BASE BAND DATA

DATA
TRANSMISSION
MODULE 3
PART II
DIGITAL COMMUNICATION

Module III – Part II
Base band data transmission: - Discrete PAM
signals – Power spectra of discrete PAM signals -
Inter symbol interference- Nyquist’s criterion for
distortion less baseband binary transmission - Eye
diagram - Adaptive equalization.
Line Codes or Digital Formats

To send the encoded digital data over a communication channel we
require the use of a format or waveform for representing the binary
data.

Such digital formats are also called line codes.

Line codes specify how 0’s and 1’s are represented in a
communication channel.
Line Codes or Digital Formats
0 1 1 0 1 0 0 0 1

UNIPOLAR NRZ
SIGNALING

POLAR NRZ SIGNALING

UNIPOLAR RZ SIGNALING

BIPOLAR AMI SIGNALING

Line Codes or Digital Formats
0 1 1 0 1 0 0 0 1 1

UNIPOLAR SIGNALING

SPLIT PHASE OR
MANCHESTER ENCODING

POLAR RZ SIGNALING

POLAR QUARTERNARY
SIGNALING
Line Codes or Digital Formats

Unipolar NRZ format: Symbol 1 is represented by transmitting a
pulse of constant amplitude for the duration of the symbol and the
symbol 0 is represented by switching off the pulse. This type of
format is referred to as on off signaling.

Polar NRZ format: Symbols 0 and 1 are represented by pulses of
equal positive and negative amplitudes.

Unipolar RZ format: A rectangular pulse of half the symbol duration
is used for a 1 and no pulse for 0.

Bipolar NRZ format: Positive and negative pulses of equal
amplitudes are used alternatively to represent symbol 1 and no
pulse for symbol 0.

Split phase or Manchester encoding: Symbol 1 is represented by a
positive pulse followed by a negative pulse, with both pulses being
of equal amplitude and half symbol wide. For sym,bol 0 the polarities
are reversed.
Line Codes or Digital Formats

Polar RZ format: A rectangular pulse of half the symbol duration and
positive amplitude is used to represent symbol 1. A similar
rectangular pulse of equal negative amplitude is used to represent
symbol 0.

Polar quaternary format: Here the bits are grouped in to pairs and
the pairs of bits are represented by 4 voltage levels.
Bit Pair Voltage
00 -3/2
01 -1/2
10 1/2
11 3/2

M-ary coding: In this format we group together k bits and hence we

get M=2k distinct levels. According to the occurrence of a particular
group of bits we transmit the corresponding symbol.
Properties of Line Codes
 Power efficiency: For a given bandwidth and a specified detection error
probability the transmitted power should be as small as possible.
 Error detection and correction capacity: It should be possible to detect
and correct errors. In the bipolar case a single error will cause bipolar
violation and can be easily detected.
 Average DC value should be as small as possible: When DC and AC are
transmitted simultaneously through cable pairs AC coupling need to be
used. If the DC value is small ac coupling can easily be used.
 Adequate timing content: It should be possible to extract timing or clock
information from the signal.
 Transparency: It should be possible to transmit a digital signal correctly
regardless of the pattern of 1’s and 0’s. If the data is coded in such a way
that for every possible sequence of data the coded signal is received
faithfully the code is transparent.
 Transmission BW: Line codes should make the BW as small as possible
Power spectra of discrete PAM signals


 Base band transmission of PAM modulated
signals.
Input binary Clock Pulses HT ( f ) HC ( f )
Data
x (t )
PULSE TRANSMITTING
FILTER
CHANNEL
GENERATOR
bk
TRANSMITTER

HR ( f ) y (t ) sample at time t = iTb

Output
RECEIVING DECISION
binary data
FILTER DEVICE
y (ti )
Threshold
RECEIVER
 Base band transmission of PAM modulated
signals.

Consider a discrete pulse amplitude modulation system in which the
amplitude of the transmitted pulses is varied in a discrete manner in
accordance with the given digital data.
 The signal applied to the input of the system consists of a binary
data sequence bk with a bit duration Tb , bk is in the form of 1 or 0.
This signal is applied to a pulse generator, producing the pulse
waveform.
∞
x (t ) = ∑ A g (t − kT )
k = −∞
k b

 g(t) denotes a shaping pulse with its value at time t=0 defined by
g(0) =1. The amplitude Ak depends on the identity of the input bit bk.
+ a if bk is symbol 1
Ak = 
− a if bk is symbol 0
Base band transmission of PAM modulated
signals.

The PAM signal is passed through a transmitting filter of transfer
function of HT(f).

The resulting filter output defines the transmitted signal which is
modified in a deterministic fashion as a result of transmission through
the channel of transfer function HC(f).

At the receiver the signal is passed through a receiver filter of transfer
function HR(f).

This filter output is sampled synchronously with the transmitter with
the sampling instants being determined by a clock or timing signal that
is usually extracted from the receiving filter output.

The sequence of samples thus obtained is used to reconstruct the
original data sequence by means of a decision device.
 The amplitude of each sample is compared to a threshold. If the
threshold is exceeded a decision is made in favour of symbol 1. If the
threshold is not exceeded a decision is made in favour of symbol 0.
Inter Symbol interference

1 0 0

1 0 1 1

Pulse dispersion causes incorrect determination of bits at receiver

Inter Symbol interference

BIT STREAM

• • •
OUTPUT FROM
PERFECT FILTER

• • • •
• • OUTPUT FROM
•
IMPERFECT
• FILTER
• • •
 Inter Symbol interference

In the base band PAM system assume that the channel is noiseless.
Even then some errors occur in the bit determinations due to the
dispersive nature of the communication channel.

The receiving filter output of the system may be written as
∞
y (t ) = µ ∑ Ak p(t − kTb )
k =−∞

Where µ is a scaling factor and p(t) is a pulse whose shape is different
from g(t).

When the pulse Ak g(t) is applied to the input of the system we get
µAkp(t) at the output of the system after passing through the cascade
connection of transmitting filter, channel and receiving filter.
µ Ak P( f ) = Ak G ( f ) H T ( f ) H C ( f ) H R ( f )
µP( f ) = G ( f ) H T ( f ) H C ( f ) H R ( f )

Where P(f) and G(f) are the Fourier transform of p(t) and g(t)
respectively.
Inter Symbol interference

The receiving filter output y(t) is sampled at time ti=iTb (with i taking
integer values) yielding
∞ ∞
y ( ti ) = µ ∑
k = −∞
Ak p ( iTb − kTb ) = µ ∑ Ak p [ (i − k )Tb ]
k =−∞
∞
= µ Ai p(0) + µ ∑ Ak p [ (i − k )Tb ]
k =−∞
( k ≠i )
∞
= µ Ai + µ ∑ Ak p [ (i − k )Tb ] assuming p(0) = 1
k =−∞
( k ≠i )

The first term represents the contribution of the ith transmitted bit.

The second term represents the residual effect of all other
transmitted bits on the decoding of the ith received bit.

This residual effect is called inter symbol interference.

Inter Symbol interference

In the absence of ISI, y(ti)=µAi i.e., ith transmitted bit can be
decoded correctly.

The unavoidable presence of ISI introduces errors in the decision
device at the receiver output.

The channel transfer function Hc(f) and the pulse spectrum G(f) are
specified and so we have to adjust HT(f) and Hc(f) so as to enable
the receiver to correctly decode the received sequence of sample
values y(ti).
Nyquist’s criterion for distortion less
baseband binary data transmission

When the frequency response of the channel HC(f) and transmitted
pulse response G(f) are specified the problem is to determine the
frequency responses of the transmit and receive filters HT(f) and
HC(f) so as to reconstruct the original binary data sequence bk.

The correct decoding requires that y(ti)=µAi and the contribution
produced by other pulses be zero.

For this the pulse p(t) should be controlled such that

1 for i = k
p [ (i − k )Tb ] =  ........( 1 )
0 for i ≠ k

If p(t) satisfies the condition in eqn (1), y(ti)=µAi for all i which
implies zero intersymbol interference.
Nyquist’s first method

Time limited pulses cannot be band limited and vice versa. If we are
using perfect time limited pulses part of its spectra are suppressed
by a band limited channel. This causes pulse distortion. i.e.,
spreading of pulses.

One solution is to use pulses which are bandlimited so that they can
be transmitted through a bandlimited channel. But bandlimited
pulses cannot be time limited and there is spreading of pulses.

So we cannot avoid ISI even if we are using bandlimited or

timelimited pulses. It is inherent in a finite BW channel.

Pulse amplitudes can be detected correctly if there is no ISI at the

decision making instants. This can be accomplished by a properly
shaped bandlimited pulse.
Nyquist’s criterion for zero ISI

We choose a pulse shape that has a non zero amplitude at its
centre t = 0 and zero amplitudes at t = ±nTb, (n=1,2,3,….) where Tb
is the separation between successive transmitted pulse.
1 for t = 0
p (t ) = 
0 for t = ± nTb

A pulse satisfying this criterion causes zero ISI at all the remaining
pulses centers.

Consider the several pulses shown below centered at t=0,Tb, 2Tb,
3Tb,….. If we sample at 0, Tb,2Tb…. the sample value consists of
the amplitude of one pulse only with no interference from the
remaining pulses.

A pulse satisfying the above criteria is the sinc pulse.
1 for t = 0
sinc( f bt ) = 
0 for t = ± nTb
Nyquist’s criterion for zero ISI
1 1 1 1 1

0 Tb 2Tb 3Tb 4Tb 5Tb

Nyquist’s criterion for zero ISI

The Fourier transform of this pulse is
1  f 
P ( f ) = Tb rect ( fTb ) Or rect  
2B0 2
 0B
1 2 B0 1 2 B0 0 ≤ f ≤ B0
P( f ) =  1
0 f > B0 B0 =
2Tb
P( f ) = 1 2 B0 rect ( f 2 B0 )
− B0 B0

It has a bandwidth of 1/2Tb. Using this pulse we can transmit at a
rate of 1/Tb pulses per second without ISI over a bandwidth of 1/2Tb.

The parameter B0 is called Nyquist BW and it defines the minimum
transmission BW for zero ISI.
Nyquist’s criterion for distortion less
baseband binary data transmission
1
1 2 B0

− B0 B0

-4Tb -3Tb -2Tb -1Tb 0 1Tb 2Tb 3Tb 4Tb

Nyquist’s criterion for zero ISI
1 0 1 1 0 0 1 1
Nyquist’s criterion for zero ISI

Nyquist method solves the problem of ISI with minimum BW
possible but there are two practical difficulties in it.

The magnitude characteristics of P(f) should be flat from –B0 to
+B0 and zero else where. This is physically unrealizable because of
the abrupt transitions at the band edges ±B0.

The function p(t) decreases slowly because of the abrupt
discontinuity of P(f) at B0. So there is no margin of error in sampling
times at the receiver.

Since the sine function extends to infinity the tails of several pulses
get superimposed and we get large amplitudes at points away from
pulse centers.

So a little error in sampling time causes incorrect decision and bit
errors.
Sampling Errors Produces wrong results
1 1 0 1 1
Superposition
Of several sidelobes produces
large amplitude near to the
sampling point

Incorrect sampling time

produces wrong decisions
about the identity of the bit
Raised cosine spectrum

The ideal Nyquist channel solution of zero ISI is practically
unacceptable because of the following reasons.

(i) It is impossible to achieve a perfectly rectangular response.

(ii) Due to abrupt discontinuities in the spectrum the sinc
functions extends to infinity and it does not decay abruptly. It
leaves a tail extending to infinity. When such tails are
superimposed large amplitudes are available slightly near to
sampling points. So the sampling should be performed
without error.

One solution to these problems is to obtain a sinc pulse that

decays fast. For that the ideal spectrum with fast transitions
should be converted to a spectrum with edges rolling off to zero.

A spectrum meeting the above conditions is raised cosine
spectrum which is defined as follows:
 Raised cosine spectrum
 1
 2B for 0 ≤ f ≤ f1
 0
 1   π f − f1  
P( f ) =   1 + cos    for f1 ≤ f ≤ 2 B0 − f1
 4 B0   2 B0 − 2 f1  
0 for f ≥ 2 B0 − f1



The frequency parameter f1 and BW are related by

f1
α = 1− and f1 = (1 − α ) B0
B0
Raised cosine spectrum
α =0
2 B0 P( f )
α = 0.5
α =1
− 2B0 2B0
− B0 B0
−2 − 3/ 2 −1 0 p(t ) 1 3/ 2 2 f / B0
α =1
α =0
α = 0.5
2 1 0 1 2 t
Raised cosine spectrum

Parameter α is called roll off factor. It indicates the excess BW over
the ideal solution, Bo.

BW is extended from B0 to an adjustable value between B0 and 2B0

2B0-f1 is defined as transmission BW.
Transmission BW BT = 2 B0 − f1 = 2 B0 − (1 − α ) B0
= B0 + α B0 = B0 (1 + α )

The frequency response P(f) normalized by multiplying by 2B0 is
plotted for various values of α ie,0, 0.5 and 1.

The time response p(t) is the inverse FT of the frequency response
P(f).

 cos 2πα B0t 

p(t ) = sinc(2 B0t )  
 1 − 16α 2
B t
0 
Raised cosine spectrum

The time response p(t) consists of the products of two factors,
the factor sinc 2Bt characterizing the ideal Nyquist channel and a
second factor which decreases as 1/t2 for large t.

The first factor ensures that the zero crossings of p(t) are at
desired sampling instants of time iTb.

The second factor reduces the tails of the pulses considerably
below that obtained from the ideal Nyquist channel.

So transmission of binary waves using such pulses is relatively
insensitive to sampling time errors.
Eye Pattern

Eye pattern is used to study inter symbol interference.

For this we apply the received wave to the vertical deflection plates
of an oscilloscope and a saw tooth waveform at the transmitted
symbol rate 1/T to the horizontal deflection plates.

The waveforms in successive symbol intervals will be translated in
to one interval in the oscilloscope display.

The resulting display is called eye pattern.

The interior region of the pattern is called eye opening.

The eye pattern provides a number of information as given below:

1. The width of the eye opening defines the time interval over
which the wave can be sampled without error from intersymbol
interference. The preferred time for sampling is the instant of
time at which the eye is open widest.
Eye Pattern

2. The sensitivity of the system to timing error is determined by the
rate of closure of the eye as the sampling time is varied.

3. The height of the eye opening, at a specified sampling time,
defines the margin over channel noise.
 When the effect of inter symbol interference is severe, traces from
the upper portion of the eye pattern cross traces from the lower
portion, with the result that the eye is completely closed.
 In such a situation it is impossible to avoid errors due to the
combined effects of ISI and channel noise.
 In the case of an M-ary system the eye pattern consists of M-1 eye
openings stacked vertically one on the other where M is the number
of discrete amplitude levels used to construct the input signal.
Eye Pattern
1 0 1 1 0 0 1

Tb

Eye
Pattern

Tb
Eye Pattern

Eye pattern:Display on an oscilloscope which sweeps the system
response to a baseband signal at the rate 1/Tb (Tb symbol duration)

Distortion
due to ISI
Noise margin
amplitude scale

Sensitivity to
timing error

Timing jitter
time scale
Eye Pattern Best sampling
Margin over
Distortion at time
noise
sampling time
amplitude scale

time scale
Slope indicates Distortion of
Time interval during zero crossings
sensitivity to
which waveform can be
timing error
sampled
Eye Pattern

Perfect channel (no noise and no ISI)
Eye Pattern
AWGN (Eb/N0=20 dB) and no ISI
Eye Pattern

AWGN (Eb/N0=10 dB) and no ISI
Equalization

There are various amplifiers, filters and reactive circuit elements
throughout a communication system.

All these components are band limited to a specific bandwidth W
beyond which it cannot faithfully communicate a signal.

Most of the practical communication channels can be modeled as
Linear Time Invariant Systems. An LTI system has constant
amplitude and linear-phase frequency response

If we transmit the digital symbols at a rate that require slightly
greater BW than available BW, attenuation may occur but no
interference.

But for practical systems that are non ideal the amplitude response
is not flat or the phase response linear with frequency.

Transmission of digital symbols through such non ideal channel at a
transmission rate exceeding BW results in interference among a
number of adjacent symbols. Such distortion is called Inter Symbol
Interference.
Equalization

Intersymbol interference arises because of the spreading of a
transmitted pulse due to the dispersive nature of the channel, which
results in overlap of adjacent pulses.

If the channel is known precisely it is always possible to make the
ISI small by using a suitable pair of transmit and receive filters so as
to control the over all pulse shape.

Regardless of which particular pulse shape has been chosen some
amount of residual ISI remains in the output signal as a result of
imperfect filter design, incomplete knowledge of channel
characteristics etc.

The goal of equalizers is to eliminate intersymbol interference (ISI)
and the additive noise as much as possible and to overcome the
negative effects of the channel
Equalization

In general, equalization is partitioned into two broad categories

(i) Maximum Likelihood Sequence Estimation (MLSE) which entails
making measurement of channel impulse response and then providing
a means for adjusting the receiver to the transmission environment.
(Example: Viterbi equalization)

(ii) Equalization with filters, uses filters to compensate the distorted
pulses.

These type of equalizers can be grouped as preset or adaptive
equalizers.

Preset equalizers assume that channel is time invariant and try to find
H(f) and design equalizer depending on H(f). The examples of these
equalizers are zero forcing equalizer, minimum mean square error
equalizer, and decision feedback equalizer.

Adaptive equalizers assume that channel is a time varying channel
and try to design equalizer filter whose filter coefficients are varying in
time according to the change of channel, and try to eliminate ISI and
additive noise at each time. The implicit assumption of adaptive
equalizers is that the channel is varying slowly.
Adaptive Equalization

In the design of equalizers we assume that the channel
characteristics, either impulse response or frequency response
were known at the receiver.

In most communication systems that uses equalizers the channel
characteristics are not known before hand and the channel response
is time variant.

In such a case the equalizers are designed to be adjustable to the
channel response and to be adaptive to the time variations in the
channel response.

The process of equalization is said to be adaptive when the
equalizer is capable of adjusting its coefficients continuously during
the transmission of data.

The equalization is performed according to a well defined algorithm
to modify the received data.

This type of equalizers are called adaptive equalizers.
 Adaptive Equalization
SAMPLED
x(nT − T ) x(nT − 2T ) x(nT − 3T ) x(nT − MT + T )
INPUT
SIGNAL DELAY DELAY DELAY DELAY DELAY
T T T T T
x(nT)

VARIABLE
w1 w2 WEIGHTS wM −2 wM −1
w0
w

y(nT)
DESIRED
d (nT) e(nT)
ERROR
w

RESPONSE
Adaptive Equalization

Figure shows a popular structure used to design adaptive
equalizers.

The structure is a tapped delay line filter that consists of a set of
delay elements, a set of multipliers connected to delay line taps, a
corresponding set of adjustable tap weights and a summer for
adding the multiplier outputs.

Let the sequence x(nT) appearing at the output of the receiving filter
be applied to the input of the tapped delay line filter producing the
output M −1
wi ⇒ Weight of the i tap
∑
th
y (nT ) = w i x ( n T − iT )
i=0 M ⇒ Total numberof taps

The M tap weights constitute the adaptive filter coefficients.

The tap spacing is chosen equal to the symbol duration T of the
transmitted signal.

The following steps are carried out in the adaptation process.
Adaptive Equalization

(i) A known sequence d(nT) is transmitted, and in the receiver the
resulting response sequence y(nT) is obtained by measuring The
filter output at the sampling instants.

(ii) Viewing the known transmitted sequence d(nT) as the desired
response, the differences between it and the response sequence
y(nT) is computed. The difference is called error sequence,
denoted by e(nT)
e ( nT ) = d ( nT ) − y ( nT ), n = 0 ,1, 2 ,...... N − 1
N ⇒ Total length of the sequence

(iii) The error sequence e(nT) is used to estimate the direction in
which the weights wi of the filter are changed so as to make
them approach their optimum settings.
Adaptive Equalization

A criterion used for optimization is the total energy of the error sequence
defined by
N −1
E = ∑e
n=0
2
( nT )

The optimum values of the tap weights wo0, wo1, wo2,.........., woM−1 result when
the total energy E is minimized.

So an algorithm is required that adjusts the tap weights of the filter in a
recursive manner.

The present estimate of each tap weight is updated by incrementing it by
a correction term proportional to the error signal at that time.

A commonly used algorithm is Least Mean Square Algorithm.

According to LMS algorithm the tap weights are adapted as follows.
UPDATED OLD VALUE INPUT SIGNAL
VALUE OF
THE Kth TAP = OF THE Kth + STEP SIZE
PARAMETER
x APPLIED TO THE x ERROR
SIGNAL
TAP WEIGHT Kth TAP WEIGHT
WEIGHT
Adaptive Equalization
wˆ ( n T + T ) = wˆ i ( n T ) + µ e ( n T ) x ( n T − iT )
wˆ i ( nT ) ⇒ Present estimate of the optimum weight woi for tap i at time nT.
wˆ ( nT + T ) ⇒ Updated estimate µ ⇒ Adaptation constant
i = 0,1,2,.............,M - 1

The adaptation constant µ controls the amount of correction applied
to the old estimate wˆ ( nT ) to produce the updated estimate wˆ ( nT + T )

The correction depends on the filter input x ( nT − iT ) and the error
signal e( nT ) both measured in time nT

By a proper choice of the adaptation constant, the use of the
recursive equation helps the adjustment of the tap weights move
toward their optimum settings in a step by step fashion.
Adaptive Equalization
Signal Flow Graph Of
CORRECTION Least Mean Square
µx ( nT − iT )e(nT ) Algorithm

+ +
OLD VALUE UPDATED
w
VALUE
wˆ i ( nT ) wˆ ( nT + T )

UNIT DELAY
Adaptive Equalization

LMS algorithm requires a knowledge of the desired response d(nT) and
the filter response y(nT) to form the error signal e(nT).

For this, prior to data transmission, the equalizer is adjusted under the
guidance of a training sequence transmitted through the channel.

Thus there are two modes of operation for the adaptive equalizer
(i) Training mode (ii) Decision directed mode.

During the training mode a known sequence is transmitted and a
synchronized version of it is generated at the receiver where it is
applied to the equalizer as the desired response.

The tap weights of the equalizers are then adjusted in accordance with
LMS algorithm.

When the training process is completed, the adaptive equalizer is
switched to its second mode of operation.

In this mode of operation, the error signal is defined by
e ( nT ) = dˆ ( nT ) − y ( nT )
dˆ(nT) ⇒ final correct estimate of the transmitte d symbol d(nT)
 Adaptive Equalization
DECISION
DIRECTED
MODE

x (nT ) y (nT ) d (nT )

2 1 TRAINING
ADAPTIVE DECISION
SEQUENCE
EQUILIZER DEVICE
dˆ ( nT ) GENERATOR

e(nT ) −
TRAINING
W

+ MODE
Duo Binary Encoding

Consider a binary input sequence b(k) consisting of uncorrelated
binary symbols 1 and 0 each with duration Tb.

This sequence when applied to a pulse amplitude modulator
produces a two level sequence whose amplitude ak is denoted by
+ 1 if symbol bk is 1
ak = 
− 1 if symbol bk is 0

When this sequence is applied to a duo binary encoder, it is
converted in to a three level output consisting of -2, 0, +2

One of the effects of the transformation described above is to
change the input sequence bk of uncorrelated binary digits in to
sequence ck of correlated digits.

This correlation between adjacent transmitted levels may be viewed
as introducing ISI in to the transmitted signal in a deliberate manner.

This type of coding is called correlative coding also.
Duo Binary Encoding
Output
bk ak Sequence ck
BIPOLAR + IDEAL
CONVERTER CHANNEL
Hc(f)
+ Sample at
DELAY time t=kTb
Tb

Filter

TRANSFER
bk FUNCTION
OF SYSTEM
ck
H(f)
Duo Binary Encoding- An Example
ck = ak + ak −1 aˆk = ck − aˆk −1
a0 = +1 aˆ0 = +1
a1 a2 a3 a4 a5 c1 c2 c3 c4 c5
-1 -1 +1 +1 +1 0 -2 0 2 2

c1 = a1 + a0 = −1 + 1 = 0
c2 = a2 + a1 = −1 − 1 = −2
c3 = a3 + a2 = 1 − 1 = 0
c4 = a4 + a3 = 1 + 1 = 2
c5 = a5 + a4 = 1 + 1 = 2
ENCODING
aˆ1 = c1 − aˆ0 = 0 − 1 = −1
aˆ2 = c2 − aˆ1 = −2 − −1 = −1
aˆ3 = c3 − aˆ2 = 0 − −1 = +1
aˆ4 = c4 − aˆ3 = +2 − 1 = +1
aˆ5 = c5 − aˆ4 = +2 − 1 = +1
DECODING
Duo Binary Encoding

The polar sequence ak is first passed through a simple filter
consisting of the parallel combination of a direct path and an ideal
element producing a delay of Tb seconds.

For every unit impulse applied to the input of this filter we get two
unit impulses spaced Tb seconds apart.

The output of this filter in response to the incoming polar sequence
ak is then passed through the channel of transfer function Hc(f).

A continuous waveform is thus produced at the channel output. The
resulting waveform is sampled uniformly every Tb seconds, thereby
producing the duo binary encoded sequence.

The effect of the channel is included in this encoding operation.
Duo Binary Encoding

The cascade connection of the delay line filter and the channel is
called a duo binary conversion filter.

An ideal delay element producing a delay of Tb seconds, has the
transfer function e − j 2πfTb so that the transfer function of the delay
line filter is 1 + e − j 2πfTb

Hence the over all transfer function of this filter connected in
cascade with the ideal channel Hc(f) is
[
H ( f ) = HC ( f ) 1 + e− j 2πfTb ]
[
= H C ( f ) e jπfTb .e − jπfTb + e − jπfTb .e − jπfTb ]
[( )
= HC ( f ) e jπfTb + e− jπfTb e− jπfTb ]
= 2 H C ( f ) cos(πfTb )e − jπfTb
Duo Binary Encoding

For an ideal channel with band width B0=1/2Tb
1 f ≤ 1/ 2Tb
HC ( f ) = 
0 otherwise

Thus the over all frequency response has the form of a half cycle
cosine function as shown by

 2 cos (πfTb )e − jπfTb f ≤ 1 / 2Tb

H( f ) = 
0 Otherwise
H( f ) ∠H ( f )
2 π /2
1 2Tb
1 2Tb 0

− 1 2Tb 1 2Tb −π /2
Duo Binary Encoding

The corresponding value of the impulse response consists of two
sinc pulses time displaced by Tb seconds as given by
sin( πt / Tb ) sin( π ( t − Tb ) / Tb )
h (t ) = +
πt / Tb π ( t − Tb ) / Tb
sin( πt / Tb ) sin( πt / Tb )
= +
πt / Tb π ( t − Tb ) / Tb
Tb sin( πt / Tb )
2
=
πt (Tb − t ) Tb sin( πt / Tb )
2
h (t ) =
πt (Tb − t )
Duo Binary Encoding
sin( πt / Tb ) sin( π ( t − Tb ) / Tb )
π t / Tb π ( t − Tb ) / Tb

− 3Tb − 2Tb − Tb 0 Tb 2Tb 3Tb

Duo Binary Encoding
h(t )
sin( πt / Tb )
1. 0 sin( π ( t − Tb ) / Tb )
πt / Tb π ( t − Tb ) / Tb

− 3Tb − 2Tb − Tb 0 Tb 2Tb 3Tb

− 3Tb − 2Tb − Tb 0 Tb 2Tb 3Tb

Duo Binary Encoding

H( f )
2

− 1 2Tb 1 2Tb

TRANSFER
bk FUNCTION
OF SYSTEM
ck
H(f)
Duo Binary Encoding

The over all impulse response has only two distinguishable values
at the sampling instants.

The original data ak may be detected from the duo binary coded
sequence ck by subtracting the previous decoded binary digit from
the currently received digit ck .

Let â k represent the estimate of the original digit ak as detected by
the receiver at time t = kTb. Then
aˆ k = c k − aˆ k − 1

If ck is received without error and if the previous estimate aˆ k − 1 at
time t = (k-1)Tb also corresponds to a correct decision, then the
current estimate â k will be correct too.

This method of using a stored estimate of the previous symbol in the
estimation of the current symbol is called decision feedback.
Duo Binary Encoding With Precoder

In the case of duo binary decoding once errors are made they tend to
propagate as a decision on the current binary digit bk depends on the
correctness of the decision made on the previous binary digit bk-1

A practical means to avoid this error propagation is to employ
precoding before duo binary coding.

The precoding operation converts the input binary sequence bk to
another binary sequence dk as below
d k = bk ⊕ d k −1

The pre-coder output is converted to ak with polar representation as
− 1, if dk = 0
ak = 
+ 1 if dk = 1

The resulting output ak is then applied to duo binary coder thereby
producing the sequence ck as follows
c k = a k + a k −1
 Duo Binary Encoding With Precoder

What ever the value of dk-1, we get

± 2, if bk = 0
ck =  Sample at
0 if bk = 1 t = kT b

bk dk ak
POLAR DUOBINARY
CONVERTER ENCODER ck
d k −1
DELAY
Tb

PRE-CODER
Duo Binary Encoding With Precoder
 − 1, if d =0
ak = 
k
Let d k −1 = 0
+ 1 if d k =1
 − 1, if d k −1 = 0
a k −1 =
+ 1 if d k −1 = 1

Let bk = 0 Let bk = 1
d k = bk ⊕ d k −1
d k = bk ⊕ d k −1
dk = 0 ⊕ 0 = 0 dk = 1⊕ 0 = 1
c k = a k + a k −1 c k = a k + a k −1
ck = − 1 + − 1 = − 2 ck = 1 − 1 = 0

ENCODING WITH dk-1=0

Duo Binary Encoding With Precoder
 − 1, if d =0
ak = 
k
Let d k −1 = 1
+ 1 if d k =1
 − 1, if d k −1 = 0
a k −1 =
+ 1 if d k −1 = 1

Let bk = 0 Let bk = 1
d k = bk ⊕ d k −1 d k = bk ⊕ d k −1
dk = 0 ⊕ 1 = 1 dk = 1⊕ 1 = 0
c k = a k + a k −1 c k = a k + a k −1
ck = 1 + 1 = 2 ck = − 1 + 1 = 0

ENCODING WITH dk-1=1

Duo Binary Encoding With Precoder

So the following decision rule may be adapted for constructing the
decoded binary sequence bk at the receiver output.

symbol 0, if ck > 1V
bˆk =  Threshold = 1
symbol 1, if ck ≤ 1V bˆk = 0 ,if ck > 1V
bˆk = 1,if ck ≤ 1V
ck ck b̂k
THRESHOLD
RECTIFIER DETECTOR

The detector consists of a rectifier, the output of which is compared

to a threshold of 1V, and the original binary sequence is thus
detected.

No knowledge of any input sample other than the present sample is
required. Hence error propagation cannot occur in the detector.
 Duo Binary Encoding With Precoder
An Example
Let us Start with dk-1= 1
Input binary sequence bk 0 0 1 0 1 1 0
Precoded binary seq uence d k 1 1 1 0 0 1 0 0
Polar representa tion ak +1 +1 +1 -1 -1 + 1 -1 -1
Duo binary coder output ck 2 2 0 -2 0 0 -2
Decoded binary sequence bˆk 0 0 1 0 1 1 0

d k = bk ⊕ d k −1 c k = a k + a k −1
symbol 0, if ck > 1V
 − 1, if d =0 ˆb = 
ak = 
k k
symbol 1, if ck ≤ 1V
+ 1 if d k =1
 Duo Binary Encoding With Precoder
An Example
Let us Start with dk-1= 0
Input binary sequence bk 0 0 1 0 1 1 0
Precoded binary seq uence d k 0 0 0 1 1 0 1 1
Polar representa tion ak −1 −1 −1 +1 +1 −1 +1 +1
Duo binary coder output ck −2 −2 0 2 0 0 2
Decoded binary seq uence bˆk 0 0 1 0 1 1 0

WE GET THE SAME DECODED

OUTPUT WHEN WE START
WITH dk-1=0 OR dk-1=1