Professional Documents
Culture Documents
=
=
1 2
( ) ,
s
jn t
s s
n n
s s
t nT e
T T
= =
= =
Ideal Sampling ( or Impulse Sampling)
Therefore, we have:
Take Fourier Transform (frequency convolution)
1
( ) ( ) e
s
jn t
s
n
s
x t x t
T
=
| |
=
|
\ .
{ }
1 1
( ) ( ) * ( ) *
s s
jn t jn t
s
n n
s s
X f X f e X f e
T T
= =
= =
`
)
1
( ) ( ) * ( ),
2
s
s s s
n
s
X f X f f nf f
T
=
= =
1 1
( ) ( ) ( )
s s
n n
s s s
n
X f X f nf X f
T T T
= =
= =
Ideal Sampling ( or Impulse Sampling)
This shows that the Fourier Transform of the sampled signal is the
Fourier Transform of the original signal at rate of 1/T
s
Ideal Sampling ( or Impulse Sampling)
This means that the output is simply the replication of the original
signal at discrete intervals, e.g
Ideal Sampling ( or Impulse Sampling)
As long as f
s
> 2f
m
,no overlap of repeated replicas X(f - n/T
s
) will
occur in X
s
(f)
Minimum Sampling Condition:
Sampling Theorem: A finite energy function x(t) can be completely
reconstructed from its sampled value x(nTs) with
provided that =>
2
s m m s m
f f f f f > >
2 ( )
sin
2
( ) ( )
( )
s
s
s s
n
s
f t nT
T
x t T x nT
t nT
=
(
(
=
`
( ) sin (2 ( ))
s s s s
n
T x nT c f t nT
=
=
1 1
2
s
s m
T
f f
=
T
s
is called the Nyquist interval: It is the longest time interval that can
be used for sampling a bandlimited signal and still allow
reconstruction of the signal at the receiver without distortion
Practical Sampling
In practice we cannot perform ideal sampling
It is not practically possible to create a train of
impulses
Thus a non-ideal approach to sampling must be used
We can approximate a train of impulses using a train of very thin
rectangular pulses:
Note:
Fourier Transform of impulse train is another impulse train
Convolution with an impulse train is a shifting operation
( )
s
p
n
t nT
x t
=
| |
=
|
\ .
Natural Sampling
If we multiply x(t) by a train
of rectangular pulses x
p
(t),
we obtain a gated waveform
that approximates the ideal
sampled waveform, known
as natural sampling or
gating (see Figure 2.8)
( ) ( ) ( )
s p
x t x t x t =
2
( )
s
j nf t
n
n
x t c e
=
=
( ) [ ( ) ( )]
s p
X f x t x t =
2
[ ( ) ]
s
j nf t
n
n
c x t e
=
=
[ ]
n s
n
c X f nf
=
=
Each pulse in x
p
(t) has width T
s
and amplitude 1/T
s
The top of each pulse follows the variation of the signal being
sampled
X
s
(f) is the replication of X(f) periodically every f
s
Hz
X
s
(f) is weighted by C
n
Fourier Series Coeffiecient
The problem with a natural sampled waveform is that the tops of the
sample pulses are not flat
It is not compatible with a digital system since the amplitude of each
sample has infinite number of possible values
Another technique known as flat top sampling is used to alleviate
this problem
Flat-Top Sampling
Here, the pulse is held to a constant height for the whole
sample period
Flat top sampling is obtained by the convolution of the signal
obtained after ideal sampling with a unity amplitude
rectangular pulse, p(t)
This technique is used to realize Sample-and-Hold (S/H)
operation
In S/H, input signal is continuously sampled and then the
value is held for as long as it takes to for the A/D to acquire
its value
Flat top sampling (Time Domain)
'( ) ( ) ( ) x t x t t =
( ) '( ) * ( )
s
x t x t p t =
( ) * ( ) ( ) ( ) * ( ) ( )
s
n
p t x t t p t x t t nT
=
(
= =
(
=
(
=
(
1
( ) ( ) * ( )
s
n
s
P f X f f nf
T
=
(
=
(
1
( ) ( )
s
n
s
P f X f nf
T
=
=
=
= =
=
2
( ) ( ) ( ) ( )
s
j nf t
s p n
n
x t x t x t x t c e
=
= =
( ) '( ) * ( ) ( ) ( ) * ( )
s s
n
x t x t p t x t t nT p t
=
(
= =
(
Example 1:
Consider the analog signal x(t) given by
What is the Nyquist rate for this signal?
Example 2:
Consider the analog signal x
a
(t) given by
What is the Nyquist rate for this signal?
What is the discrete time signal obtained after sampling, if
f
s
=5000 samples/s.
What is the analog signal x(t) that can be reconstructed from the
sampled values?
( ) 3cos(50 ) 100sin(300 ) cos(100 ) x t t t t = +
( ) 3cos 2000 5sin 6000 cos12000
a
x t t t t = + +
Practical Sampling Rates
Speech
- Telephone quality speech has a bandwidth of 4 kHz
(actually 300 to 3300Hz)
- Most digital telephone systems are sampled at 8000
samples/sec
Audio:
- The highest frequency the human ear can hear is
approximately 15kHz
- CD quality audio are sampled at rate of 44,000
samples/sec
Video
- The human eye requires samples at a rate of at
least 20 frames/sec to achieve smooth motion
Pulse Code Modulation (PCM)
Pulse Code Modulation refers to a digital baseband signal that is
generated directly from the quantizer output
Sometimes the term PCM is used interchangeably with quantization
See Figure 2.16 (Page 80)
Advantages of PCM:
Relatively inexpensive
Easily multiplexed: PCM waveforms from different
sources can be transmitted over a common digital
channel (TDM)
Easily regenerated: useful for long-distance
communication, e.g. telephone
Better noise performance than analog system
Signals may be stored and time-scaled efficiently (e.g.,
satellite communication)
Efficient codes are readily available
Disadvantage:
Requires wider bandwidth than analog signals
2.5 Sources of Corruption in the sampled,
quantized and transmitted pulses
Sampling and Quantization Effects
Quantization (Granularity) Noise: Results when
quantization levels are not finely spaced apart enough
to accurately approximate input signal resulting in
truncation or rounding error.
Quantizer Saturation or Overload Noise: Results when
input signal is larger in magnitude than highest
quantization level resulting in clipping of the signal.
Timing Jitter: Error caused by a shift in the sampler
position. Can be isolated with stable clock reference.
Channel Effects
Channel Noise
Intersymbol Interference (ISI)
The level of quantization noise is dependent on how close any
particular sample is to one of the L levels in the converter
For a speech input, this quantization error resembles a noise-
like disturbance at the output of a DAC converter
Signal to Quantization Noise Ratio
Uniform Quantization
A quantizer with equal quantization level is a Uniform Quantizer
Each sample is approximated within a quantile interval
Uniform quantizers are optimal when the input distribution is
uniform
i.e. when all values within the range are equally
likely
Most ADCs are implemented using uniform quantizers
Error of a uniform quantizer is bounded by
2 2
q q
e <
The mean-squared value (noise variance) of the quantization error is
given by:
/ 2 / 2 / 2
2 2 2
/ 2 / 2 / 2
1
1
( )
2
q q q
q q q
e p e de e de e de
q
q
| |
= =
|
\ .
=
3
/ 2
/ 2
2
1
3 12
q
q
q
e
q
= =
Signal to Quantization Noise Ratio
The peak power of the analog signal (normalized to 1Ohms )can be
expressed as:
Therefore the Signal to Quatization Noise Ratio is given by:
2
2 2 2
2 4
1
pp
p
V
V L q
P
| |
| |
|
|
|
|
\ .
\ .
= = =
2 2
2
/ 4
/ 12
2
3
q
L q
q
SNR L = =
where L = 2
n
is the number of quantization levels for the converter.
(n is the number of bits).
Since L = 2
n
, SNR = 2
2n
or in decibels
p p
V
L
q =
2
10log (2 ) 6
10
n
S
n dB
N
dB
| |
|
|
\ .
= =
If q is the step size, then the maximum quantization error that can
occur in the sampled output of an A/D converter is q
NEXT TIME
2.7 Uniform and NonUniform Quantization
2.8 Baseband Modulation
2.9 Correlative Coding (Will not be covered)
Chapter 3: Baseband Demodulation/Detection
(start)