Digital Communication System

The term digital communication converts a broad area of communication

techniques, including digitally transmission and digitally radio. Digitally
transmission is the transmittal of digital pulses between two or more points in a
communication system. Digitally radio is the transmittal of digital modulated
analog carriers between two or more points in communication system.

# Why Digital ?
(1) Figure below shows an ideal binary digital pulses propagating along a
transmission line.

The shape of the waveform is affected by two mechanisms: -

(a) As all the transmission line and circuit have some nonideal transfer function,
there is a distorting effect on the ideal pulse.
(b) Unwanted electrical noise or other interference further distorts the pulse
waveform.
Both of these mechanisms cause the pulse shape to degrade as a function of
distance (line length). As shown in fig. During the time that the transmitted pulse
can still be reliably identified, the pulse is thus regard. Circuits that perform this
function at regular intervals along a transmission system are called regenerative
repeaters.
(2) Digital circuits are less subjected to distortion and interference than
analogue circuits.
(3) Digital circuits are more reliable and can be produce at lower cost than
analogue circuits.
(4) Digital hardware lends itself to more flexible implementation the analogue
hardware (e.g. microprocessor, digital switching, and large-scale integrated
circuits LSIC).
(5) Digital techniques lend themselves naturally to signal processing functions
that product against interference and jamming.
(6) Much data communication is computer to computer, or digital instrument or
terminal to computer. Such digital terminations are naturally best served by
digital link.

Communication System Models: -

Generally there are two types for communication system models, Baseband
(or Lowpass) and Passband (or Bandpass) models.
In baseband model, the spectrum of signal from zero to some frequency (i.e. carrier
frequency =0). For transmission of baseband signal by a digital communication
system the information is formatted so that it represented by digital symbols. Then,
pulse waveforms are assigned that represented these symbols; this step is referred
to as pulse modulation or baseband modulation. These waveforms can then be
transmitted over a cable. Baseband signal also called lowpass signal.

In the passband signal, the signal has a spectral magnitude that is nonzero for
frequency in some band concentrate about frequency (f= +f c , -f c ) and negligible
elsewhere, where f c is the carrier frequency need to be much greater than zero. The
channel over which a signal is transmitted is limited in bandwidth to an interval of
frequencies centered about the carrier, as in double sided band modulation, or
adjacent to the carrier as in a single side band modulation. For radio transmission
the carrier is covered to an electromagnetic (EM) field for propagation to desired
destination.
Baseband system:-
Sampling Theorem
-The link between an analog waveform and its sampled version is
provided by the sampling process. This process can be
implemented in several ways, the most popular being the sample
and hold operation .In this operation a switch and storage device
(trasister and a capacitor ) form a sequence of samples of the
continuous input waveform . The output of the sampling process
is called pulse amplitude modulation (PAM)because the
successive output intervals can be described as as a sequence of
pulses with amplitude derived from the input waveform pulses.
The sampling theorem
-A bandlimited signal having no spectral components above fm
hertz can be determind uniquely by values sampled at uniform
intervals of
1
Τ  ............(3.1)
s 2 fm
This statement is Known as the uniform sampling theorem .
Where Ts = sampling time (which is the upper limit )
fs = sampling frequency = 1/ Ts
f s  2 f m ............(3.2)

3-1
The sampling rate fs=2fm is called the Nyquist rate. Therefore the
theorem requires that the sampling rate be rapid enough so that at
least two samples are taken during the period corresponding to the
highest frequency in the spectrum .
Let us examine case of ideal sampling with a sequence of unit
impulse functions .
1-Assume an analog signal x(t) as shown in fig(3.3a)with a
Fourier transform X(f) ,which is zero outside the interval
(-fm < f < fm) as shown in fig(3.3b)
2-The sampling of x(t) can be viewed as the product of x(t) with a
periodic train of unit impulse functions x(t) as shown in ig(3.3c)

………….(3.3)

where Ts is the sampling period and  (t) is the unit impulse .

The

3-2
 Fig (3.3)

3) Using the frequency convolution property of the fourier

transform
F  (t) X  (t) X ( f ) * X  ( f )


  ( f n f s).............(3.5)
1
where Xs(f) =
Ts n

Fig(3.3c) and fig(3.3d) show the impulse train X(t) and its
Fourrier transform Xδ(f).respectively .Convolution with an
impulse function simply shifts the original function as follows
X (f) * δ (f  n f s )  X (f  n f s )

3-3
We can now solve for the transform X s (f) of the sampled
waveform
 1  
X (f)  X (f) * X δ
(f)  X (f) * 
  δ(f n f s ) 
s  T s n    
1 
  X (f n fs ) ..............(3.6)
T n  
Fig (3.3f) show the spectrum of Xs (f) . We therefore conclude
that:-
1-The spectrum Xs(f) of the sampled signal s(t) is to within a
constant factor (1/Ts) , exactly the same as that of (t)
2-The spectrum repeats itself periodically in frequency every fs
(Hertz)
3-When fs=2fm , the analog waveform can theoretically be
completely recovered by using sharp lowpass filter
4-If fs > 2 fm ,the analog waveform can be recovered easier by
using low pass filter .
5-If fs < 2 fm the spectrum of analog waveform components
overlap as shown in fig (3.4b) . Some information will be lost .
The result of under sampling ( fs < 2 fm )and the phenomenon is
called aliasing.

3-4
 Fig (3.4)
Natural Sampling
- The practical method of accomplishing the sampling of a
bandlimited analog signal  (t ) is to multiply (t) shown in
fig(3.5a)by the pulse train or switching waveform  p (t ) shown

in fig(3.5c) Each pulse in  p (t ) has width T and amplitude (1/T)

The sampling frequency is designated fs , and sampling time is

designnated Ts(TS=1/fs) . The sampled data  s (t ) can be

expressed as
 s (t ) =  p (t )  (t ) ………. (3.7)
The sampling here is termed natural sampling since the top of
each pulse in the s(t) sequence has the same shape of its

3-5
corresponding analog segment during the pulse interval .The
periodic pulse train can be expressed by a Fourier series

Fig (3.5 )

3-6
  j 2f s t
 p (t )   Cn e ..........................(3.8)
n  

where f s  1
Ts
 2 fm
1  nT 
Cn  sin c  s 
Ts  T 
T  pulse width
Combining eq (3.7) and eq (3.8)

j 2  n f st
 s (t )   (t ) 
n  
Cn e ...........(3.9)

The transform Xs (f) of the sampled waveform is

Xs (f) = F [s (t) ]

…………(3.10)

For linear systems ,we can interchange the operations of

summation and Fourriers transformation .

……….(3.11)

Using the Fourier translation property of the Fourier transform:-


:. Xs ( f )   Cn
n  
X ( f  nf ) ..........(3.12)
s

3-7
Eq(3.12) and fig(3.5f) illustrate that Xs(f) is a replication of X(f)
periodically repeated in frequency every fs. In this natural
sampled case , we see that Xs(f) is weighted by the Fourier series
coefficients of the pulse train compared with a constant value in
the impulse sampled case

Ex.1 Consider a given waveform (t) with Fourier transform X(f)

Let Xs1 (F) be the spectrum of s(t), which is the result of
sampling (t)with a unit impulse train (t).Let Xs2(f) be the
spectrum of s2(t) , the result of sampling (t)with a pulse train
p(t)with pulse width T, amplitude 1/T, and period Ts. Show that
in the limit, as T approaches zero, Xs1(f)=Xs2(f)
Solution :-
1 
X (f)   X ( f  n f s ) .............(3.6)
s1 Ts  
n


X s2 ( f )   Cn X ( f  n f s ) ..............(3.12)
n

As the pulse width T0 and the pulse amplitude  (the erea
of the pulse remains unity), p (t) δ (t)

3-8
 1 T /2
:. Cn lim   (t ) e  j 2  f s t d t
s

T 0 T T /2 p
s s
Ts / 2
1  j 2 n f s t 1 T /2  j 2nf s t 1
Ts  Ts / 2
Cn     
Ts T
s
 (t ) e d t (t )e dt
Ts
s/2

Substitute for Cn in eq (3.12)

:. X 1
s 2( f )
Ts

In the limit T →0



1
X s2 ( f )   ( f n f s )  X s ( f ) .
Ts n 

Aliasing
-Fig(3.6)shows aliasing in frequency domain .The overlapped
region , shown in fig(3.6) contains that part of the spectrum which
is aliased due to under sampling
-Higher sampling rate (fs) can eliminate the aliasing.

3-9
 Fig (3.6)

Sampling Theorem and Bandpass Signals

for a signal g(t) whose highest frequency spectral component is
fm, the sampling frequency fs must be no less than fs=2fm if the
lowest frequency spectral component of g(t) is fL=0
If fL ≠ 0 it may be that the sampling frequency need be no larger
than fs = 2 ( fm-fL)

Ex.2 : if the spectral range of a signal extends from 10 to 10.1 M

Hz. Find the value of sampling frequency
Solution
fs= 2 ( fm – fL)
= 2 ( 10.1-10) = 0.2 M Hz

3-10
Digital Communication I

Hussain Abdulkarim
Sampling Theorem
Unit Impulse Signal

b a
 Sampling Theory
• fs ≥ 2fm
• fs ≥ 2W
• fs = sampling frequency = 1/ Ts

• Ts = sampling time
• Nyquist rate fs = 2fm
• Nyquist rate fs =2 W
Ideal Sampling
Natural Sampling
Practical
(Flat-top)
• If fs < 2 fm the spectrum of analog waveform
components overlap Some information will
be lost .
• The result of under sampling ( fs < 2 fm )and the
phenomenon is called aliasing.
 EX.
• Determine the Nyquist rate of sampling for the
signal:
• G(t)= 10Cos(100πt) +15 Cos(150πt)+ 5Cos(300πt)
Sol

Fmax =300π/2π =150Hz

Nyquist rate=2fmax=2*150=300Hz
 Digital Communication I

Hussain Abdulkarim
Time Division Multiplexing (TDM)
Pulse Modulation (PAM,PWM,PPM)
 Time Division Multiplexing
• Time interleaving of samples from different sources to be transmitted over a
single communication channel.
 Multiplexing Techniques
• Frequency Division Multiplexing 'FDM“
• Time Division Multiplexing "TDM“
• Space Division Multiplexing "SDM“
• Wavelength Division Multiplexing "WDM“
• Code Division Multiple Access "CDMA“
• Time and Frequency Multiplexing
 Basics Time Division Multiplexing

• Synchronous TDM.
• Asynchronous TDM.
 PAM TDM

Frame Time = Tf
Tf = Ts =1/2W = 1/2fm
Pulse duration = Ʈ = Tf /K
Minimum required B.W = BT
BT =0.5/Ʈ
 EX.

• K=7
• Spaces =8
• fs = 5KHZ
• [Link] = 2
 Pulse Modulation
• Amplitude A
• Width Ʈ
• Period T
PAM
PWM
PPM
Digital Communication I

Hussain Abdulkarim
Pulse Code Modulation
PCM
The formatting and transmission of baseband signals
A sampled signal and the quantized levels
•L=2 k OR M=2N

K=Log2L OR N=Log2M
Where L ,M = LEVELS
K,N = Bits

• [Link]=B.W *N
ΔV= 2*Vm/L
ΔV/2=Quantization Error
Where ΔV is the step size
between quantization
levels
 Quantization noise in PCM
So/No=3L2 •
• (So/No)dB=4.8+20Log10L
So=peak signal power •
No=r.m.s noise power •
Also So/No=(3/2)L2 •
Where So is the r.m.s value •
 Ex.
• X(t)=4sin(2π100t)
• K=3
• Sol.
• L=2k
L= 23
=8
• L=8 Levels
• ΔV=2*Vm/L=2*4/8=1V
• ΔV/2=Quantization Error 1/2Volt
• ( 0.5 , - 0.5)
 The output SNR

• So/Nq=3L2f2(t)/Vp2
• No=Nq, So/No=So/Nq
• Vp2= fp2
• fp= peak voltage level
• So/Nq=3/2 L2 For tone modulation

• (So/Nq)dB=1.76+20LogL
• =1.76+6.02n
 Ex.
In a binary PCM system , the output signal -to-quantization
ratio is to be hold to a minimum of 40dB .If the message is
a single tone with fm=4KHZ .Determine .
1- The number of required levels and the corresponding
output signal-to-quantizing noise ratio.
2-Minimum required system bandwith.
 Sol.
1- L=2k
So/Nq=10000=40dB
So/Nq=3/2 L2
L=82
K=Log282=(6.36)=7
L=27=128
2- Minimum system B.W =k* Fmax
=7*4KHZ=28KHZ
To find minimum step size to avoid slope
overloading ΔVmin=2π(800)fp/fs
To find SNRq
So/No=3fs f2(t)/(ΔV)2.B
Where fs= sampling frequency
Digital Communication I

Hussain Abdulkarim
Quantization
 Quantization
Quantization is a non linear transformation which maps elements
from a continuous set to a finite set. It is also the second step
required by A/D conversion.

Analog Signal Sample Quantize Digital Signal

- Continuous time - Discrete time
- Continuous value - Discrete time - Discrete value
- Continuous value
 Uniform Quantization
output w2(t)
V

-V V
input w1(t)

-V
Region of operation
For M=2n levels, step size ∆:
∆ = 2V /2n = V(2-n+1)
 Quantization Error, e
output w2(t)
V

-V V
input w1(t)

-V

Error, e
∆/2
-∆/2 input w1(t)
Definition. The dynamic range of an input signal is the
ratio of the largest to the smallest power levels which the
input signal can take on and be reproduced with the
acceptable signal distortion.

The dynamic range of the quantizer input in the PCM

system is 6n dB.
 Nonuniform Quantizer

Used to reduce quantization error and increase the dynamic

range when input signal is not uniformly distributed over its
allowed range of values.

allowed
values input

values
for most time
of time
“Compressing-and-expanding” is called “companding.”
Nonuniform quantizer

Discrete Uniform digital

samples Compressor Quantizer signals

••••
Channel
••••

Decoder Expander output

received
digital
signals
 Compression Techniques
A - law compressor
w1 (t ) ≤ 1, A≥0
 A w1 (t ) 1
 0 ≤ w1 (t ) ≤
 1 + ln A A
w2 (t ) = 
1 + ln (A w1 (t ) ) 1
 ≤ w1 (t ) ≤ 1
 1 + ln A A

µ - law compressor
(very popular internationally)

w1 (t ) ≤ 1
ln (1 + µ w1 (t ) )
w2 (t ) =
ln(1 + µ )
In the U.S., µ = 255 is used.
Practical Implementation of µ-law compressor
Output SNR of 8-bit PCM systems
with and without companding.
 Baseband Signaling

Transmitter Receiver
w(t) w#(t) Optimal
Baseband Channel
Filter
Signaling H(f)

• Once the sending end prepared digital signals (e.g.,

PCM) to send, now it is the job of Baseband Signaling
to
prepare the signals suited for the channel.
• What should w(t) be?
→ Orthogonal set N of signals {φk (t), k=1,2,3, ..., N}
w(t ) = ∑ wkφk (t ), for 0 < t < To
k =1
 1

Quantization
 2

Quantization
• Amplitude quantizing: Mapping samples of a
continuous amplitude waveform to a finite set of
amplitudes. Out

In
Average quantization noise power
Quantized

Signal peak power

values

Signal power to average

quantization noise power
 3
Qunatization example
amplitude
x(t)
111 3.1867

110 2.2762 Quant. levels

101 1.3657

100 0.4552

011 -0.4552 boundaries

010 -1.3657

001 -2.2762 x(nTs): sampled values

xq(nTs): quantized values
000 -3.1867
Ts: sampling time
PCM t
codeword 110 110 111 110 100 010 011 100 100 011 PCM sequence
4
 5

Quantization Effect
• 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.
 6

Non-uniform Quantization

• Nonuniform quantizers have unequally spaced levels

▫ The spacing can be chosen to optimize the Signal-to-Noise Ratio
for a particular type of signal
• It is characterized by:
▫ Variable step size
▫ Quantizer size depend on signal size
 7

 M any signals such as speech have a nonuniform distribution

Basic principle is to use more levels at regions with large probability density
function (pdf)
 Concentrate quantization levels in areas of largest pdf
Or use fine quantization (small step size) for weak signals and coarse
quantization (large step size) for strong signals
8
 9
Non-uniform Quantization

Non-uniform quantization is achieved by, first passing the input signal through a
“compressor”. The output of the compressor is then passed through a uniform
quantizer.

The combined effect of the compressor and the uniform quantizer is that of a non-
uniform quantizer.

At the receiver the voice signal is restored to its original form by using an
expander.

This complete process of Compressing and Expanding the signal before and after
uniform quantization is called Companding.
 10
Non-uniform Quantization (Companding)

Compressor Uniform Quantizer Expander

m(t ) mˆ (t )
 11
Non-uniform Quantization (Companding)

Compressor Uniform Quantizer Expander

m(t ) mˆ (t )

The 3 stages combine to give

the characteristics of a Non-
uniform quantizer.
 12

• Basically, companding introduces a nonlinearity into the signal

▫ This maps a nonuniform distribution into something that more
closely resembles a uniform distribution
▫ A standard ADC with uniform spacing between levels can be used
after the compandor (or compander)
▫ The companding operation is inverted at the receiver

• There are in fact two standard logarithm based companding

techniques
▫ US standard called µ-law companding
▫ European standard called A-law companding
 13
Nonuniform quantization using companding
• Companding is a method of reducing the number of bits required in
ADC while achieving an equivalent dynamic range or SQNR
• In order to improve the resolution of weak signals within a converter,
and hence enhance the SQNR, the weak signals need to be
enlarged, or the quantization step size decreased, but only for
the weak signals
• But strong signals can potentially be reduced without significantly
degrading the SQNR or alternatively increasing quantization step size
• The compression process at the transmitter must be matched with an
equivalent expansion process at the receiver
 14

• The signal below shows the effect of compression, where the

amplitude of one of the signals is compressed
• After compression, input to the quantizer will have a more
uniform distribution after sampling

 At the receiver, the signal is

expanded by an inverse operation
 The process of CO M pressing and
exPANDING the signal is called
companding
 Companding is a technique used
to reduce the number of bits
required in ADC or DAC while
achieving comparable SQNR
 15
Input/Output Relationship of Compander

• Logarithmic expression Y = log X is the most commonly

used compander
• This reduces the dynamic range of Y
 16
Types of Companding
µ -Law Companding Standard (North & South America, and
Japan)

log e [1 + µ (| x | / xmax ]
y = ymax sgn( x)
log e (1 + µ )

where
• x and y represent the input and output voltages
• µ is a constant number determined by experiment
• In the U.S., telephone lines uses companding with µ = 255
▫ Samples 4 kHz speech waveform at 8,000 sample/sec
▫ Encodes each sample with 8 bits, L = 256 quantizer levels
▫ Hence data rate R = 64 kbit/sec
• µ = 0 corresponds to uniform quantization
Delta Modulation (DM): -
In delta modulation (DM), an incoming message signal is over
sampled (i.e. at a rate much higher than the Nyquist rate) to purposely
increase the correlation between adjacent samples of the signal. This is
done to permit the use of a simple quantization strategy for constructing
the encoded signal. In the basic form, DM provides a stair case
approximation to the over sampled version of the message signal, as
shown in figure below the difference between the input and the
approximation is quantized in to only two level, namely,+∆, -∆,
corresponding to positive and negative difference, respectively. Thus, if
the approximation below the signal at any sampling epoch, it is increase
by ∆, on other hand, the approximation lies above the signal, it is
diminished by ∆. We find that the stir case approximation remains +∆, -∆
at the input signal.

Denoting the input signal as m(t), and its stair case approximation as
mq(t), the basic principle of DM may be formalized in the following set of
discrete time relations:-
e[n]=m[n]-mq[n-1] ……..(A)
eq[n]= sgn(e[n]) ……..(B)
 mq[n]=mq[n-1]+eq[n] …….(C)
Where e[n]=error signal representing difference between m[n] and the
latest approximation to it mq[n-1], eq[n]=quantized version of e[n], and
sgn(.) is the segnum function. Finally, the quantizer output e q[n] coded to
produce the DM signal.
DM modulator-demodulator shown below: -

The comparator computes the difference between the inputs. The

quantizer consists of hard limiter with an input/output relation that is
scaled version of the signum function. The accumulator increments the
approximation by a step in positive or negative direction, depending on
the algebraic sign of the error signal e[n]. Demodulation is subjected to
two types of error:-
(1) Slop over load distortion
(2) Granular noise.
Equation (C) may be observed as a digital equavelent of integration in the
sense that it represents the accumulation of positive and negative
increments of magnitude , also, denoting the quantization error by q[n]
as given by
mq[n]=m[n]+q[n]
from equation (A) may be observed
e[n]=m[n]-m[n-1]-q[n-1]

Thus, except for the quantization error q[n-1], the quantizer input is a first
backward difference of the input signal, which may be viewed as a digital
approximation to the derivative of the input signal or, equivalently, as the
inverse of the digital integration process.
If we consider the maximum slope of input m(t), it is clear that the
order for the sequence of samples [mq(nTs)] to increase as fast as the
input sequence [m(nTs)] in a region of maximum slope of m(t), we
required that the conditions

Be satisfied. We find the step size (∆) is too small for the staircase
approximation mq(t) to follow a steep segment of the input signal m(t),
with the result that mq(t) falls behined m(t), as shown in figure below: -

This condition called slope over load, and the resulting quantization
noise called slope over load distortion. DM using a fixed step size (∆) is
often referred to as a linear delta modulation. In contrast to slope over
load distortion, granular noise occurs when the step size ∆ is too large
relative to the local slope characteristics of the input waveform m(t),
thereby causing the staircase approximation mq(t) to hunt around a
relatively flat segment of the input waveform. This noise also illustrated
in above figure. When the step size (∆) is automatically varied depending
on the amplitude characteristics of the analog input signal in this case
called adaptive delta modulation.

Delta-sigma modulation
Differential pulse code modulation (DPCM)
Linear prediction

e
Digital Communication I

Hussain Abdulkarim
Delta Modulation
DPCM,DM,ADM
One Implementation of DPCM
Quantization error is accumulated.
Another Implementation of DPCM
Quantization error is not accumulated.
 Delta Modulation (DM)
- Special type of DPCM with M = 2.
Inexpensive and simple to implement.
DM Waveform
Some notes about DM
Bit rate = sampling rate
Reconstructed signal n
where y(iTs) = +1 or
-1 z (nTs ) = ∑ y (iTs )δ
and δ is the step size. i =1

Types of noise
* Quantization noise: step size δ takes place of smallest
quantization level.
* Granular noise: z(nTs) is always different from z((n-1)Ts).
* Slope overload noise: maximum slope of output signal is
δ / Ts.
δ too small: slope overload noise
too large: quantization noise and granular noise

There is an optimum value for δ in terms of signal

bandwidth,
signal power, and sampling frequency.
Example. Let w(t) = A cos(2πf ot ) and the sampling frequency, f s = kf o
where k is an integer, k ≥ 2. What is the minimum value of δ for no slope overload?

= −2 Af o π sin (2πf ot ) which has the maximum value of 2 Af o π.

dw(t )
dt
1 1
Ts = =
f s kf o
δ 2 Aπ
For no slope overload, ≥ 2 Aπf o . ⇒ δ ≥
Ts k
Adaptive Delta Modulation
Inappropriate step size δ creates noise.
Make δ adaptive.
number of successive 1’s or 0’s step
size
1 δ
2 δ
3 2δ
4 4δ
. .
ADM Block Diagram.
 Speech Coding
- Waveform coders: output approximates original voice
signal.
* PCM, DPCM, DM, CVSD (24 – 64 k bits/s)

- Vocoder: parameterize voice signals based on speech

models
* CELP, VSELP (2 -16 k bits/s)
Digital Communication I

Hussain Abdulkarim
Line Coding
 Line Code
• On the channel, we might want to send binary numbers
directly.
• The resulting bit patterns on the channel might create a
static voltage, which is not desired.
• Use line code to eliminate the average static voltage.
- Save power
- Save bandwidth (possibly)

1 1 1 1 1
5 volt

average
static voltage
0 volt
0 0 0 0 0 0
 Types of Line Code
• Unipolar signaling: 1 = +A volt, 0 = 0 volt
• Polar signaling: 1 = +A volt, 0 = -A volt
• Bipolar signaling: 1 = +A or –A, 0 = 0 volt
(Also called the alternate mark inversion – AMI)
• Machester signaling:
1 = +A (half duration) followed by –A (half duration)
0 = -A (half duration) followed by +A (half duration)

Additional combinations can be made along with RZ

(return to zero) and NRZ (non return to zero).
 Desired Properties of Line Code
• Self synchronization
• Low probability of bit error
• Spectral efficiency
• Low transmission speed
• Error detection capability
• Transparency
 Power Spectral Density for Line Code

N
At the source, wT (t ) = ∑ an f (t − nTb )
n =1

where f (t ) is a symbol pulse.

wT (t ) is the signal observed for 0 < t < T (T = NTb ).
an is data value for the n th symbol.
1
Ps ( f ) = limT → ∞
2
wT (t )
T
 Regenerative Repeater

Suppose for any given bit, Pe = probability that this bit is incorrectly regenerated by a
regenerative repeater. If this bit were to go through a series of m regenerative repeaters,
m
Pi =   Pei (1 − Pe )
m −i
where Pi is the probability that this bit is incorrectely regenerated
i
by i regenerative repeaters. After m regenerative repeaters, this bit will be in error,
if an odd number of errors take place.
m
m i
∑   Pe (1 − Pe ) ≈ mPe
m −i
Pme = probability a bit in error after m regenerative repeaters =
i =1 i
i is odd
 Bit Synchronization
To accurately detect received signals,
synchronization timing is needed.
- derived from received data
- separate signal sent from source

Synchronization
- bit level
- frame level
- carrier level
Binary-to-Multilevel Conversion
 Spectral Efficiency
R
Definition. Spectral efficiency η = bits per second.
B

C  S
By Shannon : ηmax = = log 2 1 + 
B  N

Line Code First Null Bandwidth Spectral Efficiency

(Hz) η=R/B bits/s
Unipolar NRZ R 1
Polar NRZ R 1
Unipolar RZ 2R 0.5
Bipolar RZ R 1
Manchester NRZ 2R 0.5
Multilevel polar NRZ R/l l