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Sampling Process of A Discrete Time Signal

The document discusses the sampling process for converting analog signals to digital signals. It explains that sampling involves taking snapshots of the analog signal at specific time intervals. If the sampling rate is higher, it is possible to accurately reproduce the original wave shape. However, a higher sampling rate requires more memory. The sampling theorem states that to avoid aliasing, the sampling frequency must be at least twice the highest frequency component of the signal.

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Ghubaida Hassani
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89 views11 pages

Sampling Process of A Discrete Time Signal

The document discusses the sampling process for converting analog signals to digital signals. It explains that sampling involves taking snapshots of the analog signal at specific time intervals. If the sampling rate is higher, it is possible to accurately reproduce the original wave shape. However, a higher sampling rate requires more memory. The sampling theorem states that to avoid aliasing, the sampling frequency must be at least twice the highest frequency component of the signal.

Uploaded by

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

• Many natural phenomena produce analogue


signals including sensor outputs, sound,
video, etc
• In order to transmit/or process analogue
signals digitally the analogue signals must
be converted to digital.
• Signals that are converted to digital involve
sampling and quantization.

1
The Sampling Process
• Sampling a continuous-time signal
implies taking snap shots of the signal
at specific instances of time

• If we take very few samples we will


not be able to obtain the original wave-
shape by interpolation

• If we sample at higher rates it is


possible to reproduce a wave-shape
almost identical to the original wave-
shape

• Sampling at higher rates creates a


much larger demand for memory to
store the samples.

2
The Sampling Process in the
time-domain
• Let x(t) be the analogue signal
• and s(t) be the impulse train

s( t )   (t  nT )
n  
s

• Sampled data is given by


y (t )  x(t )  s (t )

 x(t )    (t  nT )
n  
s


  x(nT ) (t  nT )
n  
s s
3
The Sampling Process in the
frequency-domain
• x(f) is the spectrum of the
analogue signal
• The spectrum of the impulse
train is given by

1
S(f ) 
Ts
 (f 
k  
k
Ts )

• The spectrum of the sampled


signal is given by
 
1 1
Y(f )  X(f )  S(f )  X(f ) 
Ts

k  
(f  k )
Ts
Ts
 X(f 
k  
k
Ts )

4
The Sampling Theorem
• In the figure above the sampling rate fs = 1/Ts is
high and the replica of the original spectrum in
each period do not overlap with each other.

• For this to be true as seen from the figure


fs  f m  f m
• This can be simplified to f s  2 f m
and is referred to as Sampling Theorem
5
Sampling below Nyquist
frequency i.e when f s  2f m
• In the figure the ‘tails’ and the
‘heads’ of the spectral components
of adjacent periods overlap.
• It is impossible to remove the error
introduced by the spectral
components from the adjacent
period in order to recover the
baseband signals
• The inherent error is referred to as
the aliasing error.
• In order to eliminate the aliasing
error the baseband signal must be
properly bandlimited and sampling
done such that
sf 2fm

6
The Sampling Theorem
For any baseband signal that is bandlimited to
a frequency fm, the sampling rate fs must be
selected to be greater or equal to twice the
highest frequency fm in order for the original
baseband signal to be recovered without
distortion using an ideal lowpass filter with a
cut-off frequency fc such that f m  f c  f s  f m

7
Digital Signal Processing of
Continuous-Time Signals
• Conversion of a continuous-time signal into
digital form is carried out by an analog-to-
digital (A/D) converter
• The reverse operation of converting a
digital signal into a continuous-time signal
is performed by a digital-to-analog (D/A)
converter

8
Digital Signal Processing of
Continuous-Time Signals
• Since the A/D conversion takes a finite
amount of time, a sample-and-hold (S/H)
circuit is used to ensure that the analog
signal at the input of the A/D converter
remains constant in amplitude until the
conversion is complete to minimize the
error in its representation

9
Digital Signal Processing of
Continuous-Time Signals
• To prevent aliasing, an analog anti-aliasing
filter is employed before the S/H circuit
• To smooth the output signal of the D/A
converter, which has a staircase-like
waveform, an analog reconstruction filter
is used

10
Digital Signal Processing of
Continuous-Time Signals
A typical digital signal processing system
Anti- Reconstruction
aliasing S/H A/D Processor D/A filter
filter

• Both the anti-aliasing filter and the


reconstruction filter are analog lowpass
filters.

11

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