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ANALOG-TO-DIGITAL CONVERSION

INTRODUCTION:
Usually most of the signals such as speech signals, RADAR signals and various
communication signals are analog in nature. Because of many advantages, digital signal
processing is preferable to analog signal processing. To process the analog signals by
digital means, it is first necessary to convert them into digital form. That is to convert
them to a sequence of numbers. This procedure is called Analog-to-Digital Conversion
(A/D) and the corresponding devices which perform this are called Analog-to-Digital
Converters (ADCs). The Analog signal is simply a continuous signal which is continuous
both in terms of time and amplitude, i.e, it has a continuous range of values both on time
axis and in amplitude axis, it is simply called as Continuous Time Continuous Value
(CTCV) signal. Whereas on the other hand Digital signal is discretized both in time and
amplitude and the corresponding amplitudes are represented in a sequence of binary
digits. In order to do this Analog to Digital Conversion it is necessary to discretize both
the time and amplitude axis and this process is done by first doing on the time axis and
then moving to the amplitude axis. So this entire process is done in three stages namely
Sampling, Quantization and Encoding and block diagram of the A/D process and
corresponding waveforms of the continuous , discrete and digital signals are shown in
figure1.







Figure 1: Block diagram of Analog-to-Digital Conversion
Encoder Quantizer Sampler
Continuous
time signal x
a
(t)
Discrete time
signal x(nT)
Quantized
signal x
q
(n)
digital data

011001100
Sampling:
The first step of the A/D is sampling process, discretizing the time axis i.e, it
converts the analog/CTCV signal into Discrete time Continuous Value (DTCV) / discrete
signal. This conversion is done by taking the value of the signal at specified instances of
time instead of taking the signal for all values of time. And these values are taken
precisely at a periodic instances with a period T, called as a sampling Period. So the
sampler converts the analog x(t) signal into a discrete signal x(nT) which can also be
denoted as x[n]
i.e, x[n] = x(nT) .(1)
The reciprocal of this sampling interval is the sampling frequency f
s
. Selection of this
sampling frequency plays a crucial role in sampling. If this sampling frequency is not
properly selected, then we cannot reconstruct the original signal from the sampled one.
The selection is conveyed by the sampling theorem.

Sampling Theorem says that, if we have a continuous time signal x(t) and its equally
spaced samples x(nT), sampled at a sampling period T and
- If x(t) is band limited i.e
X(w) = 0 for |w| >w
m
(2)
where X(w) is Fourier Transform of x(t) and w
m
is the highest frequency in the
spectrum of X(w).

- Then under the condition that the sampling frequency (w
s
)
w
s
>
H
~
T
2
2w
m
.(3)
Then x(t) is uniquely recoverable from the set of samples.

The Sampling process is performed by multiplying the continuous time signal x
a
(t) with
impulse train. The impulse train is a periodic impulse signal with a period namely
sampling period T.















The impulse train p(t) can be given as

=
=
n
nT t t p ) ( ) ( o
where, ) (t o is the unit impulse function, or Dirac delta function. The time domain
representation of the continuous time, and discrete time signals are shown in figure 2.

The frequency domain equations of the above multiplication process can be given as
convolution of the Fourier transforms of continuous time signal X
a
(w) and Fourier
Transform of the Impulse signal P(w).
X
s
(w) = | | ) ( ) (
2
1
w P w Xa
[

Where X
s
(w) is the fourier transform of the sampled signal. And the P(w) can be given as

=
|
.
|

\
| [

[
=
k
T
k w
T
w P
2 2
) ( o


substituting the value of P(w) in X
s
(w), we get

X
s
(w) =
(

|
.
|

\
| [

[

= k
T
k w w Xa
T
2
) (
2
2
1
o

X
s
(w) =

=
(

|
.
|

\
| [

k
T
k w w Xa
T
2
) (
1
o

x
a
(t)
Continuous-
time signal
p(t) = Impulse Train
x
s
(nT)
Discrete-
time signal

Figure 2: Time domain representation of continuous, impulse train and discrete signals

But from the product property of the impulse signal, the equation can be re-written as

X
s
(w) =

=
(

[
k
T
k Xa
T
)
2
(
1

That is, the spectrum of the sample signal is a sum of frequency shifted replications of
the Fourier transform of the original signal. If we consider the original continuous time
signal as continuous band of frequencies then the frequency domain representation of the
continuous time, impulse train and discrete-time signals are shown in figure3.

Figure 3: spectrum of continuous, impulse train and sampled signals.

If the sampling interval is not properly selected, that is if the sampling frequency is not
properly chosen as suggested by the sampling theorem, then the time shifted replicas of

Figure 4: spectrum of original and sampled signal with improper selection of f
s
.

the original signal may overlap upon each other, thereby causing loss of information and
the original signal cannot be recovered back. The results of improper selection of the
sampling frequency is shown in figure4.
The above said effect is called as Aliasing effect, where a signal of some
frequency appears to be a signal of different frequency. To recover the original signal
from the sampled signal, it is sufficient to pass the sampled signal through the low pass
filter, then we can obtain the original signal provided that the replicas of the original
signal are not overlapped. The overall process of sampling and reconstruction of original
signal from the sampled signal can be given as shown in the figure5. where H(w) is the
impulse response of the ideal low pass filter(LPF).

Figure 5: Block diagram of sampling and reconstruction using Ideal LPF.

Figure 6: spectrum of sampled, filter and reconstructed signal
The spectrum of reconstructed signal along the filter response is given in Figure6. The
necessary and sufficient condition to recover the original signal without any loss of signal
is
w
s
>> 2w
m

and this condition is called Nyquist sampling theorem. If this condition is not satisfied
then the aliasing effect will occur.

Quantization:
Quantization is the process of converting a Discrete time-continuous valued
(DTCV) signal to Discrete time-Discrete valued signal (DTDV) by expressing each
sample value as a finite number of digits (instead of infinite). The difference between the
unquantized and quantized signals is, in the prior one there are continuous range of the
amplitude levels where as in the next signal, there are only a finite set of amplitude
levels. Quantization is nothing but rounding off the sampled signal to the nearest
quantized levels. This rounding of the sampled data leads to loss of data and the error
introduced in representing the continuous-valued signal by a finite set of discrete value

Figure 7: Discrete sinusoidal signal
levels is called quantization error or quantization noise. If x(n) is the continuous valued
signal and x
q
(n) is quantized signal, then the quantization error e
q
(n) is defined as
e
q
(n) = x
q
(n) x(n)
The quantization process can be explained with the help of a discrete sine wave which is
shown in figure 7 and the corresponding quantization of each sample is shown in the
table1, this table shows the quantized value of each sample and also displays the
corresponding quantization error. The values allowed in the digital signal are called
quantization levels, and the distance between two successive quantization levels is called
the quantization step size () or resolution. The quantization error e
q
(n) in rounding is
limited to the range of -/2 and /2, that is
s
A

2
e
q
(n)
2
A

In other words, the instantaneous quantization error cannot exceed half of the
quantization step. If x
min
and x
max
represents the minimum and maximum value of x(n)
and L is the number of quantization levels, then resolution can be defined as
1
Xmax - Xmin

= A
L

In the example specified in table 1, the x
min
= -4 and x
max
= 4, and the number of levels
Samples
Number
Discrete signal
value x(n)
Quantized value
x
q
(n)
Quantization
error e
q
(n)
1,11,21 0 0 0
2,10 1.25 1 -0.25
3,9 2.3 2 -0.3
4,8 3.1 3 -0.1
5,7 3.8 4 0.2
6 4 4 0
12,20 -1.25 -1 0.25
13,19 -2.3 -2 0.3
14,18 -3.1 -3 0.1
15,17 -3.8 -4 -0.2
16 -4 -4 0

Table 1: Quantization of the discrete signal
is L = 9, which leads to = 1. and from the table it is abvious that the quantization error
never exceeded 0.5, which is half of the resolution. As we increase the number of
quantization levels L then the resolution decreases, so that the dynamic range of the
quantization error also decreases which increases the accuracy of the quantization. So in
practice we can reduce the quantization error to an insignificant value by choosing a
sufficient number of quantization levels. The quantized signal x
q
(n) is shown in figure 8.

Figure 8: Quantized signal

Encoder:

The purpose of the encoder is to represent the value of every quantized sample in
the form of b-bit binary code. where b is the number of bits present in the binary
equivalent. Simply, Encoding is the process of assigning a binary sequence to every
sampled value of the quantized signal. There are a variety of ways to do the encoding
process and they are called as coding techniques. All these coding techniques aim at
reducing the redundancy of the data. Where redundancy is nothing but the repetition of
data which increases the length of code and ultimately increases the memory required. So
if we remove the redundant data then we can represent the same data with less number of
bits. Some of the coding methods are Huffman coding, Shannon-Fano coding, Arithmetic
coding etc., Table 2 shows the simple coding technique of simply assigning a binary
equivalent without reducing the redundancy.
Samples
Number
Quantized value Binary equivalent
1,11,21 0 0000
2,10 1 0001
3,9 2 0010
4,8 3 0011
5,7,6 4 0100
12,20 -1 0101
13,19 -2 0110
14,18 -3 0111
15,17 -4 1000
16 -4 1001

Table2: Encoding of the quantized samples
After the Encoding is done, the entire analog signal is now can be represented as a
sequence of binary bits, which is nothing but the signal in digital form.

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