ANALOG-TO-DIGITAL

CONVERSION

Eng.R.Hirshan
Lecturer
DETE
FE/SEUSL
 ANALOG-TO-DIGITAL CONVERSION

A digital signal is superior to an analog signal because it is more

robust to noise and can easily be recovered, corrected and amplified.
For this reason, the tendency today is to change an analog signal to
digital data.

This process of converting analog signal to digital signal

is known as:
• Pulse Code Modulation (PCM)

4.2
 PCM
 PCM consists of three steps to digitize an analog
signal:
1. Sampling
2. Quantization
3. Binary encoding
 Before we sample, we have to filter the signal to limit
the maximum frequency of the signal as it affects the
sampling rate.
 Filtering should ensure that we do not distort the
signal, that is remove high frequency components that
affect the signal shape.

4.3
 Components of PCM encoder

4.4
 Sampling
 Analog signal is sampled every TS secs.
 Ts is referred to as the sampling interval.
 fs = 1/Ts is called the sampling rate or sampling
frequency.
 There are 3 sampling methods:
 Ideal - an impulse at each sampling instant
 Natural - a pulse of short width with varying amplitude
 Flattop - sample and hold, like natural but with single amplitude
value
 The process is referred to as pulse amplitude modulation
PAM and the outcome is a signal with analog (non
integer) values
4.5
 Three different sampling methods for PCM

4.6
 Nyquist Theorem

According to the Nyquist theorem, the

sampling rate must be
at least 2 times the highest frequency
contained in the signal.

4.7
 Nyquist sampling rate for low-pass and band-pass signals

4.8
 Example 1
For an intuitive example of the Nyquist theorem, let us sample
a simple sine wave at three sampling rates: fs = 4f (2 times the
Nyquist rate), fs = 2f (Nyquist rate), and
fs = f (one-half the Nyquist rate). Figure 4.24 shows the
sampling and the subsequent recovery of the signal.

It can be seen that sampling at the Nyquist rate can create a

good approximation of the original sine wave (part a).
Oversampling in part b can also create the same
approximation, but it is redundant and unnecessary. Sampling
below the Nyquist rate (part c) does not produce a signal that
looks like the original sine wave.
4.9
 Recovery of a sampled sine wave for different sampling rates

4.10
 Aliasing

Sampling below the Nyquist rate (part c) does not produce a

signal that looks like the original wave.

This effect is known as Aliasing.

Avoid Aliasing by :

•Using Anti- aliasing filter

•Sampling at a rate higher than Nyquist rate.

4.11
 Example 2

Telephone companies digitize voice by assuming a maximum

frequency of 4000 Hz. The sampling rate therefore is 8000
samples per second.

4.12
 Example 3

A complex low-pass signal has a bandwidth of 200 kHz. What

is the minimum sampling rate for this signal?

Solution
The bandwidth of a low-pass signal is between 0 and f, where
f is the maximum frequency in the signal. Therefore, we can
sample this signal at 2 times the highest frequency (200
kHz). The sampling rate is therefore 400,000 samples per
second.

4.13
 Example 4

A complex band-pass signal has a bandwidth of 200 kHz.

What is the minimum sampling rate for this signal?

Solution
We cannot find the minimum sampling rate in this case
because we do not know where the bandwidth starts or ends.
We do not know the maximum frequency in the signal.

4.14
 Quantization
 Sampling results in a series of pulses of varying
amplitude values ranging between two limits: a min and
a max.
 The amplitude values are infinite between the two limits.
 We need to map the infinite amplitude values onto a
finite set of known values.
 This is achieved by dividing the distance between min
and max into L zones, each of height 
 = (max - min)/L

4.15
 Quantization

4.16
 Quantization Levels

 The midpoint of each zone is assigned a value from 0 to L-1

(resulting in L values)
 Each sample falling in a zone is then approximated to the
value of the midpoint.

4.17
 Quantization Zones
 Assume we have a voltage signal with amplitutes Vmin=-20V
and Vmax=+20V.
 We want to use L=8 quantization levels.
 Zone width = (20 - -20)/8 = 5
 The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to 0, 0
to +5, +5 to +10, +10 to +15, +15 to +20
 The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5,
17.5

4.18
 Assigning Codes to Zones
 Each zone is then assigned a binary code.
 The number of bits required to encode the zones, or the
number of bits per sample as it is commonly referred to,
is obtained as follows:
nb = log2 L
 Given our example, nb = 3
 The 8 zone (or level) codes are therefore: 000, 001, 010,
011, 100, 101, 110, and 111
 Assigning codes to zones:
 000 will refer to zone -20 to -15
 001 to zone -15 to -10, etc.

4.19
 Quantization and encoding of a sampled signal

4.20
 Quantization Error
 When a signal is quantized, we introduce an error - the
coded signal is an approximation of the actual amplitude
value.
 The difference between actual and coded value
(midpoint) is referred to as the quantization error.
 The more zones, the smaller  which results in smaller
errors.
 BUT, the more zones the more bits required to encode
the samples -> higher bit rate

4.21
 Quantization Error and SNQR
 Signals with lower amplitude values will suffer more
from quantization error as the error range: /2, is fixed
for all signal levels.
 Non linear quantization is used to alleviate this problem.
Goal is to keep SNQR fixed for all sample values.
 Two approaches:
 The quantization levels follow a logarithmic curve. Smaller ’s
at lower amplitudes and larger’s at higher amplitudes.
 Companding: The sample values are compressed at the sender
into logarithmic zones, and then expanded at the receiver. The
zones are fixed in height.

4.22
 Bit rate and bandwidth requirements of
PCM
 The bit rate of a PCM signal can be calculated form the number of
bits per sample x the sampling rate
Bit rate = nb x fs
 The bandwidth required to transmit this signal depends on the
type of line encoding used. Refer to previous section for
discussion and formulas.
 A digitized signal will always need more bandwidth than the
original analog signal. Price we pay for robustness and other
features of digital transmission.

4.23
 Example 5

We want to digitize the human voice.What is the bit rate,

assuming 8 bits per sample?

Solution
The human voice normally contains frequencies from 0 to
4000 Hz. So the sampling rate and bit rate are calculated as
follows:

4.24