal of this lecture is to convince you that bandlimited CT signals, when
sampled properly, can be represented as discrete-time signals with NO loss of
information. This remarkable result is known as the Sampling Theorem.
• Recall our expression for a pulse train: x(t)

p(t )    (t  nT ) … …
n   t
• A sampled version of a CT signal, x(t), is: -2T -T 0 T 2T
 
x s (t )  x(t ) p (t )   x(t )  t  nT    x(nT )  t  nT 
n   n  
This is known as idealized sampling.

• We can derive the complex Fourier series of a pulse train:

p(t )  c e
k  
k
jk 0t
where  0  2 / T
 
T /2 T /2
1 1 1  jk0t 1
T T/ 2 T T/ 2
 jk0t  jk 0t
ck  p (t ) e dt   (t ) e dt  e t 0 
T T

1 jk0t
p(t )  
k   T
e
EE 3512: Lecture 18, Slide 2

Fourier Transform of a Sampled Signal
• The Fourier series of our sampled signal, xs(t) is:

1
x s (t )  p(t ) x(t )   x(t )e jk0t
k   T
• Recalling the Fourier transform properties of linearity (the transform of a sum

is the sum of the transforms) and modulation (multiplication by a complex
exponential produces a shift in the frequency domain), we can write an
expression for the Fourier transform of our sampled signal:
    1
X s e  F p (t ) x(t )  F   x(t )e
j jk0 t 
1 

   F x(t )e 0
jk t
k   T  T k  
1 
  X (e j (  k0 ) )
T k  
• If our original signal, x(t), is bandlimited: X (e j )  0 for   B

Signal Reconstruction
 
• Note that if  s  2 B , the replicas of X e j do not overlap in the frequency
domain. We can recover the original signal exactly.
• The sampling frequency,  s  2 B , is referred to as the Nyquist sampling

frequency.
• There are two practical problems associated with this approach:
 The lowpass filter is not physically realizable. Why?
 The input signal is typically not bandlimited. Explain.
Signal Interpolation
• The frequency response of the lowpass, or interpolation, filter is:
T ,  B    B
H ( e j )  
 0, elsewhere
• The impulse response of this filter is given by:
BT sin  Bt /   BT
h(t )   sinc (Bt/πB    t  
  Bt /   
• The output of the interpolating filter is given by the convolution integral:

y (t )  h(t ) * x s (t )   x ( )h(t   )d

s
 
   
    x(nT )  t  nT  h(t   )d    x(nT )  t  nT  h t   d
  n     n  
 
   x(nT )  t  nT  h t   d
n    
• Using the sifting property of the impulse:
 
y (t )    x(nT )  t  nT  h t   d
n    

  x(nT )h t  nT 
n  

Signal Interpolation (Cont.)
• Inserting our expression for the
impulse response:
BT  B
y (t )  
 n 
x ( nT ) sinc (

(t  nT ))
• This has an interesting graphical
interpretation shown to the right.
• This formula describes a way to
perfectly reconstruct a signal from
its samples.
• Applications include digital to
analog conversion, and changing
the sample frequency (or period)
from one value to another, a process
we call resampling (up/down).
• But remember that this is still a
noncausal system so in practical
systems we must approximate this
equation. Such implementations
are studied more extensively in an
introductory DSP class.
Aliasing
• Recall that a time-limited signal cannot be bandlimited. Since all signals are
more or less time-limited, they cannot be bandlimited. Therefore, we must
lowpass filter most signals before sampling. This is called an anti-aliasing
filter and are typically built into an analog to digital (A/D) converter.
• If the signal is not bandlimited distortion will occur when the signal is
sampled. We refer to this distortion as aliasing:
• How was the sample frequency for CDs and MP3s selected?

Sampling of Narrowband Signals
• What is the lowest sample frequency
we can use for the narrowband signal
shown to the right?
• Recalling that the process of
sampling shifts the spectrum of the
signal, we can derive a generalization
of the Sampling Theorem in terms of
the physical bandwidth occupied by
the signal.
• A general guideline is 2 B  f s  4 B , where B = B2 – B1.
• A more rigorous equation depends on B1 and B2:

r f  B/2
f s  2B where r   c
r B
and
f c  ( B1  B2 ) / 2
r   r  (greatest integer greater than or equal to r )
• Sampling can also be thought of as a modulation operation, since it shifts a

signal’s spectrum in frequency.
Undersampling and Oversampling of a Signal

Sampling is a Universal Engineering Concept
• Note that the concept of
sampling is applied to many
electronic systems:
 electronics: CD players,
switched capacitor filters,
power systems
 biological systems: EKG,
EEG, blood pressure
 information systems: the
stock market.
• Sampling can be applied in
space (e.g., images) as well
as time, as shown to the
right.
• Full-motion video signals are sampled spatially (e.g., 1280x1024 pixels at 100
pixels/inch) , temporally (e.g., 30 frames/sec), and with respect to color (e.g.,
RGB at 8 bits/color). How were these settings arrived at?

Downsampling and Upsampling
• Simple sample rate conversions, such as converting from 16 kHz to 8 kHz,
can be achieved using digital filters and zero-stuffing:

Oversampling
• Sampling and digital signal processing can be combined to create higher
performance samplers 
• For example, CD players use an oversampling approach that involves
sampling the signal at a very high rate and then downsampling it to avoid the
need to build high precision converter and filters.

Summary
• Introduced the Sampling Theorem and discussed the conditions under which
analog signals can be represented as discrete-time signals with no loss of
information.
• Discussed the spectrum of a discrete-time signal.
• Demonstrated how to reconstruct and interpolate a signal using sinc
functions that are a consequence of the Sampling Theorem.
• Introduced a variety of applications involving sampling including
downsampling and oversampling.

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ECE

LECTURE 18: THE SAMPLING THEOREM

This is known as idealized sampling.

EE 3512: Lecture 18, Slide 2

• Recalling the Fourier transform properties of linearity (the transform of a sum

• If our original signal, x(t), is bandlimited: X (e j )  0 for   B

EE 3512: Lecture 18, Slide 3

• The sampling frequency,  s  2 B , is referred to as the Nyquist sampling

EE 3512: Lecture 18, Slide 5

EE 3512: Lecture 18, Slide 7

• A more rigorous equation depends on B1 and B2:

• Sampling can also be thought of as a modulation operation, since it shifts a

EE 3512: Lecture 18, Slide 9

EE 3512: Lecture 18, Slide 10

EE 3512: Lecture 18, Slide 11

EE 3512: Lecture 18, Slide 12

EE 3512: Lecture 18, Slide 13

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