Pemodulatan Digit

•In the early 90’s, telecommunication networks is changing towards
digital world. With the rapid advancement in the fields of VLSI and
microprocessor, several telecommunication components such as
transmission line and switching has been using digital signals in
their operation.

Therefore, information signals must be changed to digital form so
that it can be transmitted through this network.
•Several techniques requiring full coding of the original signal will
be used:
4.0 Introduction
4.0 Introduction
-      
• Pulse Code Modulation (PCM)
• Differential PCM (DPCM)
• Adaptive Differential PCM (ADPCM)
• Delta Modulation (DM)
• Adaptive Delta Modulation (ADM)

Advantages :
◦ Immunity to noise
◦ Easy storage and processing:
◦ Regeneration
◦ Easy to measure
◦ Enables encryption
◦ Data from several sources can be integrated and
transmitted using the same digital communication
system
◦ Error correction detection can be utilized

Disadvantages :
◦ Requires a bigger bandwidth
◦ Analog signal need to be changed to digital first
◦ Not compatible to analog system
◦ Need synchronization
Pemodulatan Digit
Voice : Analog : 4 kHz
Digit : 2 x 4 kHz x 8 bit = 64 kb/s
BW
min
⇔ 32 kHz
MP, DSP, RAM, ROM, Computer
4.2 TRANSMISSION METHOD FOR ANALOG &
DIGITAL SIGNALS
Analog
input
Analog channel
Baseband
Analog
output
Analog
input
Modulator
De
modulator
Analog
output
Analog
channel
Digital
input
encoder
decoder Digital
channel
Digital
output
Digital
input
Modem
Modem
Analog
channel
Digital
output
Analog
input
ADC &
encoder
Decoder
& DAC
Analog
output
Digital
channel
Analog
input
Analog
output
Analog
channel
ADC &
encoder
Modem
ADC &
decoder
Modem
4.3 Pulse Modulation
4.3 Pulse Modulation
• PAM (Pulse Amplitude Modulation) => V
PAM
α V
m

• PWM (Pulse Width Modulation) => τ α V
m
• PPM (Pulse Position Modulation) => τ
d
(pulse delay) α V
m
• PCM (Pulse Code Modulation)
Pulse Modulation consists of:
Easily effected by
noise
Less susceptible to
noise
Less susceptible to
noise compared to
PAM
m s
f f 2 ≥
Pemodulatan Digit
X
Digital signal
s(t)
m
s
(t) m(t)
m(t)
t
m
s
(t)
t
s(t)
t
T
s
Nyquist theorem
states that:
( ) [ ]
s
s
s
s s s
s
f
T
t t t
T
t s
π
π
ω
ω ω ω
2
2
..... 3 cos 2 2 cos 2 cos 2 1
1
· ·
+ + + + ·
where
s
s
f
T
1
·
( ) ( ) ( )
( ) ( ) ( ) ( ) [ ] ..... 3 cos 2 2 cos 2 cos 2
1
+ + + + ·
·
t t m t t m t t m t m
T
t s t m t m
s s s
s
s
ω ω ω
Fourier series for impulse train :
Therefore :
Pemodulatan Digit
s
ω −
s
ω
m s
ω ω +
m
ω −
m
ω 0
m s
ω ω −
m s
ω ω + −
m s
ω ω − −
) (ω
s
M
s
T
1
ω
m
ω −
m
ω 0
ω
) (ω M
1
s
ω −
s
ω 0
) (ω S
s
T
π 2
ω
0
t
) (t m
s
T 6 −
s
T 6 0
) (t s
t
s
T
s
T 6 −
s
T 6 0
) (t m
s
t
s
T
Time domain
Frequency domain
Pemodulatan Digit
m
ω −
m
ω 0
ω
) (ω
r
M
1
m
ω −
m
ω 0
ω
) (ω M
1
X
Pulse signal
s(t)
m
s
(t)
m(t) h(t)
m
r
(t)
TX
RX
Low pass filter
s
ω −
s
ω
m s
ω ω +
m
ω −
m
ω 0
m s
ω ω −
m s
ω ω + −
m s
ω ω − −
) (ω
s
M
s
T
1
ω
n
ω −
n
ω 0
) (ω H
s
T
ω

Sampling process shown previously uses an ideal pulse
signal

However, it is quite difficult to generate an ideal pulse
signal practically

The usual pulse signal generated is as shown below:
1
2 2
( ) kos
di mana
sin
n
n
s s s
s
n
s
A A nt
s t c
T T T
n
T
c
n
T
τ τ π
πτ
πτ

·
· +
·

Pemodulatan Digit
t
s(t)
T
s
τ
A
τ - pulse width
T
s
– pulse period
s
T

sinc ·
Pemodulatan Digit
Natural Sampling Flat-top Sampling
Information signal
Pulse signal
Sampled signal (PAM)
t
m(t)
t
s(t)
T
s
τ
t
m
s
(t)
T
s
τ
t
m
s
(t)
T
s
τ
( )
s
T << τ

In every sampling methods, the pulse amplitude is directly
proportional to the amplitude of the information signal

Practically, an ideal sampling is difficult to generate

However, by using an ideal and natural sampling, noise
can be eliminated, which is not the case for flat-top
sampling
Pemodulatan Digit
Ideal Sampling Flat-top Sampling
m
s
(t)
t
Natural Sampling
Pemodulatan Digit
X
Pulse signal
s(t)
m
s
(t) m(t)
m(t)
t
m
s
(t)
t
s(t)
t
Mathematical analysis:
s
n
s s s
T
nt
T
n
T T
t s
π τ τ τ 2
cos sinc
2
) (
1


·
+ ·
Fourier series for pulse signal, s(t) :
) ( ) ( ) ( t s t m t m
s
·
Therefore, the sampled signal:

,
`

.
|
+ ·


·1
2
cos
2
). ( ) (
n
s
n
s s
s
T
nt
c
T T
t m t m
π τ τ


·
+ ·
1
2
cos
2 ) ( ) (
) (
n
s
n
s s
s
T
nt
c
T
t m
T
t m
t m
π τ τ
....
6
cos
2 ) (
4
cos
2 ) ( 2
cos
2 ) ( ) (
) (
3
2 1
+
+ + + ·
s s
s s s s s
s
T
t
c
T
t m
T
t
c
T
t m
T
t
c
T
t m
T
t m
t m
π τ
π τ π τ τ
For n = 1, 2 , 3 …..
The above expression shows that the frequency components of the
sampled signal is at f
s
, 2f
s
and 3f
s
. Components 2f
s
and 3f
s
is a replica of
the spectrum of the sampled signal.
....
6
cos
2 ) (
4
cos
2 ) ( 2
cos
2 ) ( ) (
) (
3
2 1
+
+ + + ·
s s
s s s s s
s
T
t
c
T
t m
T
t
c
T
t m
T
t
c
T
t m
T
t m
t m
π τ
π τ π τ τ
f
3f 2
s
2f
s
3f
s
f
s
0
f
s
-f
m
f
s
+f
m 2f
s
+f
m
3f
s
+f
m
3f
s
-f
m
2f
s
-f
m
m
s
(f)
Spectrum of the sampled
signal
The spectrum of the sampled signal has sidebands f
s
t f
m
, 2f
s
t f
m
, 3f
s

t f
m
and so on.
The choice of sampling frequency, f
s
must follow the sampling theorem
to overcome the problem of aliasing and loss of information
(a) Sampling frequency=>  f
s1
< 2f
m(max)
f
2f
s1
3f
s1
f
s1
f
m
Aliasing
m
s
(f)
(b) Sampling frequency=>  f
s2
> 2f
m(max)
f
2f
s2
3f
s2
f
s2
f
m
m
s
(f)
Shannon sampling
theorem=> f
s
≥ 2f
m
Nyquist frequency
⇒f
s
= 2f
m
= f
N
A bandlimited signal that
has a maximum
frequency, f
max
can be
regenerated from the
sampled signal if it is
sampled at a rate of at
least 2f
max.
4.4 Detection of Sampled Signal
4.4 Detection of Sampled Signal
By using LPF to the sampled signal, m
s
(t)
LPF m
s
(t) m(t)
Cut-off frequency ,  f
o
 for LPF must be within the range:  f
m
≤ f
o
≤ f
s
- f
m
• Eventhough the sampled signal can be detected easily at f
s
= 2f
m,
but usually
f
s
> 2f
m
. The main reason is to have a ‘guardband’ .
• Therefore, the maximum frequency that can be processed by the sampled 
data using sampling frequency,  f
s
(without aliasing) is:
=> f
m
= f
s
/ 2 = 1 / 2T
s


·
+ ·
1
2
cos
2 ) ( ) (
) (
n
s
n
s s
s
T
nt
c
T
t m
T
t m
t m
π τ τ
From the sampling process, the sampled signal:
s
n
T
n
c
τ
sinc · where :
s
T << τ
If:
1 sinc ≈
s
T

therefore


·
+ ·
1
2
cos
2 ) ( ) (
) (
n
s s s
s
T
nt
T
t m
T
t m
t m
π τ τ
Therefore
( ) ( ) t t m
m
ω cos 1+ ·
Taking:
Mathematical analysis:
( ) ( ) ( )
( ) ( )
( )
( )
( ) ( )
( ) ( )
]
]
]

+ − + + + +
− + + + + +
·

,
`

.
|
+ + +
+ + +
·
]
]
]

+ + + + + ·
]
]
]

+ + ·
+ + + ·
+ + + ·




·

·

·
......... 2 cos 2 cos 2 cos 2
cos cos cos 2 cos 1
... cos 2 cos 2 2 cos 2
cos cos 2 cos 2 cos 1
.... 3 cos 2 2 cos 2 cos 2 1 cos 1
cos 2 1 cos 1
cos 2 cos 1 cos 1
cos
2
cos 1 cos 1
1
1
1
t t t
t t t t
T
t t t
t t t t
T
t t t
T
t
t n
T
t
t n
T
t
T
t
t n
T
t
T
t t m
m s m s s
m s m s s m
s
m s s
m s s m
s
s s s
s
m
n
s
s
m
n
s
s
m
s
m
n
s
s
m
s
m s
ω ω ω ω ω
ω ω ω ω ω ω
τ
ω ω ω
ω ω ω ω
τ
ω ω ω
τ
ω
ω
τ
ω
ω
τ
ω
τ
ω
ω
τ
ω
τ
ω
replacing ( ) ( ) t t m
m
ω cos 1+ ·


·
+ ·
1
2
cos
2 ) ( ) (
) (
n
s s s
s
T
nt
T
t m
T
t m
t m
π τ τ
inside
It can be shown that the output sampled signal is the same as the output
PAM signal when :
s
T << τ
that is, the pulse width τ is much smaller compared to the pulse
period T
s
.
Voltage
translator
v
m
(t)
v
d
(t)
v
PAM
(t)
LPF
v
m
(t)
v
PAM
(t)
(a) PAM generation
(b) PAM detection
=> m
s
(t) = V
PAM
4.5 Pulse Width Demodulation (PWD)
4.5 Pulse Width Demodulation (PWD)
) (t v
m
∝ τ

τ  (pulse width) follows the instantaneous value of the information
signal v
m
(t) :
t
m o o
ω τ τ τ cos + ·
( ) t
m o
ω τ τ cos 1+ ·
τ
o
 represents the width that is 
fixed according to the minimum 
value of the information signal
The equation shows that the pulse 
width, τ  of the output signal 
PWM varies according to the 
instantaneous value of the 
information signal.
( )
{ ¦ ... 2 cos 2 cos 2 1 + + + ·
·
t t
T
v
s s
s
PWM
PWM
ω ω
τ
τ τ
Replacing τ   inside the general equation of the sampled signal:
( )
( )
{ ¦ ... 2 cos 2 cos 2 1
cos 1
+ + +
+
·
·
t t
T
t
v
s s
s
m o
PWM
PWM
ω ω
ω τ
τ τ
¹
'
¹
¹
'
¹
+ +
+ + + +
·
... cos 2 cos 2
cos cos 2 cos 2 cos 2 cos 2 1
t t
t t t t t
T
v
m s
m s m s s
s
o
PWM
ω ω
ω ω ω ω ω
τ
¹
'
¹
¹
'
¹
+ + − + + +
− + + + +
·
... ) 2 cos( ) 2 cos( ) cos(
) cos( cos 2 cos 2 cos 2 1
t t t
t t t t
T
v
m s m s m s
m s m s s
s
o
PWM
ω ω ω ω ω ω
ω ω ω ω ω
τ
Generation of PWM signal is by changing the value of sample signal of the
PAM signal into a specific period
v
PWM
(t)
v
PAM
(t)
555 timer
(a) PWM generation using voltage to time converter
LPF v
PWM
(t)
v
m
(t)
(b) PWM detection using LPF
f
s
> 2f
m

f
s
= 2f
m

f
s
< 2f
m

Sampling Quantization Coding
A method used to represent an analog signal in terms of digital word
Constitutes 3 process:
1. Sampling the analog signal
2. Quantization of the amplitude of the sampled signal
3. Coding of the quantized sample into digital signal
LPF
S/H  ADC PCM
S/H : Sample and hold 
circuit
Analog 
signal
Anti aliasing 
filter
ADC : analog to digital converter
PCM process:
f
s
4.6.1 Sampling 4.6.1 Sampling
• An analog signal must be sampled at Nyquist rate to avoid
aliasing
4.6.2 Quantization & Coding
• Process of estimating the sampled amplitude into a value suitable for
coding (ADC).
• A fixed number of levels including the maximum and minimum value of
the analog signal
• Number of levels is determined by the number of bits used for coding
• Quantization Interval
Represent the voltage value for each quantized level
For example: For a sampled signal that has 5V amplitude, V
pp
= 10 V
divide by the quantized level, L 

= 8 level,
Therefore, quantized interval ,
• Quantization level, L = 2
n
Quantization level depends on the number of binary bits, n used to
represent each sample.
For example:For = 3; Quantization level, L = 2
3
= 8 level.
In this example, first level (level 0) is represented by 000, whereas bit
111 represents the eigth level
V 25 . 1
8
V 10
· · ∆V
3 terms that are commonly used in the quantization 
process:
• Quantization value, V
k

The middle voltage for each quantized level
For example: for n = 3, quantized level, L

= 8 and a sampled sinusoidal
signal with +5 V ,
The middle quantized value for level 0,
In this example, for a sample that is in level 0 segment will be
represented by bit 000 with a voltage value of –4.375 V. The difference
between the sampled value and the quantized value results in
quantization noise.
V 375 . 4
2
V 25 . 1
V 5
0
− · + − · V
For voice communication 256 levels
are commonly used (i.e n = 8)
t
Level 0 : 000
Level 1 : 001
Level 2 : 010
Level 3 : 011
Level 4 : 100
Level 5 : 101
Level 6 : 110
Leve l 7 : 111
1.9V
+5.0V
-5.0V
4.375V
3.125V
1.875V
0.625V
-0.625V
-1.875V
-3.125V
-4.375V
4.3V
1.9V
-3.2V
-4.5V
Quantization level & 
binary representation 
Quantized 
value
Sampled signal
4.6.3 UNIFORM QUANTIZATION
Uniform quantization is a quantization process with a uniform (fixed)
quantization interval.
Example : n = 3 , L

= 8 , signal +5 V ; => V
k
= 1.25 V . Bit rate:
s b
nf f ·
Pemodulatan Digit
+m
p
-m
p
0
0 11
0 10
0 01
0 00
1 00
1 01
1 10
1 11

value
Sign but
t
000 001 011 011 011 010 001 100 110 111 111 110 100 001 010 010 010 000
Quantization error
Q
e
PCM code
t
The same code representing several
samples with different amplitudes
Step size
Pemodulatan Digit
May add to or substract from the
actual signal
Quantization error (Q
e
) is also called Quantization noise (Q
n
) . And its
maximum magnitude is one half of the voltage of the minimum step
size .
4.6.3.2 Quantization error 4.6.3.2 Quantization error
Input voltage range: –14 mV
to +14 mV
Binary
number
Input voltage
range (mV)
1 11

10 to 14
1 10 6 to 10
1 01 2 to 6
1 00 0 to 2
0 00 -2 to 0
0 01 -6 to -2
0 10 -10 to -6
0 11 -14 to -10
Example : Uniform Quantization error
Q

= LSB voltage /2 = ∆ /2 
t  14 mV = 28 mV with 8 steps and 8 codes. 
Therefore  ∆  = 28/8 = 3.5 mV.
Therefore : Q

= 3.5 mV / 2 = 1.75 mV
SNR
q
= [1.76 + 6.02n] dB : (for details, refer
to monograph page 122)
Noise from quantization error can be
reduced by increasing the quantization
level i.e increase n.
Nonuniform quantization
using µ  Law:
( ) [ ]
dB 02 . 6
1 ln
3
log 10
2
n SNR
q
+
+
·
µ
PCM 
system
Example :
V
pp
= 31.5 V

6 bit code (5 bits for
magnitude and 1 bit
for sign
(a) No of levels: 2
6
= 64
(b) LSB voltage, ∆ : 31.5/64 = 0.492 V
(c) Maximum quantization level, ∆ /2 = 0.25 V
(d) Voltage value for 101101 ; +(13 x 0.492) = +6.4 V
(e) Voltage value for 011001 ; –(25 x 0.492) = -12.3 V
(f) Code for input +13.62 V
= 13.62/0.492 = 27.68 ≈ 28 => 111100
(g)Code for input –9.37 V
= 9.37/0.492 = 19.04 ≈ 19 => 010011
4.6.4 Non uniform quantization
4.6.4 Non uniform quantization
nonuniform: to improve SNR (SQR)

More levels is available for low level amplitudes compared to high
amplitude

Increase SNR for low level amplitude and decrease SNR for higher
amplitudes
analog compression is done to the input signal before sampling and
quantization at the transmitter
Expansion is done at the receiver
COMPANDING (compression and expanding)
example : Non-Linear
Quantization
Pemodulatan Digit
Companding => Compress - Expanding
A method used to produce a uniform SNR for all input signal range is
compression-expansion (Companding).
Input signal is compressed at the transmitter and expanded at the
receiver
=> Analog – Compression process is done on the input signal
before sampling and coding
=> Digital – compression process is done after the signal is
sampled
Companding => Compress - Expanding
analog signal
(input)
analog
compressor
ADC
DAC
Analog
expander
Analog signal
(output)
PCM with analog compress-expand
To digital channel
analog signal
(input)
ADC
DAC
Digital
expander
Analog signal
(output)
PCM with digital compress-expand
To digital channel
Digital
compressor
Pemodulatan Digit
2 Popular companding system (standardized by ITU)
• EUROPE => A - Law
• USA/NORTH AMERICA => µ  - Law
A
x for
x
A
for
A
Ax
A
Ax
y
1
0
1
1
log 1
log 1
) log( 1
〈 〈
〈 〈
¹
¹
¹
¹
¹
'
¹
+
+
+
·
A - compressor paramater. Usually
the value of A is 87.6.
Pemodulatan Digit
USA/NORTH AMERICA => µ  - Law
µ Law  is a standard compress-
expand that is used in America
and Japan. The value of µ used
is 255 (8 bit).
( ) µ
µ
+
+
·
1 log
) 1 log( x
y
) (mak i
i
E
E
x ·
) (mak o
o
E
E
y ·
For both laws, the values of x and
y refers to the equation below:
Example 4.3 :
A compress-expand system using µ Law (µ = 255) is used for a signal
with range 0 to 10V. Determine the output of the system if the input is 0 and
7.5V.
Solution :
Given µ = 255 and E
i(mak)
= 10 V
For E
i
= 0 V
) (mak i
i
E
E
x · 0
10
0
· · x ;
Output :
( ) µ
µ
+
+
·
1 log
) 1 log( x
y
;
( ) 255 1 log
)) 0 ( 255 1 log(
+
+
· y
0 · y
For E
i
= 7.5 V
) (mak i
i
E
E
x · 75 . 0
10
5 . 7
· · x ;
( ) µ
µ
+
+
·
1 log
) 1 log( x
y
Output :
( ) 255 1 log
)) 75 . 0 ( 255 1 log(
+
+
· y
948 . 0 · y
Example 4.4 :
A random signal has gone through a 256 level quantization process.
Determine the quantization signal to noise ratio for this system.
Solution :
From the above statement, the number of sampling bits is not known.
But, given L=256
L = 2
n

therefore, n = 8
Given SNR
q
dB 02 . 6 76 . 1 n SNR
q
+ ·
dB 50 ) 8 ( 02 . 6 76 . 1 ≈ + ·
q
SNR
Europe bit rate(Mb/s)
2.048
8.448
34.368
139.264
565.148
Telephone
channel
30
120
480
1920
7680
SDH 2.5Gb/s
Telephone
channel
North America bit
rate(Mb/s)
24 1.544
48 3.152
96 6.321
672 44.736
4032 274.176
4.6.5 Bit rate for PCM transmission
European standard : A-Law
30 + 2 control channel = 32
Bit rate= 32 x 8 bit/sample x 8000 sample/s
= 2.048 Mb/s
North American standard (NAS) : µ -Law
For every 24 sample, 1 bit is added for
synchronization

For 24 sampel => 24 x 8 bit/sample
+ 1 bit = 193 bits
∴ Bit rate= 193 x 8000 = 1.544 Mb/s
Needs Multiplexing – Process of transmitting two or Needs Multiplexing – Process of transmitting two or
more signals simultaneously more signals simultaneously
Example : PCM-TDM CEPT System Example : PCM-TDM CEPT System
Frame structure and Timing : European standard PCM system : E Line
(a) bits per time slot (b) time slots per frame (c) frames per multiframe
488 ns
3.9
µ s
3.9
µ s
125
µ s
125
µ s
2 ms
8 bits per
time slot
Bit duration
30 signal + 2 control = 32 channels = 1 frame
Signalling & synchronization
16 frames = 1 multiframe
Duration of multiframe
Frame structure and timing
Number of channel = 32
Number of bits in one time slot = 8
32 channels = 1 frame
Number of bits in a frame = 32 x 8 = 256 bits
CEPT system – 32 channels (30 signals + 2 control)
This frame must be transmitted within the sampling period
and thus 8 x 10
3
frames are transmitted per second.
Therefore :
Transmission rate = 8 x 10
3
x 256 = 2.048 Mb/s
Bit duration = 1 / 2.048 x 10
6
= 488 ns
Duration of a time slot = 8 x 488 ns = 3.9 µ s
Duration of a frame = 32 x 3.9 µ s = 125 µ s => (= 1 / 8 kHz = 125
µ s)
Duration of a multi frame = 16 x 125 µ s = 2 ms
MUX
1
MUX
2
MUX
3
MUX
4
30
Voice 
channels
.
.
.
.
.
.
E1 line
2.048 Mbps
 E2 line
8.448 Mbps
E1
E1
E1
E2
E2
E2
E3
E3
E3
 E3 line
34.368 Mbps
E4 line
139.264 Mbps
CEPT telephone system hierarchy

There are 2 main components in the DM generator circuit,
i.e comparator and integrator.
Pemodulatan Digit





X ∑
-
+
Pulse signal
s(t)
comparator
integrator
d(t)
x
DM
(t) m(t)
e(t)
) (
~
t m

Comparator will compare the error signal e(t), where

Output signal from comparator has the following function:

The output from the comparator will be sampled with a
pulse signal at a rate of 1/T
s
.

Next, DM signal will be generated with the equation below:

The DM signal will be feed back, but before that this signal
will be integrated first

This signal will determine the error value e(t).
) (
~
) ( ) ( t m t m t e − ·



−∞ ·

−∞ ·
− ∆ ·
− ∆ ·
n
s s
n
s DM
nT t nT e
nT t t e t x
) ( )] ( sgn[
) ( )] ( sgn[ ) (
δ
δ
Pemodulatan Digit
¹
'
¹
∆ −
∆ +
· ∆ · )] ( sgn[ ) ( t e t d
0 ) (
0 ) (
<
>
t e
t e


−∞ ·
∆ ·
n
s
nT e t m )] ( sgn[ ) (
~
Pemodulatan Digit
) (t m
t
T
s
Δ
) (
~
t m
Effects of steep
slope
0001010111111101100010000000
If e(t) < 0 or -∆ , it will be coded as 0
If e(t) > 0 or +∆, it will be coded as 1
A steep slope results in noise in DM signal. To avoid this from
happening, it has to follow the following condition:
dt
t dm
mak
T
s
) (
>

Pemodulatan Digit
• Binary 1 and 0 in PCM signal can be represented by several formats
known as line coding.
information
PCM
Line
 coder
channel
Reasons for line coding:
1. Synchronization
2. Error detection
3. Error correction
4.8.1 Line code format
4.8.1 Line code format
Digital Signal Encoding Formats
A. NRZ (Non Return to Zero)
- Popular method
- easy
- Data does not return to 0 in one
clock interval
- No synchronization. Can use ‘start
bit’ for synchronization purposes
1. NRZ-L (NRZ-Level)
1 => High level
0 => Low level
2. NRZ-M (NRZ-Mark)
1 => transition at the starting interval
0 => no transition
3. NRZ-S (NRZ-Space)
1 => no transition
0 => transition at the starting interval
Digital Signal Encoding Formats
B. RZ (Return to Zero)
• Return to 0 at the half bit interval
• The same
advantages/disadvantages with
NRZ
• Overcome by using bipolar signal
and alternating pulse for
synchronization
4. RZ (Unipolar)
1 => High level
0 => Low level
5. RZ (Bipolar)
1 => Alternately +ve
0 => Alternately –ve
6. RZ (AMI – Alternately Mark Inversion)
1 => Alternately +ve and -ve
0 => Low level
Digital Signal Encoding Formats
C. Bi phase
• Used in optical communication
system, satellite and video
recorder
• Self synchronizing
7. Bi phase M
1 => transition at the middle of the
interval
0 => no transition at the middle of the
interval
8. Bi phase L (Manchester Coding)
1 => transition from HI to LO at the
middle of the interval
0 => transition from LO to HI at the
middle of the interval
used in Ethernet IEEE 802.3 standard in
LAN
9. Bi phase S – inverse of Bi phase M
1 => no transition in the middle of the
interval
0 => transition in the middle of the
interval

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