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Sampleing

The document discusses the sampling theorem, which states that bandlimited continuous-time signals can be represented as discrete-time signals with no loss of information if sampled above the Nyquist rate. It covers representing signals using impulse functions, the Fourier transform of sampled signals, reconstructing signals from samples using an interpolation filter, and issues like aliasing that can occur from undersampling. Applications of sampling discussed include digital signal processing techniques like downsampling, upsampling, and oversampling.

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anthony.onyishi.242680
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13 views

Sampleing

The document discusses the sampling theorem, which states that bandlimited continuous-time signals can be represented as discrete-time signals with no loss of information if sampled above the Nyquist rate. It covers representing signals using impulse functions, the Fourier transform of sampled signals, reconstructing signals from samples using an interpolation filter, and issues like aliasing that can occur from undersampling. Applications of sampling discussed include digital signal processing techniques like downsampling, upsampling, and oversampling.

Uploaded by

anthony.onyishi.242680
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
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SAMPLING THEOREM

• Objectives:
Representation Using Impulses
FT of a Sampled Signal
Signal Reconstruction
Signal Interpolation
Aliasing
Multirate Signal Processin
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.
x(t)
• Recall our expression for a pulse train:

p(t )   (t  nT ) …
n   t
-2T -T 0 T 2T
• A sampled version of a CT signal, x(t), is:
 
xs (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 1
ck   p(t )e  jk0t dt    (t )e  jk0t dt  e  jk0t t 0 
T T / 2 T T / 2 T T

1 jk0t
p(t )  
k   T
e
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 
    1 
 
X s e  Fp(t ) x(t )  F   x(t )e jk0t    F x(t )e jk0t
j

k   T  T k  
1 
  X (e j (  k0 ) )
T k 

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

 s  2B
• The sampling frequency, , 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 s ( )h(t   )d

 
   
    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 , where B = B2 – B1.

• A more rigorous equation depends on B1 and B2:

2B  f s  4B

r f  B/2
f s  2B where r   c
r B
• and can also be thought of as a modulation operation, since it shifts a signal’s spectrum in frequency.
Sampling

f c  ( B1  B2 ) / 2
r  r  (greatest integer greater than or equal to r )
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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