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= 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
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 ) ( ) (
Let
) (t x
o
denote the result of the sampling
original signal by impulses at
S
nT time instants.
Sampling Theorem:
Then:
=
=
n
S S
nT t nT x t x ) ( ) ( ) ( o
o
We can write
) (t x
o
as:
=
=
n
S
nT t t x t x ) ( ) ( ) ( o
o
=
=
n
S S
nT t nT x t x ) ( ) ( ) ( o
o
=
=
n
S
nT t t x t x ) ( ) ( ) ( o
o
Sampling Theorem:
f
W -W -fc fc 0
. . . .
. . . .
W f
c
o c
x f
) ( 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 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
=
| |
=
|
\ .
=
|
|
.
|
\
|
- =
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:
Now if
W
T
S
2
1
> then the replicated spectrum of
) (t x
overlaps, and reconstruction of the original
signal is not possible. This type of distortion
that 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.
Of course, one obvious (though not practical)
choice is an ideal low pass filter with bandwidth
W
'
W
' W
T
W W
S
<
'
s
1
where
satisfies , i.e.
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
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
=
|
|
.
|
\
|
=
=
n
s S
nT t nT x t x o
o
) (
Pulse Amplitude Modulation:
(a)
Input
PAM Output
) (t h
(b)
) (t h
0
t
t
Slope= -
(d)
f
) ( f H Z
t
t
2/ 1/
-1/ -2/
f
0
(c)
) ( f H
2/ 1/
-1/ -2/
f
0
(c)
) ( f H
Figure (3): Generation of PAM.
(a) Holding network.
(b) Impulse response of holding network.
(c) Amplitude response of holding network.
(d) Phase response of holding network.
Pulse Amplitude Modulation:
From figure (2) a PAM signal can be written as:
( )
=
(
(
(
(
|
.
|
\
|
+
[ =
n
S
S PAM
nT t
nT x t x
t
t
2
1
) (
The waveform is generated by placing the
impulse function in (11) on the output of a
holding network having the impulse response.
Pulse Amplitude Modulation:
(
(
(
(
|
.
|
\
|
+
[ =
t
t
2
1
) (
S
nT t
t h
And the transfer function is:
( )
t t
t t
f j
e f c f H
= sin ) (
Pulse Amplitude Modulation:
Since the holding network dose not have a
constant amplitude response over the bandwidth
of
) (t x
t
, unless of course the pulse width
is sufficiently narrow, amplitude distortion
results. This amplitude distortion can be
removed by passing the samples, prior to
reconstruction of
) (t x
Pulse Amplitude Modulation:
through a filter having an amplitude response
equal to
) ( 1 f H
, over the bandwidth of ). (t x
.
Since the phase response of the holding
network is linear, the effect is a time delay and
can usually be neglected.
Pulse Width Modulation:
A (PWM) waveform, as illustrated in figure (2),
consists the sequence of pulse width each pulse
having a width proportional to the values of the
a message signal at the sampling instants.
If the message is (0) at the sampling time, the
width of the (PWM) pulse is
.
2
1
S
T
Pulse Width Modulation:
Thus, pulse widths less than
S
T
2
1
correspond
to negative sample values and the pulse widths
greater than correspond to positive sample
S
T
2
1
values.
PWM is seldom used in modern communications
systems.
Pulse Width Modulation:
PWM is used extensively for DC motor
control in which motor speed is proportional
to the width of the pulses.
Since thee pulses have equal amplitude, the
energy in a given pulse is proportional to the
pulse width. Thus, the sample values can be
recovered from a PWM waveform by low pass
filtering.
Pulse Position Modulation:
A (PPM) signal consists of a sequence of
pulses in which the pulse displacement from
a specified time reference is proportional to
the sample values of the information-bearing
signal.
A (PPM) signal is illustrated in figure (2),
and can be represented by the expression:
Pulse Position Modulation:
=
=
n
n
t t g t x ) ( ) (
Where
) (t g
represents the shape of the
individual pulses, and occurrence times
n
t
are related to the values of the message signal
) (t x
S
nT
at the sampling instants
, as discussed
previously.
Pulse Position Modulation:
The spectrum of a PPM signal is very similar to
the spectrum of a PWM signal.
If the time axis is slotted so that a given range
of sample values is associated with each slot,
the pulse positions are quantized and pulse is
assigned to given slot depending on the sample
value.
Pulse Position Modulation:
Slots are non-overlapping and are therefore
orthogonal.
If a given sample value is assigned to one
of (M) slot, the result is (M-ary) orthogonal
communications. PPM is finding new
applications in area of ultra-wideband
communications.