Sampling Process

:
The message signal is usually analog
in nature, as in a speech signal or
video signal
It has to be converted into digital
form before it can be transmitted by
digital means.
Sampling Process:
The sampling processing is the first
process performed in analog-to-digital
conversion.
In the sampling process, a continuous-time
signal is converted into a discrete-time
signal by measuring the signal at periodic
instants of time.

Sampling Process:
For the sampling process to be of
practical utility, it is necessary that we
choose the sampling rate properly
So the discrete-time resulting from
the process uniquely defines the
original continuous-time signal.
Sampling Theorem:
Let the signal
) (t x
be band limited with
bandwidth
W
i.e., let
0 ) ( ÷ f X . W f >
for
Let
) (t x be sampled at multiples of some
basic sampling interval
S
T , where
W
T
S
2
1
s
to yield the sequence
( ) | |
·
÷· = n
S
nT x
Then it is
possible to reconstruct the original
Sampling Theorem:
signal ) (t x from the sampled values by the
reconstruction formula:
( ) ( )] 2 [ sin 2 ) (
S
n
S S
nT t W c nT x T W t x ÷
' '
=
¿
·
÷· =
W
'
Where ( ) is any arbitrary number that
satisfies .
1
W
T
W W
S
÷ s
'
s
Sampling Theorem:
special case where
W
T
S
2
1
=
the
reconstruction relation simplifies to:
¿ ¿
·
÷· =
·
÷· =
(
¸
(

¸

|
.
|

\
|
÷
|
.
|

\
|
=
|
|
.
|

\
|
÷ =
n
S
n
S
W
n
t W c
W
n
x n
T
t
c nT x t x
2
2 sin
2
sin ) ( ) (
Prove of Sampling Theorem:
Then:
We can write
) (t x
o
as:
( ) ( ) ( )
S S
n
x t x nT t nT
o
o
·
=÷·
= ÷
¿
¿
·
÷· =
÷ =
n
S
nT t t x t x ) ( ) ( ) ( o
o
Let
) (t x
o
denote the result of the sampling
original signal by impulses at
S
nT
time instants.
Sampling Theorem:
Now if we find the Fourier transform of
both sides of the above relation and apply
the dual of the convolution theorem to the
right-hand side, we obtain:
( ) ( ) ( ) .....(4)
S
n
X f X f F t nT
o
o
·
=÷·
(
= - ÷
(
¸ ¸
¿

Sampling Theorem:
By using Fourier Transform we obtain:
(
¸
(

¸

÷
¿
·
÷· = n
S
nT t F ) ( o
1
....(5)
n
S S
n
f
T T
o
·
=÷·
| |
= ÷
|
\ .
¿
By substituting equation (5) into equation (4),
we obtain:
¿
·
÷· =
|
|
.
|

\
|
÷ - =
n
S S
T
n
f
T
f X f X o
o
1
) ( ) (
¿
·
÷· =
|
|
.
|

\
|
÷ =
n
S S
T
n
f X
T
1
Sampling Theorem:
•Where in the last step we have employed the
convolution property of the impulse signal.
•This relation shows that ) ( f X
o
, the Fourier
transform of the impulse-sampled signal is a
replication of the Fourier transform of the
original signal at a
S
T
1
rate.
Figure (1) shows this situation.
Sampling Theorem:
f
W -W -fs fs 0
. . . .
. . . .
s
f W ÷
s o
f x
) ( f x
o
f
-W W
0
o
x
) ( f x
Figure (1):
Signal spectra for low pass sampling.
(a) Assumed spectrum for x(t).
(b) Spectrum of sampled signal.
Sampling Theorem:
•Now if
W
T
S
2
1
> then the replicated spectrum of
) (t x

overlaps, and reconstruction of the original
is not possible.
•This type of distortion results from under-
sampling is known as:
aliasing error or aliasing distortion.
Sampling Theorem:
•However, if
W
T
S
2
1
s no overlap occurs, and by

employing an appropriate filter we can
reconstruct the original signal back.
•To obtain the original signal back, it is
sufficient to filter the sampled signal by a low
pass filter with frequency response
Characteristic:
Sampling Theorem:
S
T f H = ) (
W f <
0 ) ( = f H
W
T
f
S
÷ >
1
1.
for
.
2.
for
For
W
T
f W
S
÷ < s
1
, the filter can have any
characteristics that make its implementation easy.
choice is an ideal low Of course, one obvious
pass filter with bandwidth
W
'
W
'
W
T
W W
S
÷ <
'
s
1
where satisfies:
Sampling Theorem:
|
.
|

\
|
'
H =
W
f
T f H
S
2
) (
With this choice we have:
|
.
|

\
|
'
H =
W
f
T f X f X
S
2
) ( ) (
o
Taking inverse Fourier transform of both sides,
we obtain:
( ) t W c T W t x t x
S
' '
- = 2 sin 2 ) ( ) (
o
i.e.
Sampling Theorem:
( ) ( ) ( ) t W c T W nT t nT x
S
n
S S
' '
- |
.
|

\
|
÷ =
¿
·
÷· =
2 sin 2 o
( ) ( ) | |
¿
·
÷· =
÷
' '
=
n
S S S
nT t W c nT x T W 2 sin 2
This relation shows that if we use sine functions
for interpolation of the sampled values, we can
reconstruct the original signal perfectly.
Sampling Theorem:
•The sampling rate
W
f
S
2
1
=
is the minimum
sampling rate at which no aliasing occurs.
•This sampling rate is known as the Nyquist
sampling rate.
•If sampling is done at the Nyquist rate,
then the only choice for the reconstruction
filter is an ideal low pass filter and .
2
1
S
T
W W = =
'
Sampling Theorem:
Then:
( )
¿
·
÷· =
÷
|
.
|

\
|
=
n
n Wt c
W
n
x t x 2 sin
2
) (
( )
¿
·
÷· =
|
|
.
|

\
|
÷ =
n
S
S
n
T
t
c nT x t x sin ) (
•In practical systems, sampling is done at a rate
higher than the Nyquist rate.
•This allows for the reconstruction filter to be
realizable and easier to build.
Sampling Theorem:
In such cases the distance between two adjacent
replicated spectra in the frequency domain; i.e.
W f W W
T
S
S
2
1
÷ = ÷
|
|
.
|

\
|
÷
, is known as the guard band.
Problem 1
Problem 2
problem3
Problem 4

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