Study Digital Communication with Rejin Paul

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EC6501
DIGITAL COMMUNICATION
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OBJECTIVES:
To know the principles of sampling &
quantization
To study the various waveform coding
schemes
To learn the various baseband transmission
schemes
To understand the various Band pass signaling
schemes
To know the fundamentals of channel coding
SYLLABUS www.rejinpaul.com
UNIT I SAMPLING & QUANTIZATION 9
Low pass sampling Aliasing- Signal Reconstruction-Quantization - Uniform & non-uniform
quantization - quantization noise - Logarithmic Companding of speech signal- PCM - TDM 56
UNIT II WAVEFORM CODING 9
Prediction filtering and DPCM - Delta Modulation - ADPCM & ADM principles-Linear Predictive
Coding
UNIT III BASEBAND TRANSMISSION 9
Properties of Line codes- Power Spectral Density of Unipolar / Polar RZ & NRZ Bipolar NRZ Manchester- ISI Nyquist criterion for distortionless transmission Pulse shaping Correlative
coding - Mary schemes Eye pattern - Equalization
UNIT IV DIGITAL MODULATION SCHEME 9
Geometric Representation of signals - Generation, detection, PSD & BER of Coherent BPSK,
BFSK & QPSK - QAM - Carrier Synchronization - structure of Non-coherent Receivers - Principle
of DPSK.
UNIT V ERROR CONTROL CODING 9
Channel coding theorem - Linear Block codes - Hamming codes - Cyclic codes - Convolutional
codes - Vitterbi Decoder
TOTAL: 45 PERIODS
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OUTCOMES
Upon completion of the course, students will be
able to
Design PCM systems
Design and implement base band transmission
schemes
Design and implement band pass signaling
schemes
Analyze the spectral characteristics of band
pass signaling schemes and their noise
performance
Design error control coding schemes
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EC6501
DIGITAL COMMUNICATION
UNIT - 1
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INTRODUCTION
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UNIT I
SAMPLING & QUANTIZATION (9)
Low pass sampling

Aliasing
Signal Reconstruction
Quantization
Uniform & non-uniform quantization
Quantization Noise
Logarithmic Companding of speech signal
PCM
TDM
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Digital communication system

Input
Signal
Analog/
Digital
Low
Pass
Filter
Sampler
Quantizer
Source
Encoder
Channel
Encoder
Multiplexer
Carrier
To Channel
Modulator
Pulse
Shaping
Filters
From Channel
DeModulator
Receiver
Filter
Line
Encoder
Detector
Carrier Ref.
Signal
at the
user end
Digital-to-Analog
Converter
Channel
Decoder
DeMultiplexer
12
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Key Questions
How can a continuous wave form be

converted into discrete samples?
How can discrete samples be converted back

into a continuous form?
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Low Pass Sampling

Sampling (in time) is
Measure amplitude at regular intervals
How many times should we sample?
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Nyquist Theorem
For lossless digitization, the sampling rate
should be at least twice the maximum
frequency of the signal to be sampled.
In mathematical terms:
fs > 2*fm
where fs is sampling frequency and fm is the

maximum frequency in the signal
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Limited Sampling
But what if one cannot sample fast

enough?
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Limited Sampling
Reduce signal frequency to half of

maximum sampling frequency
low-pass filter removes higher-frequencies
(e.g.) If max sampling frequency is 22kHz, the it
is a must to low-pass filter a signal down to
11kHz
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Aliasing effect
LP filter
Nyquist rate
aliasing
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Three different sampling methods
Practical Sampling Methods are Natural Sampling

and Flat-top Sampling
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Natural Sampling
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Pulse-Amplitude Modulation
Pulse-Amplitude Modulation (PAM)

The amplitude of regularly spaced pulses are
varied in proportion to the corresponding sample
values of a continuous message signal.
Two operations involved in the generation of the
PAM signal
Instantaneous sampling of the message signal m(t)
every Ts seconds,
Lengthening the duration of each sample, so that it
occupies some finite value T.
Fig. 5
28
Fig.5
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Fig.6
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Fig.7
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The advantages offered by digital pulse modulation www.rejinpaul.com

Performance
Digital pulse modulation permits the use of regenerative repeaters,
when placed along the transmission path at short enough distances,
can practically eliminate the degrading effects of channel noise and
signal distortion.
Ruggedness
A digital communication system can be designed to withstand the
effects of channel noise and signal distortion
Reliability
Can be made highly reliable by exploiting powerful error-control
coding techniques.
Security
Can be made highly secure by exploiting powerful encryption
algorithms
Efficiency
Inherently more efficient than analog communication system in the
tradeoff between transmission bandwidth and signal-to-noise ratio
System integration
To integrate digitized analog signals with digital computer data
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Quantization Processwww.rejinpaul.com
Amplitude quantization
The process of transforming the sample amplitude m(nTs) of a
baseband signal m(t) at time t=nTs into a discrete amplitude
v(nTs) taken from a finite set of possible levels.
I k : {mk < m mk +1}, k = 1,2,..., L (17)
Fig. 9
Representation level (or Reconstruction level)

The amplitudes vk , k=1,2,3,,L
Quantum (or step-size)
The spacing between two adjacent representation levels
v = g (m ) (18)
Fig. 10
33
Fig.9
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Fig.10
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Two types of quantization are

a) Mid-tread
b) Mid-rise Get useful study materials from www.rejinpaul.com
35
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Linear Quantization
Applicable when the signal is in a
finite range (fmin, fmax)
The entire data range is divided
into L equal intervals of length Q
(known as quantization interval or
quantization step-size)
Q=(fmax-fmin)/L Interval i is
mapped to the middle value of this
interval
We store/send only the index of
quantized value min
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Signal Range is Symmetric
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Quantization Noise
Non-Uniform Quantization
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Many signals such as speech have a nonuniform distribution.
The amplitude is more likely to be close to zero than to be at higher levels.
Nonuniform quantizers have unequally spaced levels
The spacing can be chosen to optimize the SNR for a particular type of signal.
Output sample
XQ
6
4
-8
-6
-4
-2
Example: Nonuniform 3 bit quantizer
2
-2
Input sample
X
-4
-6
39
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Non-Linear Quantization
The quantizing intervals are not of equal size
Small quantizing intervals are allocated to small
signal values (samples) and large quantization
intervals to large samples so that the signal-toquantization distortion ratio is nearly independent of
the signal level
S/N ratios for weak signals are much better but are
slightly less for the stronger signals
Companding is used to quantize signals
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Function representation
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Uniform and Non-uniform Quantization
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Companding
Formed from the words compressing and
expanding.
A PCM compression technique where analogue
signal values are rounded on a non-linear scale.
The data is compressed before sent and then
expanded at the receiving end using the same
non-linear scale.
Companding reduces the noise and crosstalk
levels at the receiver.
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u-LAW and A-LAW definitions

A-law and u-law are companding schemes used in
telephone networks to get more dynamics to the
8 bit samples that is available with linear coding.
Typically 12..14 bit samples (linear scale) sampled
at 8 kHz sample are companded to 8 bit
(logarithmic scale) for transmission over 64 kbit/s
data channel.
In the receiving end the data is then converted
back to linear scale (12..14 bit) and played back.
converted back
Compressor
A particular form of compression law : -law
v=
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log(1 + m )
(5.23)
log(1 + )
d m log(1 + )
(1 + m ) (5.24)
=
dv
-law is neither strictly linear nor strictly logarithmic

A-law :
Am
1
m
,
0
1 + log A
A
(5.25)
v =
1 + log( A m ) , 1 m 1
1 + log A
A
1
1 + log A
,
0
m
d m A
A
=
(5.26)
Fig. 5.11
1
dv
(1 + log A) m , A m 1
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Fig.11
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Example: -law Companding

1
x[n]=speech /song/
0.5
0
-0.5
-1
1000
2000
3000
4000
5000
6000
7000
8000
9000
10000
1000
2000
3000
4000
5000
6000
7000
8000
9000
10000
y[n]=C(x[n])
Companded Signal
0.5
0
-0.5
-1
Segment of x[n]
0.5
0
Close View of the Signal
-0.5
-1
2200
2300
2400
2500
2600
2700
2800
2900
3000
2300
2400
2500
2600
2700
2800
2900
3000
Segment of y[n]
Companded Signal
0.5
0
-0.5
-1
2200
Eeng 360
47
A-law and law Companding

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These two are standard companding methods.

u-Law is used in North America and Japan
A-Law is used elsewhere to compress digital telephone signals
Eeng 360
48
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Quantization - why do we need such classification ?! - (3)
Comparison Uniform Vs. Non-Uniform Usage
Speech signals doesnt require high quantization resolution for
high amplitudes (50% Vs. 15%).
wasteful to use uniform quantizer ?
The goal is decrease the SQNR, more levels for low amplitudes, less levels for
high ones.
Maybe use a Non-uniform quantizer ?
Technical
Presentation
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Concepts
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Quantization
More About Non-Uniform Quantizers (Companding)
Uniform quantizer = use more levels when you need it.
The human ear follows a logarithmic process in which high amplitude sound doesnt
require the same resolution as low amplitude sounds.
One way to achieve non-uniform quantization is to use what is called as Companding
Companding = Compression + Expanding
Compressor
Function
Uniform
Quantization
Expander
Function
(-1)
Technical
Presentation
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Pulse-Code Modulation
PCM (Pulse-Code Modulation)
A message signal is represented by a sequence of coded pulses, which is

accomplished by representing the signal in discrete form in both time and
amplitude
The basic operation
Transmitter : sampling, quantization, encoding
Receiver : regeneration, decoding, reconstruction
Operation in the Transmitter

1. Sampling
1. The incoming message signal is sampled with a train of rectangular pulses
2. The reduction of the continuously varying message signal to a limited
number of discrete values per second
2. Nonuniform Quantization
1. The step size increases as the separation from the origin of the inputoutput amplitude characteristic is increased, the large end-step of the
quantizer can take care of possible excursions of the voice signal into the
large amplitude ranges that occur relatively infrequently.
51
Fig.11
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3. Encoding
1.To translate the discrete set of sample vales to a
more appropriate form of signal
Fig. 11
2.A binary code
The maximum advantage over the effects of noise in a
transmission medium is obtained by using a binary
code, because a binary symbol withstands a relatively
high level of noise.
The binary code is easy to generate and regenerate
Table. 2
53
Regeneration Along the Transmission Path
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The ability to control the effects of distortion and noise produced by

transmitting a PCM signal over a channel
Equalizer
Shapes the received pulses so as to compensate for the effects of
amplitude and phase distortions produced by the transmission
Timing circuitry
Provides a periodic pulse train, derived from the received pulses
Renewed sampling of the equalized pulses
Fig. 13
Decision-making device
The sample so extracted is compared o a predetermined threshold
ideally, except for delay, the regenerated signal is exactly the same as the
information-bearing signal
1. The unavoidable presence of channel noise and interference causes
the repeater to make wrong decisions occasionally, thereby
introducing bit errors into the regenerated signal
2. If the spacing between received pulses deviates from its assigned
value, a jitter is introduced into the regenerated pulse position,
thereby causing distortion.
54
Fig.13
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Operations in the Receivers

1. Decoding and expanding
1.Decoding : regenerating a pulse whose amplitude is
the linear sum of all the pulses in the code word
2.Expander : a subsystem in the receiver with a
characteristic complementary to the compressor
1. The combination of a compressor and an expander is a
compander
2. Reconstruction
1.Recover the message signal : passing the expander
output through a low-pass reconstruction filter
56
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Categories of multiplexing
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Time Division Multiplexing (TDM)
TDM is a technique used for

transmitting several message signals
over a single communication channel
by dividing the time frame into slots,
one slot for each message signal
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Time Division Multiplexing
Entire spectrum is allocated for a channel (user) for a limited time.

The user must not transmit until its
next turn.
k1
k2
k3
k4
k5
Used in 2nd generation
c
k6
Frequency
f
t
Time
Advantages:
Only one carrier in the medium at any given time
High throughput even for many users
Common TX component design, only one power amplifier
Flexible allocation of resources (multiple time slots).
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Disadvantages
Synchronization
Requires terminal to support a much higher data
rate than the user information rate therefore
possible problems with intersymbolinterference.
Application: GSM
GSM handsets transmit data at a rate of 270 kbit/s in

a 200 kHz channel using GMSK modulation.
Each frequency channel is assigned 8 users, each
having a basic data rate of around 13 kbit/s

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At the Transmitter
Simultaneous transmission of several signals on a time-sharing basis.
Each signal occupies its own distinct time slot, using all frequencies, for
the duration of the transmission.
Slots may be permanently assigned on demand.
At the Receiver
Decommutator (sampler) has to be synchronized with the incoming
waveform Frame Synchronization
Low pass filter
ISI poor channel filtering
Feedthrough of one channel's signal into another channel -- Crosstalk
Applications of TDM: Digital Telephony, Data communications, Satellite
Access, Cellular radio.
61

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Conceptual diagram of multiplexing-demultiplexing.
PAM TDM System
62
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TDM-PAM: Transmitter
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TDM-PAM : Receiver
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Samples of Signal -1
g1(t)
time
0
Ts
2Ts
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Samples of signal - 2
g2(t)
Ts
Ts
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Multiplexing of TWO signals
Ts
2Ts
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TDM-PAM for 4 signals.
4
4
4
1
1
2
1
2
2
3
Time
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Problem
Two low-pass signals of equal bandwidth
are sampled and time division
multiplexed using PAM. The TDM signal is
passed through a Low-pass filter & then
transmitted over a channel with a
bandwidth of 10KHz.
Continued.
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Problem (continued)
a) What is maximum Sampling rate for each
Channel?
b) What is the maximum frequency content
allowable for each signal?
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Problem: Solution
Channel Bandwidth = 10 KHz.
Number of samples that can be transmitted
through the channel = 20K
Maximum Sampling rate for each channel =
10K Samples/sec.
Maximum Frequency for each Signal = 5KHz
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End of Unit-1
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Unit II
Waveform Coding
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Syllabus
Prediction filtering and DPCM - Delta
Modulation - ADPCM & ADM principles-Linear
Predictive Coding
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Introduction to Waveform Coding

Waveform coding is some kind of approximately
lossless coding, as it deals with speech signal as
any kind of ordinary data.
The resulting signal is close as possible as the
original one.
Codecs using this techniques have generally low
complexity and give high quality at rates 16 Kbps.
The simplest form of waveform coding is Pulse
Code Modulation (PCM).
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Pulse Code Modulation (PCM)

It involves sampling and quantizing the input
waveform.
PCM consists of three steps to digitize an
analog signal:
1. Sampling
2. Quantization
3. Binary encoding
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Prediction Filtering
Linear prediction is a mathematical operation
where future values of a discrete-time signal
are estimated as a linear function of previous
samples.
In digital signal processing, linear prediction is
often called linear predictive coding (LPC).
linear prediction can be viewed as a part of
mathematical modelling or optimization.
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The Prediction Model

The most common representation is
Where
is the predicted signal value, x(n-i)
the previous observed values, and the
predictor coefficients.
The error generated by this estimate is
Where x(n) is the true value.
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Differential pulse-code modulation

(DPCM)
Differential pulse-code modulation (DPCM) is

a signal encoder that uses the baseline of
pulse-code modulation (PCM) but adds some
functionalities based on the prediction of the
samples of the signal.
The input can be an analog signal or a digital
signal.
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Block Diagram of DPCM
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DPCM code words represent differences between samples

unlike PCM where code words represented a sample value.
Basic concept of DPCM - coding a difference, is based on
the fact that most source signals show significant
correlation between successive samples so encoding uses
redundancy in sample values which implies lower bit rate.
Realization of basic concept (described above) is based on a
technique in which we have to predict current sample value
based upon previous samples (or sample) and we have to
encode the difference between actual value of sample and
predicted value.
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Principle of DPCM
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Delta Modulation
A Delta modulation (DM or -modulation) is an analog-to-digital
and digital-to-analog signal conversion technique used for
transmission of voice information where quality is not of primary
importance.
To achieve high signal-to-noise ratio, delta modulation must use
oversampling techniques, that is, the analog signal is sampled at a
rate several times higher than the Nyquist rate.
Derived forms of delta modulation are continuously variable slope
delta modulation, delta-sigma modulation, and differential
modulation.
Differential pulse-code modulation is the super-set of DM.
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Features
the analog signal is approximated with a series of
segments
each segment of the approximated signal is compared to
the original analog wave to determine the increase or
decrease in relative amplitude
the decision process for establishing the state of
successive bits is determined by this comparison
only the change of information is sent, that is, only an
increase or decrease of the signal amplitude from the
previous sample is sent whereas a no-change condition
causes the modulated signal to remain at the same 0 or 1
state of the previous sample.
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Principle of delta modulation
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Differential Pulse Code Modulation

(DPCM)
What if we look at sample differences, not the
samples themselves?
dt = xt-xt-1
Differences tend to be smaller
Use 4 bits instead of 12, maybe?
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Differential Pulse Code Modulation

(DPCM)
Changes between adjacent samples small
Send value, then relative changes
value uses full bits, changes use fewer bits
E.g., 220, 218, 221, 219, 220, 221, 222, 218,.. (all values between 218
and 222)
Difference sequence sent: 220, +2, -3, 2, -1, -1, -1, +4....
Result: originally for encoding sequence 0..255 numbers need 8 bits;
Difference coding: need only 3 bits
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Adaptive Differential Pulse Code

Modulation (ADPCM)
Adaptive similar to DPCM, but adjusts the width of the
quantization steps
Encode difference in 4 bits, but vary the mapping of bits to
difference dynamically
If rapid change, use large differences
If slow change, use small differences
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Adaptive Delta Modulation (ADM)
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A large step size is required when sampling those parts

of the input waveform of steep slope. But a large
step size worsens the granularity of the sampled
signal when the waveform being sampled is changing
slowly.
A small step size is preferred in regions where the
message has a small slope. This suggests the
need for a controllable step size the control
being sensitive to the slope of the sampled signal
Hence ADM is prefered.
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Adaptive Delta Modulation
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VCA
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Linear Predictive Coding (LPC)
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Basic Concepts of LPC

It is a parametric de-convolution algorithm
x(n) is generated by an unknown sequence e(n)
exciting a unknown system V(Z) which is supposed to
be a linear non time-variant system.
V(Z) = G(Z)/A(Z),
E(Z)V(Z) = X(Z)
G(Z) = j=0Q gjZ-j, A(Z) = i=0P aiZ-i
Where ai and gj are parameters, real and a0 = 1
If an algorithm could estimate all these parameters,
then V(Z) could be found, and E(Z) could be found
also. This finishes de-convolution.
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There are some limitations for the model

(1) G(Z) = 1 then V(Z) = 1/A(Z) this is so called Full
Poles odels and the parametric de-convolution
became coefficients(ai) estimation problem.
(2) e(n) sequence is of form Ge(n), where e(n) is a
periodic pulse or a Gaussian white noise sequence.
For the first case e(n) = (n-rNp) and for the second
case R(k) = E[e(n)e(n+k)] = (k) and the value of
e(n) satisfied with Normal distribution. G is a nonnegative real number controlling the amplitude.
The way is x(n)->V(Z)(P,ai)->e(n),G->type of e(n)
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Suppose x(n) and type of e(n) are known, what is

the optimized estimation of P and ai, e(n) and G? It is
the LMS algorithm.
Suppose x(n) is the predicted value of x(n), it is the
linear sum of previous P known values of x:
x(n) = i=1P ai x(n-i)
The predicted error
(n) = x(n)-x(n) = x(n) - i=1P ai x(n-i)
It is a stochastic sequence. The variance of it could
be used to evaluate the quality of prediction.
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2 = n 2(n) (time average replaced means)

It could be proved that if x(n) is generated by full
poles model : x(n) = -i=1P ai x(n-i) + Ge(n) and
optimized P = P, optimized ai = ai, 2 is minimal.
2 = n [x(n) -i=1P ai x(n-i)]2
={n x2(n)}-2i=1P ak{n x(n-k)x(n)}+
k=1Pi=1P akai{n x(n-k)x(n-i)}

By setting (2 )/ ak = 0 we can get
-2 {n x(n-k)x(n)}+2i=1P ai{n x(n-k)x(n-i)}=0
Or i=1P ai(k,i) = (k,0)
if (k,i) =n x(n-k)x(n-i) 1<=i<=P and 1<=k<=P
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i=1P ai(k,i) = (k,0), k=1~P is called PC

canonical equations. There are some different
algorithms to deal with the solution.
[2]min = k=0P ak (k,0) with a0 = 1
So if we have x(n), (k,i) could be calculated,
and equations could be solved to get ai and
[2]min also could be obtained. For short-time
speech signal according to different lower and
upper limitation of the summary we could
have different types of equations. We will
discuss these different algorithms later.
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Auto-Correlated Solution of LPC

Suppose windowed signal is xw(n)
(k,i) = n xw(n-k)xw(n-i)
If window length is N-1 then the summation
range will be 0~N+P-1
(k,i) = m xw(m+(i-k))xw(m) = R(i-k) if n-i
=m
(k,i) = R(i-k) = R(k-i) = R(|i-k|) <= R(0)
The equations became i=1P aiR(|i-k|)= - R(k)
1<=k<=P
These are Toplitz equations and have high

efficient solution.
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|R(0) R(1) R(P-1)|

|R(1) R(0) R(P-2)|
|.|
|R(P-1) R(P-2) R(0) |
6.2.1 Durbin Algorithm

1. E(0) = R(0)
2. Ki = [ R(i) - aj(i-1)R(i-j)]/E(i-1)
1<=i<=p
3. ai(i) = Ki
4. aj(i) = aj(i-1) Kiai-j(i-j)
1<=j<=i-1
5. E(i) = (1-Ki2)E(i-1)
Final solution is aj = aj(p)
1<=j<=p
|a1|
|a2|
|...|
|ap|
| R(1) |
| R(2) |
=
...
| R(P) |
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For iteration i we got a set of

coefficients for the predictor of i-th
order and the minimal predicted error
energy E(i). We also can get it by {R(k)}
:
E(i) = R(0) k=1i akR(k), 1<=i<=p
Ki is the reflect coefficient : -1<=Ki<=1
It is a sufficient and necessary condition
for stable H(z)
during
iteration.
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6.2.2 Schur algorithm

At first an auxilary sequence is defined. Its properties
are :
(1) qi(j) = R(j) when i = 0
(2) qi(j) = 0
when i > 0, j=1~p
(3) qp(0) = E(p) is the predicted error energy.
(4) |qi(j)| <= R(0), it is equal only if i=j=0
The algorithm is as following:
1. r(j) = R(j)/R(0), r(-j) = r(j), j=0~p
2. a0 = 1, E(0) = 1
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3. q0(j) = r(j) Get -p<j<p
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4. i = 1, k1 = r(1)
5. For i-p<=j<=p
qi(j) = qi-1(j) + ki *qi-1 (i-j)
ki = qi-1(j)/qi(0)
aj(i) = qi-1(i-j)
E(i) = E(i-1)(1-ki2)
6. If i<p, back to step 5
7. Stop
If we only calculate ki, then only first two expressions
in step 5 are enough. It is suitable for fix-point
calculation (r<=1) or hardware implementation.
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Covariance Solution of LPC

If not using windowing, but limiting the range of
summation, we could get :
2 = n=0N-1 2(n) n=0~N-1
(k,i) = n=0N-1 x(n-k)x(n-i)
k=1~p, i=0~p
= m=-iN-i-1 x(m+(i-k))x(m) let n-i=m, m=-i~N-i-1
The equations will be like following :
|(1,1) (1,2) (1,p)| |a1|
|(1,0)|
|(2,1) (2,2) (2,p)| |a2|
|(2,0)|
.=
|(p,1) (p,2) (p,p)| |ap|
|(p,0)|
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Covariance Solution of LPC

The matrix is a covariance matrix and it is positive
determined, but not Toplitz. There is no high efficient
algorithm to solve. Only common used LU algorithm
could be applied. Its advantage is not having big
predicted error on the two ends of window. So when
N~P the estimated parameters have more accuracy
than auto-correlated method. But in speech
processing very often N>>P, so the advantage is not
obvious.
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LPC parameters and their relationships
(1) Reflect Coefficients

Also known as PARCOR coefficients
If {aj} are known, ki could be found as following :
1<=j<=p
aj(p) = aj
ki = ai(i)
aj(i-1) = (aj(i) + aj(i) ai-j(i))/(1-ki2) 1<=j<=i-1
The inverse process :
aj(i) = ki
aj(i) = aj(I-1) - kj ai-j(i-1) at last aj= aj(p) 1<=j<=p
-1<=ki<=1 is the sufficient and necessary condition
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(2) Coefficients of Logarithm Area Ratio

gi = log(Ai+1/Ai) = log[(1-ki)/1+ki]) i=1~p
Where A is the intersection area of i-th
segment of the lossless tube.
i=1~p
ki = (1-exp(gi))/(1+exp(gi))
(3) Cepstrum Coefficients
cn = an + k=1n kckan-k/n, 1<=n<=p+1
= an + k=n-pn-1 kckan-k/n, n>p+1
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(4) The Roots of Predictor

A(z) = 1 k=1p akz-k = k=1p (1-zk z-1) = 0
Transfer to S-plane: zi = exp(siT)
Suppose si = i + ji , zi = zir + jzii , then
i = tan(zii/zir)/T andi= log(zii2 + zir2 )/(2T)
(5)The impulse response of full poles system

h(n) = k=1p akh(n-k)+(n) n>=0
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(6) Auto-correlated Coefficients of impulse

response of the full poles system
H(z) = S(z)/U(z) = G/(1- k=1p akz-k)
The auto-correlated coefficients of h(n) is :
R(i) = n=0 h(n)h(n-i) = R(-i)
It could be proved that :
R(i) = k=1p ak R(|i-k|) 1<=i<=p
And R(0) = k=0p ak R(k) + G2
{ak} -> {R(i)} and {R(i)} -> {ak} are
equivalent
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(7) Auto-correlated coefficients of impulse
response of the predicted error filter (inverse

filter)
A(z) = 1 - akz-k
The impulse response is :
a(n) = (n) - k=1p ak (n-k)
= 1, n = 0; an , 0<n<=p; 0, otherwise
Its auto-correlated function is :
R(i) = k=1p a(k)a(k+i)
0<=i<=p
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(8)Line Spectrum Pair (LSP) or Line

Spectrum Frequency (LSF)
A(p)(z)=1-k=1p akz-k ( p is even)
Define P(z) = A(p)(z)+z-(p+1)A(p)(z-1)
Q(z) = A(p)(z)- z-(p+1)A(p)(z-1)
It could be proved that : All roots of P(z) and
Q(z) are on the unit circle and alternatively
arranged on it provided the roots of A(z) are
inside the unit
circle.
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Replace z with expj:

P(expj)=|A(P)(expj)|expj()[1+exp[-j((p+1)+
2())]
Q(expj)=|A(P)(expj)|expj()[1+exp[-j((p+1)+
2()+)]
If the roots of A(P)(z) are inside the unit circle, whenis
0~, () changes from 0 and returns to 0, the amount
[(p+1)+2()] will be 0~(p+1)
P(expj)=0 : [(p+1)+2()]=k, k=1,3,P+1
Q(expj)=0 : [(p+1)+2()]=k, k=0,2,P
The roots of P and Q : Zk = expjk [(p+1)+2()]=k,
k=0,1,2,P+1
And 0 < 1 < 2 < < P < P+1
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If a1(p)~ap(p) are known, LSP could be found by

A(z) -> P(z) -> p(w) -> f1, f2, fp
If f1~fp are known, ai(p) could be found by P(z)
and Q(z) -> A(z) = P(z) + Q(z) -> a1(p)~ap(p)
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Thank you
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Unit 3
Baseband Transmission
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Syllabus
Properties of Line codes- Power Spectral Density of Unipolar /
Polar RZ & NRZ Bipolar NRZ - Manchester- ISI Nyquist
criterion for distortionless transmission Pulse shaping
Correlative coding - Mary schemes Eye pattern - Equalization
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Baseband Transmission
The digital signal used in baseband
transmission occupies the entire bandwidth
of the network media to transmit a single data
signal.
Baseband communication is bidirectional,
allowing computers to both send and receive
data using a single cable.
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Baseband Modulation
An information bearing-signal must conform to the limits of its channel

Generally modulation is a two-step process
baseband: shaping the spectrum of input bits to fit in a limited spectrum
passband: modulating the baseband signal to the system rf carrier
Most common baseband modulation is Pulse Amplitude Modulation
(PAM)
data amplitude modulates a sequence of time translates of basic pulse
PAM is a linear form of modulation: easy to equalize, BW is pulse BW
Typically baseband data will modulate in-phase [cos] and quadrature
[sine] data
streams to the carrier passband
Special cases of modulated PAM include
phase shift keying (PSK)
quadrature amplitude modulation (QAM)
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Need for Baseband Modulation
An analog signal has a finite bandwidth.

A digital stream or signal, with sharp transitions, has an infinite
bandwidth.
Due to the limited available system bandwidth, only the major
portion of
a digital signal spectrum can be transmitted and restored. Even if
there is
no loss or noise in the communication system, the received signal
will
have distortion due to the limited channel bandwidth.
To avoid or to reduce this signal distortion, we
use baseband modulation techniques
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Line Codes
In telecommunication, a line code (also called digital baseband
modulation or digital baseband transmission method) is a code
chosen for use within a communications system for baseband
transmission purposes.
Line coding is often used for digital data transport.
Line coding consists of representing the digital signal to be
transported by an amplitude- and time-discrete signal that is
optimally tuned for the specific properties of the physical channel
(and of the receiving equipment).
The waveform pattern of voltage or current used to represent the
1s and 0s of a digital data on a transmission link is called line
encoding.
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Common types of Line Codes
The common types of line encoding are

unipolar
polar
bipolar
Manchester encoding
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Need for Line Codes
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Properties of Line Codes
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Unipolar coding
Unipolar encoding is a line code. A positive voltage represents a
binary 1, and zero volts indicates a binary 0. It is the simplest line
code, directly encoding the bitstream, and is analogous to on-off
keying in modulation.
Its drawbacks are that it is not self-clocking and it has a significant
DC component, which can be halved by using return-to-zero, where
the signal returns to zero in the middle of the bit period.
With a 50% duty cycle each rectangular pulse is only at a positive
voltage for half of the bit period.
This is ideal if one symbol is sent much more often than the other
and power considerations are necessary, and also makes the signal
self-clocking.
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NRZ(Non-Return-to-Zero) - Traditionally, a unipolar scheme was

designed as a non-return-to-zero (NRZ) scheme, in which the positive
voltage defines bit 1 and the negative voltage defines bit 0.
It is called NRZ because the signal does not return to zero at the
middle of the bit.
Compared with its polar counterpart, Uni Polar NRZ, this scheme is
very expensive.
The normalized power (power required to send 1 bit per unit line
resistance) is double that for polar NRZ.
For this reason, this scheme is not normally used in data
communications today.
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Return-to-zero
Return-to-zero (RZ) describes a line code used in
telecommunications signals in which the signal
drops (returns) to zero between each pulse.
This takes place even if a number of consecutive
0s or 1s occur in the signal.
The signal is self-clocking. This means that a
separate clock does not need to be sent
alongside the signal, but suffers from using twice
the bandwidth to achieve the same data-rate as
compared to non-return-to-zero format.
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Polar RZ
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BiPolar Signalling
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Manchester Encoding
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Inter Symbol Interference (ISI)

In
telecommunication,
inter
symbol
interference (ISI) is a form of distortion of
a signal in which one symbol interferes with
subsequent symbols.
This is an unwanted phenomenon as the previous
symbols have similar effect as noise, thus making
the communication less reliable.
ISI is usually caused by multipath propagation or
the inherent non-linear frequency response of
a channel causing successive symbols to "blur"
together.
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Nyquist criterion for distorionless

transmission
To design
under the following two conditions:
(a). There is no ISI at the sampling instants (Nyquist criterion).

(b). A controlled amount of ISI is allowed (correlative coding)
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Design of Bandlimited Signals for Zero

ISI - Nyquist criterion
Recall the output of the receiving filter,
sampled at t = kT, is given by
Thus, in time domain, a sufficient condition

for p(t) such that it is ISI free is
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Theorem: (Nyquist) A necessary and and sufficient condition

for p(t) to satisfy (1) is that the Fourier transform P(f) satisfies
This is known as the Nyquist pulse-shaping criterion or

Nyquist condition for zero ISI.
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Investigate possible pulses which satisfy the

Nyquist criterion
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Disadvantages:
(a) an ideal LPF is not physically realizable.
(b) Note that
Thus, the rate of convergence to zero is low since the

tails of p(t) decay as 1/|t|.
Hence, a small mistiming error in sampling the
output of the matched filter at the demodulator
results in an infinite series of ISI components.
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Correlative Coding and Equalization

Correlative Coding
For zero ISI, the symbol rate R = 1/T < 2W, the
Nyquist rate.
We may relax the condition of zero ISI in order to
achieve R = 2W.
The schemes which allow a controlled amount

of ISI to achieve the symbol rate 2W are called
correlative coding or partial response signaling
schemes.
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2nd Nyquist Criterion

Values at the pulse edge are distortionless
p(t) =0.5, when t= -T/2 or T/2; p(t)=0, when t=(2k-1)T/2, k ,
where - /T f /T
Pr ( f ) Re[ (1) n P ( f n / T )] T cos( fT / 2)

n
PI ( f ) Im[ ( 1) n P ( f n / T )] 0
n
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3rd Nyquist Criterion

Within each symbol period, the integration of signal (area) is proportional to
the integration of the transmit signal (area)
( wt ) / 2
,
T
P ( w) sin( wT / 2)
0,
w
p(t )
/T
( wt / 2)
/ T
sin (wT / 2)
2 n1T
2
A 2 n1
2
1,
p(t )dt
0,
e jwt d w
n0
n0
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Eye Diagram
Eye diagram is a means of evaluating the quality of a received
digital a eform
By quality is meant the ability to correctly recover symbols and timing

The received signal could be examined at the input to a digital receiver
or at some stage within the receiver before the decision stage
Eye diagrams reveal the impact of ISI and noise

Two major issues are 1) sample value variation, and 2) jitter
and sensitivity of sampling instant
Eye diagram reveals issues of both
Eye diagram can also give an estimate of achievable BER
Check eye diagrams at the end of class for participation
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Interpretation of Eye Diagram
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Raised Cosine Eye Diagram

The larger , the wider the
opening.
The larger , the larger
bandwidth (1+ )/Tb
But smaller will lead to
larger errors if not sampled
at the best sampling time
which occurs at the center
of the eye.
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Cosine rolloff filter: Eye pattern

2nd Nyquist
2nd Nyquist:
1st Nyquist:
2nd Nyquist:
1st Nyquist:
1st Nyquist
2nd Nyquist:
1st Nyquist:
2nd Nyquist:
1st Nyquist:
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Eye Diagram Setup

Eye diagram is a retrace display of
data waveform
Data waveform is applied to
input channel
Scope is triggered by data
clock
Horizontal span is set to cover
2-3 symbol intervals
Measurement of eye opening is
performed to estimate BER
BER is reduced because of
additive interference and noise
Sampling also impacted by
jitter
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Partial Response Signals

Previous classes: Sy(w)=|P(w)|^2 Sx(w)
Control signal generation methods to reduce Sx(w)

Raise Cosine function for better |P(w)|^2
This class: improve the bandwidth efficiency

Widen the pulse, the smaller the bandwidth.
But there is ISI. For binary case with two symbols, there is
only few possible interference patterns.
By adding ISI in a controlled manner, it is possible to achieve
a signaling rate equal to the Nyquist rate (2W symbols/sec)
in a channel of bandwidth W Hertz.
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Thank you
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UNIT IV
DIGITAL MODULATION SCHEME
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Geometric Representation of
Signals
Objective: To represent any set of M energy

signals {si(t)} as linear combinations of N
orthogonal basis functions, where N M
Real value energy signals s1(t), s2(t),..sM(t),
each of duration TOrthogonal
sec basis
function
si (t ) sij j (t ),
j 1
0 t T
i==1,2,....,M
4.1
(5.5)
coefficient
Energy signal
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Coefficients:
T
sij si (t ) j (t )dt ,
0
i=1,2,....,M
j=1,2,....,M
(5.6)
Real-valued basis functions:
1 if i j
0 i (t ) j (t )dt ij 0 if i j
T
(5.7)
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The set of coefficients can be viewed as a

N-dimensional vector, denoted by si
Bears a one-to-one relationship with the
transmitted signal si(t)
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(a) Synthesizer for generating the signal si(t). (b) Analyzer for
generating the set of signal vectors si.
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So,
Each signal in the set si(t) is completely
determined by the vector of its coefficients
si1
s
i2
.
si ,
.
.

siN
i 1,2,....,M
(5.8)
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Finally,
The signal vector si concept can be extended to 2D, 3D
etc. N-dimensional Euclidian space
Provides mathematical basis for the geometric
representation of energy signals that is used in noise
analysis
Allows definition of
Length of vectors (absolute value)
Angles between vectors
Squared value (inner product of si with itself)
si
siT si
N
= sij2 ,
Matrix
Transposition
i 1,2,....,M
(5.9)
j 1
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Figure 5.4
Illustrating the geometric
representation of signals
for the case when N 2
and M 3.
(two dimensional space,
three signals)
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Also,
What is the relation between the vector
representation of a signal and its energy value?
start with the
definition of average
energy in a signal
Where si(t) is
Ei si2 (t )dt
(5.10)
si (t ) sij j (t ), (5.5)
j 1
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N
N
After substitution: Ei sij j (t ) sikk (t ) dt
k 1
0 j 1
T
After regrouping: Ei
j 1
s s (t ) (t )dt
k 1
ij ik
(5.11)
N
j(t) is orthogonal, so
2
E
i
ij = s i
finally we have:
j 1
The energy of a
signal is equal to the
squared length of its
vector
(5.12)
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Formulas for two signals

Assume we have a pair of signals: si(t) and sj(t),
each represented by its vector,
Then:
T
sij si (t )sk (t )dt s s

0
Inner product of the

signals is equal to the
inner product of their
vector representations
[0,T]
T
i k
(5.13)
Inner product is invariant

to the selection of basis
functions
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Euclidian Distance
The Euclidean distance between two points
represented by vectors (signal vectors) is equal
to
||si-sk|| and the squared value is given by:
si s k
= (sij -skj ) 2
(5.14)
j 1
T
= ( si (t ) sk (t ))2 dt
0
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BPSK - BINARY PHASE

SHIFT KEYING
Generation of BPSK:
Consider a sinusoidal carrier. If it is modulated
by a bi-polar bit stream, its polarity will be
reversed every time the bit stream changes
polarity.
This, for a sinewave, is equivalent to a phase
reversal (shift). The multiplier output is a BPSK 1
signal.
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BPSK signal in time domain

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sychronous demodulation of BPSK
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Frequency Shift Keying (FSK)

Binary FSK
Frequency of the constant amplitude carrier is changed
according to the message state high (1) or low (0)
s1 (t ) A cos(2f c 2f )t 0 t Tb (bit 1)
s2 (t ) A cos(2f c 2f )t 0 t Tb (bit 0)
Discontinuous / Continuous Phase
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Discontinuous Phase FSK

Switching between 2 independent oscillators for binary 1
&0
input data
phase jumps
cos w1t
switch
cos w2t
binary 1
sBFSK(t)= vH(t)
=
binary 0
2 Eb
cos(2f H t 1 )
Tb
0 t Tb
sBFSK(t)= vL(t)
=
2 Eb
cos(2f Lt 2 ) 0 t Tb
Tb
results in phase discontinuities

discontinuities causes spectral spreading & spurious transmission
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Continuous Phase FSK

single carrier that is frequency modulated using m(t)
2 Eb
sBFSK(t) =
cos(2f ct (t ))
Tb
=
2 Eb
cos 2f c t 2k FSK m( )d
Tb
where (t) =
2k FSK m( )d
m(t) = discontinuous bit stream

(t) = continuous phase function proportional to integral of m(t)
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FSK Example
Data
1
FSK
Signal
x
a0
a1
0
1
VCO
modulated composite
signal
cos wct
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Spectrum & Bandwidth of BFSK Signals

complex envelope of BFSK is nonlinear function of m(t)
spectrum evaluation - difficult - performed using actual time
averaged measurements
PSD of BFSK consists of discrete frequency components at
fc
fc nf , n is an integer
PSD decay rate (inversely proportional to spectrum)
PSD decay rate for CP-BFSK
1
f 4
PSD decay rate for non CP-BFSK
f = frequency offset from fc
1
f 2
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Spectrum & Bandwidth of BFSK Signals

Transmission Bandwidth of BFSK Signals (from Carsons Rule)
B = bandwidth of digital baseband signal
BT = transmission bandwidth of BFSK signal
BT = 2f +2B
assume 1st null bandwidth used for digital signal, B
- bandwidth for rectangular pulses is given by B = Rb
- bandwidth of BFSK using rectangular pulse becomes
BT = 2(f + Rb)
if RC pulse shaping used, bandwidth reduced to:
BT = 2f +(1+) Rb
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General FSK signal and orthogonality

Two FSK signals, VH(t) and VL(t) are orthogonal if
T
VH (t )VL (t )dt 0
interference between VH(t) and VL(t) will average to 0 during

demodulation and integration of received symbol
received signal will contain VH(t) and VL(t)
demodulation of VH(t) results in (VH(t) + VL(t))VH(t)
T
VH (t )VL (t )dt 0
(t )VH (t )dt 0
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An FSK signal for 0 t Tb

vH(t) =
then
and
2 Eb
cos( 2 ( f c f )t )
Tb
Tb
Eb
0 Tb cos(4f ct ) cos(4ft )dt
Tb
2 Eb
cos( 2 ( f c f )t )
Tb
2Eb
cos( 2 ( f c f )t ) cos( 2 ( f c f )t )
Tb
Eb
=
cos(2 (2 f c )t ) cos(2 (2f )t )
Tb
vH(t) vL(t) =
VH (t )VL (t )dt
0
and vL(t) =
Eb sin( 4f c t ) sin( 4ft )
Tb 4f c
4f 0
Eb sin( 4f cTb ) sin( 4fTb )
Tb
4f c
4f
vH(t) vL(t) are orthogonal if f sin(4fcTb) = -fc(sin(4f Tb)

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QPSK
Quadrature Phase Shift Keying (QPSK)
can be interpreted as two independent
BPSK systems (one on the I-channel
and one on Q-channel), and thus the
same performance but twice the
bandwidth (spectrum) efficiency.
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QPSK Constellation Diagram

Q
Carrier phases
{0, /2, , 3/2}
Carrier phases
{/4, 3/4, 5/4, 7/4}
Quadrature Phase Shift Keying has twice the

bandwidth efficiency of BPSK since 2 bits are
transmitted in a single modulation symbol
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Types of QPSK
Q
Conventional QPSK
Offset QPSK
/4 QPSK
Conventional QPSK has transitions through zero (i.e. 1800 phase

transition). Highly linear amplifiers required.
In Offset QPSK, the phase transitions are limited to 900, the transitions on
the I and Q channels are staggered.
In /4 QPSK the set of constellation points are toggled each symbol, so
transitions through zero cannot occur. This scheme produces the lowest
envelope variations.
All QPSK schemes require linear power amplifiers
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Quadrature Phase Shift Keying (QPSK):
Also a type of linear modulation scheme
Quadrature Phase Shift Keying (QPSK) has twice the bandwidth efficiency of
BPSK, since 2 bits are transmitted in a single modulation symbol.
The phase of the carrier takes on 1 of 4 equally spaced values, such as
where each value of phase corresponds to a unique pair of message bits.
The QPSK signal for this set of symbol states may be defined as:
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QPSK
The striking result is that the bit error probability of QPSK is identical to
BPSK, but twice as much data can be sent in the same bandwidth. Thus,
when compared to BPSK, QPSK provides twice the spectral efficiency with
exactly the same energy efficiency.
Similar to BPSK, QPSK can also be differentially encoded to allow noncoherent detection.
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Quadrature amplitude modulation
Quadrature amplitude modulation (QAM) is both an

analog and a digital modulation scheme.
It conveys two analog message signals, or two
digital bit streams, by changing (modulating)
the amplitudes of two carrier waves, using
the amplitude-shift keying(ASK) digital modulation
scheme or amplitude modulation (AM) analog
modulation scheme.
The two carrier waves, usually sinusoids, are out of

phase with each other by 90 and are thus
called quadrature carriers or quadrature components
hence the name of the scheme.
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QAM Transmitter
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First the flow of bits to be transmitted is split

into two equal parts: this process generates
two independent signals to be transmitted.
They are encoded separately just like they
were in an amplitude-shift keying (ASK)
modulator.
Then one channel (the one "in phase") is
multiplied by a cosine, while the other
channel (in "quadrature") is multiplied by a
sine.
This way there is a phase of 90 between
them. They are simply added one to the other
and sent through
the
real
channel.
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study
materials
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QAM Receiver
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The receiver simply performs the inverse

operation of the transmitter.
Multiplying by a cosine (or a sine) and by a
low-pass filter it is possible to extract the
component in phase (or in quadrature).
Then there is only an ASK demodulator
and the two flows of data are merged
back.
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Carrier Synchronization
Synchronization is one of the most critical
functions of a communication system with
coherent receiver. To some extent, it is the
basis of a synchronous communication
system.
Carrier synchronization
Symbol/Bit synchronization
Frame synchronization
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Receiver needs estimate and compensate for frequency and phase

differences between a received signals carrier wave and the
receivers local oscillator for the purpose of coherent demodulation,
no matter it is analog or digital communication systems.
To extract the carrier
1. Pilot-tone insertion method
Sending a carrier component at specific spectral-line along with

the signal component. Since the inserted carrier component has
high frequency stability, it is called pilot.
2. Direct extraction method
Directly extract the synchronization information from the

received signal component.
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1. Pilot-tone insertion method

insert pilot to the modulated signal
x(t)
Modulator
Bandpass
filter
Add
s(t)
-asin(ct)
cos(ct)
/2phase
shift
The pilot signal is generated by shift the

carrier by 900 and decrease by several dB,
then add to the modulated signal. Assume
the modulated signal has 0 DC component,
then the pilots is
t f t cos c t a sin ct
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2. Direct extraction method

If the spectrum of the received signal
already contains carrier component, then
the carrier component can be extracted
simply by a narrowband filter or a PLL.
If the modulated signal supresses the
carrier component, then the carrier
component may be extracted by
performing nonlinear transformation or
using a PLL with specific design
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DPSK
DPSK is a kind of phase shift keying which
avoids the need for a coherent reference
signal at the receiver.
Differential BPSK
0 = same phase as last signal element
1 = 180 shift from last signal element
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DPSK modulation and

demodulation
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Thank you
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Unit - 5
Error Controlling Codes
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ENTROPY
The e t op of a sou e is defi ed as the sou e hi h p odu es a e age
i fo atio pe i di idual essage o s
ol i a pa ti ula i te al .
Then the number of messages is given as
The amount of information in message m1 is given as
The total amount of information
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Co t
Thus the total amount of information due to L
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Co t
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PROPERTIES OF ENTROPY
The entropy of a discrete memoryless channel
source
Property 1
1. Entropy is zero, if the event is sure or its impossible
Co t
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Property 2
2. Entropy H= log2 K when the symbols are equally likely for K
symbols PK= 1/K
Co t
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Property 3
3. Maximum upper bound on entropy is
Co t
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Co t
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Cont..
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RATE OF INFORMATION
RATE OF INFORMATION
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SOURCE CODING
An important problem in communication system is the
efficient representation of data generated by a source, which
can be achieved by source encoding (or) source coding
process
The device which performs source encoding is called source
encoder
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EFFICIENCY OF SOURCE ENCODER

What are the requirements that a source
encoder should satisfy for it to be efficient--------- Binary codewords & Discrete memory-less
source
What are the advantages of source coding?
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NOISELESS CODING THEOREM

Shanons information theory for a discrete memoryless
sources and channels involves three theorems.
(i) Shanons first theorem (or) Source coding theorem
(ii) Shanons second theorem (or) channel coding theorem (or)
shanons theorem on channel capacity
(iii) Shanons third theorem (or) information capacity theorem
(or) shanon hartley theorem.
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SHANONS FIRST THEOREM
This theorem is also called as source coding theorem. Consider a discrete

memoryless source
Encoding process
The average code word length L is given as
The coding efficiency of the source encoder is
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STATEMENT
Shanon first theorem is stated as Given a discrete memoryless source of
entropy H, the average codeword length L for any distortionless source
encoding is bounded as
L>= H
According to source coding theorem the entropy H represents as the
fundamental limit on the average number of bits per source symbol
necessary to represent a discrete memoryless source
It can be made as small as, but not smaller than the entropy H thus, with
Lmin=H, then is represented as
= H/L
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SHANONS SECOND THEOREM

Why channel coding?
Goal of channel coding
Process
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Co t
Approach
The code rate is given as
Suppose a discrete memoryless source alphabet S, Entropy H(S) bits per

source symbol and source emits and delivers symbols for every Ts seconds
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STATEMENT
Two parts
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SHANONS THIRD THEOREM

The channel capacity of a white band limited
gaussian channel is
Signal power
Noise power
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MUTUAL INFORMATION
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MUTUAL INFORMATION
The difference between these two values is

called mutual information
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PROPERTIES OF MUTUAL INFORMATION

1. The mutual info of channel is symmetric
2. The mutual information is always non negative
3. The mutual information of a channel is related to the joint
entropy of the channel input and the channel output by
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CHANNEL CAPACITY
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CHANNEL CAPACITY
What is channel capacity?
It must satisfy two constraints
Co t
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Transmission efficiency
Redundancy
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Error Control
Coding
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Introduction
Error Control Coding (ECC)
Extra bits are added to the data at the transmitter

(redundancy) to permit error detection or
correction at the receiver
Done to prevent the output of erroneous bits
despite noise and other imperfections in the
channel
The positions of the error control coding and
decoding are shown in the transmission model
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Transmission Model
Error
Digital
Source
Source
Encoder
Control
Coding
Line
Coding
Modulator
X(w)
(Transmit
Filter, etc)
Hc(w)
Transmitter
N(w)
Channel
Noise
+
Error
Digital
Source
Sink
Decoder
Control
Decoding
Demod
Line
Decoding
(Receive
Filter, etc)
Y(w)
Receiver
Error Models
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Binary Symmetric Memoryless Channel

Assumes transmitted symbols are binary
E o s affe t s a d s ith e ual p o a ilit
(i.e., symmetric)
Errors occur randomly and are independent
from bit to bit (memoryless)
1-p
0
IN
0
OUT
p is the probability of bit

error or the Bit Error Rate
(BER) of the channel
p
1
1
1-p
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Error Models
Many other types
Burst errors, i.e., contiguous bursts of bit
errors
output from DFE (error propagation)
common in radio channels
Insertion, deletion and transposition errors
We will consider mainly random errors
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Error Control Techniques

Error detection in a block of data
Can then request a retransmission, known as
automatic repeat request (ARQ) for sensitive
data
Appropriate for
Low delay channels
Channels with a return path
Not appropriate for delay sensitive data, e.g.,

real time speech and data
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Error Control Techniques

Forward Error Correction (FEC)
Coding designed so that errors can be corrected at
the receiver
Appropriate for delay sensitive and one-way
transmission (e.g., broadcast TV) of data
Two main types, namely block codes and
convolutional codes. We will only look at block
codes
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Block Codes
We will consider only binary data
Data is grouped into blocks of length k bits
(dataword)
Each dataword is coded into blocks of length n
bits (codeword), where in general n>k
This is known as an (n,k) block code
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Block Codes
A vector notation is used for the datawords
and codewords,
Dataword d = (d1 d2.dk)
Codeword c = (c1 c2..cn)
The redundancy introduced by the code is

quantified by the code rate,
Code rate = k/n
i.e., the higher the redundancy, the lower the
code rate
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Block Code - Example
Dataword length k = 4
Codeword length n = 7
This is a (7,4) block code with code rate = 4/7
For example, d = (1101), c = (1101001)
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Error Control Process

101101
1000
1000
Source code
data chopped
into blocks
Channel
Dataword
(k bits)
Dataword
(k bits)
Codeword
(n bits)
coder
Codeword +
possible errors
(n bits)
Channel
decoder
Channel
Error flags
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Error Control Process

Decoder gives corrected data
May also give error flags to
Indicate reliability of decoded data
Helps with schemes employing multiple layers of
error correction
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Hamming Distance
Error control capability is determined by
the Hamming distance
The Hamming distance between two
codewords is equal to the number of
differences between them, e.g.,
10011011
11010010 have a Hamming distance = 3
Alternatively, can compute by adding

codewords (mod 2)
=01001001 (now count up the ones)
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Hamming Distance
The Hamming distance of a code is equal to

the minimum Hamming distance between
two codewords
If Hamming distance is:
1 no error control capability; i.e., a single error
in a received codeword yields another valid
codeword
XXXXXXX
X is a valid codeword
Note that this representation is diagrammatic
only.
In reality each codeword is surrounded by n
codewords. That is, one for every bit that
could be changed
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Hamming Distance
2 can detect single errors (SED); i.e., a single error
will yield an invalid codeword
XOXOXO
O in not a valid codeword
See that 2 errors will yield a valid (but
incorrect) codeword
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Hamming Distance
3 can correct single errors (SEC) or can detect
double errors (DED)
XOOXOOX
O in not a valid codeword
See that 3 errors will yield a valid but incorrect
codeword
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Hamming Distance - Example

Hamming distance 3 code, i.e., SEC/DED
Or can perform single error correction (SEC)
10011011
11011011
11010011
11010010
X
O
O
X
This code corrected this way
This code corrected this way
O is an invalid codeword
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Hamming Distance
The maximum number of detectable errors is
d min 1
That is the maximum number of correctable
errors is given by,
d min 1
t
where dmin is the minimum Hamming distance

between 2 codewords and. means the smallest
integer
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Linear Block Codes

As seen from the second Parity Code
example, it is possible to use a table to hold
all the codewords for a code and to look-up
the appropriate codeword based on the
supplied dataword
Alternatively, it is possible to create
codewords by addition of other codewords.
This has the advantage that there is now no
longer the need to held every possible
codeword in the table.
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Linear Block Codes

If there are k data bits, all that is required is to hold k
linearly independent codewords, i.e., a set of k
codewords none of which can be produced by linear
combinations of 2 or more codewords in the set.
The easiest way to find k linearly independent
ode o ds is to hoose those hi h ha e i just
one of the first k positio s a d i the othe k-1 of
the first k positions.
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Linear Block Codes

For example for a (7,4) code, only four
codewords are required, e.g.,
1
So, to obtain the codeword for dataword 1011,

the first, third and fourth codewords in the list
are added together, giving 1011010
This process will now be described in more detail
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Linear Block Codes

An (n,k) block code has code vectors
d=(d1 d2.dk) and
c=(c1 c2..cn)
The block coding process can be written as

c=dG
where G is the Generator Matrix
a11 a12 ... a1n a1
a
21 a22 ... a2n a 2
.
.
...
. .

ak 2 study
akn fromwww.rejinpaul.com
... materials
ak
ak1Get useful
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Linear Block Codes

Thus,
k
c di a i
i 1
ai must be linearly independent, i.e.,

Since codewords are given by summations
of the ai vectors, then to avoid 2 datawords
having the same codeword the ai vectors
must be linearly independent
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Linear Block Codes

Sum (mod 2) of any 2 codewords is
also a codeword, i.e.,
Since for datawords d1 and d2 we have;
d3 d1 d 2
So,
k
i 1
i 1
i 1
i 1
c3 d 3i a i (d1i d 2i )a i d1i a i d 2i a i
c3 c1 c2
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Linear Block Codes

0 is always a codeword, i.e.,
Since all zeros is a dataword then,
k
c 0 ai 0
i 1
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Error Correcting Power of LBC

The Hamming distance of a linear block code (LBC) is
simply the minimum Hamming weight (number of
s o e ui ale tl the dista e f o the all
codeword) of the non-zero codewords
Note d(c1,c2) = w(c1+ c2) as shown previously
For an LBC, c1+ c2=c3
So min (d(c1,c2)) = min (w(c1+ c2)) = min (w(c3))
Therefore to find min Hamming distance just need
to search among the 2k codewords to find the min
Hamming weight far simpler than doing a pair
wise check for all possible codewords.
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Linear Block Codes example 1

For example a (4,2) code, suppose;
1 0 1 1
G
0
1
0
1
a1 = [1011]
a2 = [0101]
For d = [1 1], then;

c
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Linear Block Codes example 2

A (6,5) code with
1
0
G 0
1
1
1
1
Is an even single parity code
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Systematic Codes
For a systematic block code the dataword

appears unaltered in the codeword
usually at the start
The generator matrix has the structure,
k
1
0
G
..
R=n-k
0 .. 0 p11 p12 .. p1R

1 .. 0 p21 p22 .. p2 R
I | P
.. .. .. .. .. .. ..
0 .. 1 pk1 pk 2 .. pkR
P is often referred to as parity bits

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Systematic Codes
I is k*k identity matrix. Ensures dataword
appears as beginning of codeword
P is k*R matrix.
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Decoding Linear Codes
One possibility is a ROM look-up table

In this case received codeword is used as an address
Example Even single parity check code;
Address
000000
000001
000010
000011
Data
0
1
1
0
Data output is the error flag, i.e., 0 codeword ok,

If no error, dataword is first k bits of codeword
For an error correcting code the ROM can also store
datawords
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Decoding Linear Codes

Another possibility is algebraic decoding, i.e.,
the error flag is computed from the received
codeword (as in the case of simple parity
codes)
How can this method be extended to more
complex error detection and correction
codes?
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CYCLIC CODES
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Motivation & Properties of cyclic code

Cyclic code are a class of linear block codes.
Thus, we can find generator matrix (G) and
parity check matrix (H).

The reason is that they can be easily
implemented with externally cost effective

electronic circuit.
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Definition
An (n,k) linear code C is cyclic if every cyclic
shift of a codeword in C is also a codeword in
C.
If c0 c1 c2 . cn-2
cn-1 c0 c1 . cn-3
cn-2 cn-1 c0 . cn-4
: : :
:
c1 c2 c3 . cn-1
cn-1 is a codeword, then

cn-2
cn-3
:
c0
are all codewords.
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Example: (6,2) repetition code
C 000000, 010101, 101010, 111111

is a cyclic code.
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Example2: (5,2) linear block code
1 0 1 1 1
G
0
1
1
0
1
is a single error correcting code, the set of codeword are:
0
1
C
0
0 0 0 0 Thus, it is not a cyclic code

0 1 1 1 since, for example, the cyclic
shift of [10111] is [11011]
1 1 0 1
1 0 1 0
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Example 3
The (7,4) Hamming code discussed before is
cyclic:
1010001
1101000
0110100
0011010
0001101
1000110
0100011
1110010
0111001
1011100
0101110
0010111
1001011
1100101
0000000
1111111
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Generator matrix of a non-systematic (n,k) cyclic

codes
The generator matrix will be in this form:
g0
0
G0
g1
gnk 1
gnk
g0
g1
gnk 1
gnk
g0
g1
gnk 1 gnk
0
0
0
gnk
g0
gnk 1
notice that the row are merely cyclic shifts of the

g g0g1 gnk 1gnk 000
basis vector
r n
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The code vector are:

C m G ; where m [m0m1 mk 1 ]
c 0 m0 g 0
c1 m0 g1 m1g 0
c 2 m0 g 2 m1g1 m2 g 0
c n 1 m0 g n k m1g n k 1 mn k g 0
Notice that,k 1
C m j g 1 ; where m j 0, if j 0 or j k 1
j 0
This summation is a convolution between
and .
It would be much easier if we deal with multiplication, this transform is done

using Polynomial Representation.
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Code Polynomial
Let c = c0 c1 c2 . cn-1. The code polynomial
of c: c(X) = c0 + c1X+ c2 X2 + . + cn-1 Xn-1
where the power of X corresponds to the bit position,
and
the coefficients are 0s and 1s.
Example:
1010001 1+X2+X6
0101110 X+X3+X4+X5
Each codeword is represented by a polynomial of degree
less than or equal n-1. deg[ c(X) ] n 1
The addition and multiplication are as follow:
ax j bx j (a b) x j
(ax j ).(bx k ) (a . b) x j k
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Where (a+b) and (a.b) are under GF(2). But

j+k is integral addition
Example:
m( x ) m0 m1 x m2 x 2
g ( x ) g 0 g1 x
addition
m( x ) g ( x ) (m0 g 0 ) (m1 g1 )x (m2 0)x 2

Multipliation
m( x )g ( x ) m0g 0 (m0 g1 m1g 0 )x (m1g1 m2 g 0 )x 2 m2 g1 x 3
Notice that in multiplication the coefficient are the

same as convolution sum
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Implementing the Shift

Let c = c0
c1
c2 . cn-1
and c(i) = cn-i cn-i+1 c0 . cn-i-1 (i shifts to the right)
c(X) = c0 + c1X+ c2 X2 + . + cn-1 Xn-1
c (i)(X) = cn-i + cn-i+1 X + . + cn-1 Xi-1 + . +
c0Xi +. +cn-i-1 Xn-1
What is the relation between c(X) and c (i)(X)?
Apparently, shifting a bit one place to the right is
equivalent
to multiplying the term by X.
Xic(X)= c0Xi +c1X i+1 + .+ cn-i-1 Xn-1 + cn-i Xn .+ cn-1 Xn+i-1
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Implementing the Shift (contd)

Xic(X) = cn-i Xn ++cn-1 Xn+i-1 +c0Xi +c1X i+1 + + cn-i-1 Xn-1
The first i terms have powers n, and are not suitable
for
representing bit locations.

Add to the polynomial the zero-valued sequence:
(cn-i + cn-i ) + (cn-i+1 + cn-i+1 )X + . + (cn-1 + cn-1 )Xi-1
Xic(X) = cn-i (Xn +1) + cn-i+1 X (Xn +1)+. +cn-1 Xi-1 (Xn +1)+
cn-i
+ cn-i+1 X
+. +cn-1 Xi-1+
c0Xi +c1X i+1 + . + cn-i-1 Xn-1
That is:
Xic(X) = q(X)(Xn +1) + c(i)(X)
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Implementing the Shift (contd)

c(i)(X) is the remainder from dividing Xic(X) by (Xn +1).
c(i)(X) = Rem[Xic(X)/ (Xn +1)] = Xic(X) mod (Xn +1).
Example:
c = 0101110. c(X) = X + X3 + X4 + X5.
X3c(X) = X4 + X6 + X7 + X8
Rem[X3c(X)/ (X7 +1)] = 1 + X + X4 + X6 [Show]
c(3) = 1100101
Short cut of long division:
Xic(X)|Xn=1 = q(X)(Xn +1) |Xn=1 + c(i)(X) |Xn=1
That is
c(i)(X) = Xic(X)|Xn=1
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More on Code Polynomials

The nonzero code polynomial of minimum degree in a
cyclic code C is unique.
(If not, the sum of the two polynomials will be a code polynomial of degree less than
the minimum. Contradiction)
Let g(X) = g0 + g1X +.+ gr-1Xr-1 +Xr be the nonzero code

polynomial of minimum degree in an (n,k) cyclic code.
Then the constant term g0 must be equal to 1.
(If not, then one cyclic shift to the left will produce a code polynomial of degree less
than the minimum. Contradiction)
For the (7,4) code given in the Table, the nonzero code
polynomial of minimum degree is g(X) = 1 + X + X3
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Generator Polynomial
Since the code is cyclic: Xg(X), X2g(X),., Xn-r-1g(X)
are code polynomials in C. (Note that deg[Xn-r1g(X)] = n-1).
Since the code is linear:
(a0 + a1X + . + an-r-1 Xn-r-1)g(X) is also a code
polynomial, where ai = 0 or 1.
A binary polynomial of degree n-1 or less is a

code polynomial if and only if it is a multiple of
g(X).
(First part shown. Second part: if a code polynomial c(X) is not a
multiple of g(X), then Rem[c(X)/g(X)] must be a code polynomial of
degree less than the minimum. Contradiction)
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Generator Polynomial (contd)

All code polynomials are generated from the
multiplication c(X) = a(X)g(X).
deg[c(x)] n-1, deg[g(X)] = r, ==> deg[a(x)] n-r-1
(2k) = # different ways of forming a(x), 2n-r
# codewords,
Therefore, r = deg[g(X)] = n-k
Since deg[a(X)] k-1, the polynomial a(X) may be taken

to be the information
polynomial u(X) (a polynomial
whose coefficients are the information bits). Encoding

is performed by the multiplication c(X) = u(X)g(X).
g(X), generator polynomial, completely defines the
code.
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(7,4) Code Generated by 1+X+X3

Infor.
0000
1000
0100
1100
0010
1010
0110
1110
0001
Code
0000000
1101000
0110100
1011100
0011010
1110010
0101110
1000110
0001101
Code polynomials
0 = 0 . g(X)
1 + X + X3 = 1 . g(X)
X + X2 + X4 = X . g(X)
1 + X2 + X3 + X4 = (1 + X) . g(X)
X2 + X3 + X5 = X2 . g(X)
1 + X+ X2 + X5 = (1 + X2) . g(X)
X+ X3 + X4 + X5 = (X+ X2) . g(X)
1 + X4 + X5 = (1 + X + X2) . g(X)
X3 + X4 + X6 = X3 . g(X)
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(7,4) Code Generated by 1+X+X3

(Contd)
Infor.
1001
0101
1101
0011
1011
Code
1100101
0111001
1010001
0010111
1111111
0111 0100011
1111 1001011
Code polynomials
1 + X + X4 + X6 = (1 + X3) . g(X)
X+ X2 + X3 + X6 = (X+ X3) . g(X)
1 + X2 + X6 = (1 + X + X3) . g(X)
X2 + X4 + X5 + X6 = (X2 + X3). g(X)
1 + X + X2 + X 3 + X 4 + X 5 + X 6
= (1 + X2 + X3) . g(X)
X + X5 + X6 = (X + X2 + X3). g(X)
1 + X 3 + X 5 + X6
= (1 + X + X2 + X3) . g(X)
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Constructing g(X)
The generator polynomial g(X) of an (n,k) cyclic
code is a factor of Xn+1.
Xkg(X) is a polynomial of degree n.
Xkg(X)/ (Xn+1)=1 and remainder r(X). Xkg(X) = (Xn+1)+ r(X).
But r(X)=Rem[Xkg(X)/(Xn+1)]=g(k)(X) =code polynomial= a(X)g(X).
Therefore, Xn+1= Xkg(X) + a(X)g(X)= {Xk + a(X)}g(X). Q.E.D.
(1)To construct a cyclic code of length n, find the

factors of the polynomial Xn+1.
(2)The factor (or product of factors) of degree n-k
serves as the generator polynomial of an (n,k)
cyclic code. Clearly, a cyclic code of length n does
not exist for every k.
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Constructing g(X) (contd)

(3)The code generated this way is guaranteed to be cyclic.
But we know nothing yet about its minimum distance.
The generated code may be good or bad.
Example: What cyclic codes of length 7 can be
constructed?
X7+1 = (1 + X)(1 + X + X3)(1 + X2 + X3)
g(X)
Code
g(X)
Code
(1 + X)
(7,6)
(1 + X)(1 + X + X3)
(7,3)
(1 + X + X3)
(7,4)
(1 + X) (1 + X2 + X3)
(7,3)
(1 + X2 + X3)
(7,4)
(1 + X + X3)(1 + X2 + X3) (7,6)
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Circuit for Multiplying Polynomials (1)

u(X) = uk-1Xk-1 + . + u1X + u0
g(X) = grXr + gr-1Xr-1 + . + g1X + g0
u(X)g(X) = uk-1grXk+r-1
+ (uk-2gr+ uk-1gr-1)Xk+r-2 + .
+ (u0g2+ u1g1 +u2g0)X2 +(u0g1+ u1g0)X +u0g0
gr
gr-1
gr-2
g1
g0
Output
Input
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Circuit for Multiplying Polynomials (2)

u(X)g(X) = uk-1Xk-1(grXr + gr-1Xr-1 + . + g1X + g0)
+ .
+ u1X(grXr + gr-1Xr-1 + . + g1X + g0)
+ u0(grXr + gr-1Xr-1 + . + g1X + g0)
g0
g1
g2
gr-1
gr
Output
Input
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Systematic Cyclic Codes

Systematic: b0 b1 b2 . bn-k-1 u0 u1 u2 . uk-1
b(X) = b0 + b1X+.+bn-k-1Xn-k-1, u(X) = u0+u1X+ .+uk-1Xk-1
then c(X) = b(X) + Xn-k u(X)
a(X)g(X) = b(X) + Xn-k u(X)
Xn-k u(X)/g(X) = a(X) + b(X)/g(X)
Or
b(X) = Rem[Xn-k u(X)/g(X)]
Encoding Procedure:
1. Multiply u(X) by Xn-k
2. Divide Xn-k u(X) by g(X), obtaining the remainder b(X).
3. Add b(X) to Xn-k u(X), obtaining c(X) in systematic form.
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Systematic Cyclic Codes (contd)

Example
Consider the (7,4) cyclic code generated by
g(X) = 1 + X + X3. Find the systematic codeword for
the message 1001.
u(X) = 1 + X3
X3u(X) = X3 + X6
b(X) = Rem[X3u(x)/g(X)] = X3u(x) |g(X) = 0 = X3u(x) |X3
= X+1
= X3 (X3 +1) = (1 + X)X = X + X2

Therefore, c = 0111001
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Circuit for Dividing Polynomials
Output
g0
g1
g2
gr-1
gr
Input
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Encoder Circuit
Gate
g1
g2
gr-1
Gate ON. k message bits are shifted into the channel.

The parity bits are formed in the register.
Gate OFF. Contents of register are shifted into the
channel.
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(7,4) Encoder Based on 1 + X + X3

Gate
+
Input
Register :
000
initial
+
1
110
1st shift
1
101
2nd shift
3rd shift
0
100
1
100
4th shift
Codeword: 1 0 0 1 0 1 1
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Parity-Check Polynomial
Xn +1 = g(X)h(X)
deg[g(x)] = n-k, deg[h(x)] = k
g(x)h(X) mod (Xn +1) = 0.
h(X) is called the parity-check polynomial. It
plays the rule of the H matrix for linear codes.
h(X) is the generator polynomial of an (n,n-k)
cyclic code, which is the dual of the (n,k) code
generated by g(X).
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Decoding of Cyclic Codes

STEPS:
(1) Syndrome computation
(2) Associating the syndrome to the error pattern
(3) Error correction
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Syndrome Computation
Received word: r(X) = r0 + r1X +.+ rn-1Xn-1
If r(X) is a correct codeword, it is divisible by g(X).
Otherwise: r(X) = q(X)g(X) + s(X).
deg[s(X)] n-k-1.
s(X) is called
the syndrome polynomial.
s(X) = Rem[r(X)/g(X)] = Rem[ (a(X)g(X) + e(X))/g(x)]
= Rem[e(X)/g(X)]
The syndrome polynomial depends on the error pattern
only.
s(X) is obtained by shifting r(X) into a divider-by-g(X)
circuit. The register contents are the syndrome bits.
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Example: Circuit for Syndrome Computation

Gate
r = 0010110
+
Shift
1
2
3
4
5
6
7
Input
0
1
1
0
1
0
0
Register contents
0 0 0 (initial state)
000
100
110
011
011
111
1 0 1 (syndrome s)
What is g(x)?
Find the syndrome using
long division.
Find the syndrome using
the shortcut for the
remainder.
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Association of Syndrome to Error

Pattern
Look-up table implemented via a combinational logic circuit

(CLC). The complexity of the CLC tends to grow exponentially
with the code length and the number of errors to correct.
Cyclic property helps in simplifying the decoding circuit.
The circuit is designed to correct the error in a certain location
only, say the last location. The received word is shifted cyclically
to trap the error, if it exists, in the last location and then correct
it. The CLC is simplified since it is only required to yield a single
output e telling whether the syndrome, calculated after every
cyclic shift of r(X), corresponds to an error at the highest-order
position.
The received digits are thus decoded one at a time.
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Meggit Decoder
Shift r(X) into the buffer B and the syndrome
register R simultaneously. Once r(X) is completely
shifted in B, R will contain s(X), the syndrome of
r(X).
1. Based on the contents of R, the detection circuit
yields the output e (0 or 1).
2. During the next clock cycle:
(a) Add e to the rightmost bit of B while shifting
the contents of B. (The rightmost bit of B may be
read out). Call the modified content of B r1(1)(X).
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Meggit Decoder (contd)

(b) Add e to the left of R while shifting the
contents of R. The modified content of R is
s1(1)(X), the syndrome of r1(1)(X) [will be shown
soon].
Repeat steps 1-2 n times.
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General Decoding Circuit

.
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More on Syndrome Computation

Let s(X) be the syndrome of a received polynomial r(X)
= r0 + r1X +.+ rn-1Xn-1 . Then the remainder resulting
from dividing Xs(X) by g(X) is the syndrome of r(1)(X),
which is a cyclic shift of r(X).
Proof: r(X) = r0 + r1X +.+ rn-1Xn-1
r(1)(X) = rn-1 + r0X +.+ rn-2Xn-1 = rn-1 + Xr(X) + rn-1Xn
= rn-1(Xn+1) + Xr(X)
c(X)g(X) + y(X) = rn-1 g(X)h(X)+ X{a(X)g(x) + s(X)}

where y(X) is the syndrome of r(1)(X) .
Xs(X) = {c(X) + a(X) + rn-1 h(X)}g(X) + y(X)
Therefore, Syndrome of r(1)(X)= Rem[Xs(X)/g(X)]. Q.E.D.
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(contd)
Note: for simplicity of notation, let Rem[Xs(X)/g(X)]

be denoted by s(1)(X). s(1)(X) is NOT a cyclic shift of
s(X), but the syndrome of r(1)(X) which is a cyclic
shift of r(X).
Example:
r(X) = X2 + X4 + X5; g(X) = 1 + X + X3
s(X) = Rem[r(X)/g(X)] = 1 + X2
r(1)(X) = X3 + X5 + X6
s(1)(X) = Rem[r(1)(X)/g(X)] = 1 (polynomial)
Also, s(1)(X) = Rem[Xs(X)/g(X)] = 1.
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(contd)
Gate
r = 0010110
Gate +
Shift
Input
1
2
3
4
5
6
7
8 (input gate off)
9
0
1
1
0
1
0
0
-
Register contents
0 0 0 (initial state)
000
100
110
011
011
111
1 0 1 (syndrome s)
1 0 0 (syndrome s (1) )
0 1 0 (syndrome s (2) )
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(contd)
Let r(X) = r0 + r1X +.+ rn-1Xn-1 has the syndrome s(X).Then

r(1)(X) = rn-1 + r0 X + .+ rn-2Xn-1 has the syndrome:
s(1)(X) = Rem[r(1)(X)/g(X)].
Define r1 (X) = r(X) + Xn-1 = r0 + r1X +.+ (rn-1+1)Xn-1
The syndrome of r1 (X), call it s1 (X):
s1 (X)= Rem[{r(X)+ Xn-1}/g(X)] = s(X) + Rem[Xn-1/g(X)]
r1(1)(X), which is one cyclic shift of r1 (X), has the
syndrome
s1(1)(X) = Rem[X s1 (X)/g(X)] = Rem[Xs(X)/g(X)+ Xn/g(X)]
= s(1)(X) + 1 (since Xn +1 = g(X)h(X))
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Worked Example
Consider the (7,4) Hamming code generated by 1+X+X3.
Error pattern
X6
X
X4
X3
X2
X1
X0
Syndrome polynomial.
1 + X2
1 + X + X2
X + X2
1+X
X2
X
1
101
111
011
110
001
010
100
Let c = 1 0 0 1 0 1 1 and r = 1 0 1 1 0 1 1
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Cyclic Decoding of the (7,4) Code

.
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Error Correction Capability

Error correction capability is inferred from the
roots of g(X).
Results from Algebra of Finite Fields:
Xn +1 has n roots (in an extension field)
These roots can be expressed as powers of one
element, a.
The roots are a0, a1 , ., an-1.
The roots occur in conjugates.
i
a
2 j mod n
constitute a conjugate set.
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Designing a Cyclic Code

Theorem:
If g(X) has l roots (out of it n-k roots) that are consecutive
powers of a, then the code it generates has a minimum
distance d = l + 1.
To design a cyclic code with a guaranteed minimum
distance of d, form g(X) to have d-1 consecutive roots. The
parameter d is called the designed minimum distance of
the code.
Since roots occur in conjugates, the actual number of
consecutive roots, say l, may be greater than d-1. dmin = l +
1 is called the actual minimum distance of the code.
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Design Example
X15 + 1 has the roots 1= a0, a1 , ., a14.
Conjugate group
polynomial
a0)
(a, a2 , a4 , a8)
(a3 , a6 , a9 , a12)
X4
(a5 , a10)
(a7, a14 , a13 , a11)
Corresponding
f1X1 + X
f2X 1 + X + X4
f3X 1 + X + X2 + X3 +
f4X 1 + X + X2
f5X 1 + X3 + X4
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Design Example (contd)

Find g(X) that is guaranteed to be a double error
correcting code.
The code must have a, a2 , a3 and a4 as roots.
g(X) = f2X f3X = 1 + X4 + X6 + X7 + X8
This generator polynomial generates a (15, 7) cyclic code
of minimum distance at least 5.
Roots of g(X) = a, a2, a3 , a4 , a6, a8 , a9 , a12.
Number of consecutive roots = 4.
The actual minimum distance of the code is 5.
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Cyclic Codes
Some Standard Cyclic

Block Codes
BCH Codes
Linear
Codes
Hamming Codes
The Hamming Codes: single-error correcting codes

which can be expressed in cyclic form.
BCH: the Bose-Chaudhuri-Hocquenghem are among
the most important of all cyclic block codes.
Extenstion of Hamming for t-error correcting codes.
Some Burst-Correcting Codes: good burstcorrecting codes have been found mainly by
computer search.
Cyclic Redundancy Check Codes: shortened cyclic
error-detecting codes used in automatic repeat
request (ARQ) systems.
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BCH Codes
Definition of the codes:
For any positive integers m (m>2) and t0 (t0 <
n/2), there is a BCH binary code of length n =
2m - 1 which corrects all combinations of t0 or
fewer errors and has no more than mt0 paritycheck bits.
Block length
2m 1
Number of parity - check bits
n k mt 0
min imum distance
d min 2t 0 1
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Table of Some BCH Codes

n
7
15
15
15
31
31
31
k
4
11
7
5
26
16
11
d (designed) d ( actual)
3
3
3
3
5
5
7
7
3
3
5
7
7
11
g(X)*
13
23
721
2463
45
107657
5423325
* Octal representation with highest order at the left.

721 is 111 010 001 representing 1+X4+X6+X7+X8
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Burst Correcting Codes

good burst-correcting codes have been found mainly
by computer search.
The length of an error burst, b, is the total number of
bits in error from the first error to the last error,
inclusive.
The minimum possible number of parity-check bits
required to correct a burst of length b or less is given
r 2b
by the Rieger bound.
The best understood codes for correcting burst errors
are cyclic codes.
For correcting longer burst interleaving is used.
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Table of Good Burst-Correcting Codes

n
7
15
15
31
63
63
511
1023
k
3
10
9
25
56
55
499
1010
b
2
2
3
2
2
3
4
4
g(X) (octal)
35 (try to find dmin!)
65
171
161
355
711
10451
22365
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Cyclic Redundancy Check Codes

Shortened cyclic codes
Error-detecting codes
used in automatic repeat request (ARQ)
systems.
Usually concatenated with error correcting code
CRC
Encoder
Error Correction
Encoder
To
Error Correction
Decoder
CRC
Syndrom
e Checker
To
Info Sink
Transmitter
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Performance of CRC Codes

CRC are typically evaluated in terms of their
error pattern coverage
Burst error detection capability
Probability of undetected error
For a (n,k) CRC the coverage, , is the ratio of the number of invalid
blocks of length n to the total number of blocks of length n.
This ratio is a measure of the probability that a randomly chosen block is
not a valid code block. By definition,
1 2 r
where r is the number of check bits

For some near-optima CRC codes,
see table 5.6.5
Code
Coverage
CRC-12
0.999756
CRC-ANSI
0.999985
CRC-32A
0.99999999977
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Basic Definitions
k =1, n = 2 , (2,1) Rate-1/2 convolutional code

Two-stage register ( M=2 )
Each input bit influences the output for 3 intervals
(K=3)
K = constraint length of the code = M + 1
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Generator Polynomial
A convolutional code may be defined by a set
of n generating polynomials for each input bit.
For the circuit under consideration:
g1(D) = 1 + D + D2
g2(D) = 1 + D2
The set {gi(D)} defines the code completely.

The length of the shift register is equal to the
highest-degree generator polynomial.
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State Diagram Representation

The output depends on the current input and
the state of the encoder ( i. e. the contents of
the shift register).
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Trellis Diagram Representation

Expansion of state diagram in time.
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Decoding
A message m is encoded into the code
sequence c.
Each code sequence represents a path in
the trellis diagram.
Minimum Distance Decoding
Upon receiving the received sequence r, search
for the path that is closest ( in Hamming
distance) to r .
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The Viterbi Algorithm

Walk through the trellis and compute the Hamming
distance between that branch of r and those in the
trellis.
At each level, consider the two paths entering the
same node and are identical from this node onwards.
From these two paths, the one that is closer to r at this
stage will still be so at any time in the future. This path
is retained, and the other path is discarded.
Proceeding this way, at each stage one path will be
saved for each node. These paths are called the
survivors. The decoded sequence (based on MDD) is
guaranteed to be one of these survivors.
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The Viterbi Algorithm (contd)

Each survivor is associated with a metric of
the accumulated Hamming distance (the
Hamming distance up to this stage).
Carry out this process until the received
sequence is considered completely. Choose
the survivor with the smallest metric.
The Viterbi Algorithm:
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The viterbi algorithm is used to decode convolutional codes

and any structure or system that can be described by a trellis.
It is a maximum likelihood decoding algorithm that selects
the most probable path that maximizes the likelihood
function.
The algorithm is based on add-compare-select the best path
each time at each state.
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Example: For the convolutional code example in the previous lecture,

starting from state zero, Decode the following received sequence.
At the end of the
trellis, select the
path with the
minimum
cumulative
Hamming weight
This is the
survival
path in
this
example
Decoded
sequence is
m=[10 1110]
Add the weight of the
path at each state
Compute the two possible paths at

each state and select the one
with less cumulative Hamming
Getweight
useful study materials
This is called the

survival path
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Distance Properties of Conv. Codes

Def: The free distance, dfree, is the minimum Hamming
distance between any two code sequences.
Criteria for good convolutional codes:
Large free distance, dfree.

Small Hamming distance (i.e. as few differences as possible
) between the input information sequences that produce
the minimally separated code sequences. dinf
There is no known constructive way of designing a

conv. code of given distance properties. However, a
given code can be analyzed to find its distance
properties.
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Distance Prop. of Conv. Codes (contd)

Convolutional codes are linear. Therefore, the
Hamming distance between any pair of code
sequences corresponds to the Hamming distance
between the all-zero code sequence and some nonzero
code sequence. Thus for a study of the distance
properties it is possible to focus on the Hamming
distance between the all-zero code sequence and all
nonzero code sequences.
The nonzero sequence of minimum Hamming weight
diverges from the all-zero path at some point and
remerges with the all-zero path at some later point.
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Distance Properties: Illustration
sequence 2: Hamming weight = 5, dinf = 1

sequence 3: Hamming weight = 7, dinf = 3.
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Modified State Diagram

The span of interest to us of a nonzero path starts
from the 00 state and ends when the path first
returns to the 00 state. Split the 00 state (state a)
to two states: a0 and a1.
The branches are labeled with the dummy
variables D, L and N, where:
The power of D is the Hamming weight (# of 1s) of the
output corresponding to that branch.
The power of N is the Hamming weight (# of 1s) of the
information bit(s) corresponding to that branch.
The power of L is the length of the branch (always = 1).
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Modified State Diagram (contd)

.
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Properties of the Path

Sequence 2:
code sequence:
state sequence:
Labeled:
Prop. : w =5, dinf
branches.
Sequence 3:
code sequence:
state sequence:
Labeled:
Prop. : w =7, dinf
branches.
.. 00 11 10 11 00 ..
a0 b c a1
(D2LN)(DL)(D2L) = D5L3N
=1, diverges from the allzero path by 3
.. 00 11 01 01 00 10 11 00 ..
a0 b d c b c a1
(D2LN)(DLN)(DL)(DL)(LN)(D2L) = D7L6N3
=3, diverges from the allzero path by 6
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Transfer Function
Input-Output relations:
a0 = 1
b = D2LN a0 + LNc
c = DLb + DLNd
d = DLNb + DLNd
a1 = D2Lc
The transfer function T(D,L,N) = a1 /a0

D5 L3
T(D, L, N)
1 DNL(1 L)
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Transfer Function (contd)

Performing long division:
T = D5L3N + D6L4N2 + D6L5N2 + D7L5N3 + .
If interested in the Hamming distance property of

the code only, set N = 1 and L = 1 to get the
distance transfer function:
T (D) = D5 + 2D6 + 4D7
There is one code sequence of weight 5. Therefore
dfree=5.
There are two code sequences of weight 6,
four code sequences of weight 7, .
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Decoding of Convolutional Codes

Let Cm be the set of allowable code sequences of length m.
Not all sequences in {0,1}m are allowable code sequences!
Each code sequence c C

can be represented by a unique path
m
through the trellis diagram
What is the probability that the code sequence is sent and the
c
binary sequence
is received?
dH y,
c
mdH y,
c
Pr y |
c p
.(1 p)
where p is the probability of bit error of BSC from modulation
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Decoding Rule for Convolutional Codes

Maximum Likelihood Decoding Rule:
d y, c
c min
Pr y,
cCm H
cCm
Choose the code sequence through the trellis which has the
max
smallest Hamming distance to the received sequence!
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The Viterbi Algorithm (Viterbi, 1967) is a clever way of
implementing Maximum Likelihood Decoding.
Computer Scientists will recognize the Viterbi Algorithm as an
e a ple of a C te h i ue alled D a i P og a
i g
efe e e: G. D. Fo e , The Vite i Algo ith ,
Proceedings of the IEEE, 1973
Chips are available from many manufacturers which
implement the Viterbi Algorithm for K < 10
Can be used for either hard or soft decision decoding
We consider hard decision decoding initially
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Basic Idea of Viterbi Algorithm

There are 2rm code sequences in Cm .
This number of sequences approaches infinity as m
becomes large
Instead of searching through all possible sequences,

find the best code sequence "one stage at a time"
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(Hamming Distance Metric)
Initialization:
Let time i = 0.
We assign each state j a metric Z j 0 at time 0.
We know that the code must start in the state 0.
Therefore we assign:
Z j 0 0
Z j 0 for all other states
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The Viterbi Algorithm (continued)

Consider decoding of the ith segment:
Let y i be the segment of n bits received between times i
and i + 1
ci
There are several code segments
of n bits which lead
into state j at time i+1. We wish to find the most likely one.
ci
Let sc i be the state from which the code segment
emerged
ci
For each state j, we assume that
into
state j if:
Z s
c i i d
is the path leading
c ,y
H i
is the smallest of all the code segments leading into state

j.
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The Viterbi Algorithm (continued)

Iteration:
Let
Let i=i+1
i d c i , y
Z i 1 Z
s
ci
j
H
Repeat previous step

Incorrect paths drop out as i approaches infinity.
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Viterbi Algorithm Decoding Example

r =1/2, K = 3 code from previous example
c = (0 0 1 1
y
= (0 1 1 1
01
00
10
10
1 1) is sent
01
00
10
10
1 1) is received.
What path through the trellis does the Viterbi Algorithm choose?
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Viterbi Algorithm Decoding Example

(continued)
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SUMMARY
Learnt about the concepts of Entropy and the
source coding techniques
Statement and the theorems of Shanon
Concepts about mutual information and
channel capacity
Understand error control coding techniques
and the concepts about linear block codes,
cyclic codes, convolution codes & viterbi
decoding algorithm
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Thank you

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Low pass sampling

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Digital communication system

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How can a continuous wave form be

How can discrete samples be converted back

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Low Pass Sampling

Measure amplitude at regular intervals

How many times should we sample?

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where fs is sampling frequency and fm is the

But what if one cannot sample fast

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Reduce signal frequency to half of

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Three different sampling methods

Practical Sampling Methods are Natural Sampling

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Pulse-Amplitude Modulation (PAM)

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The advantages offered by digital pulse modulation www.rejinpaul.com

I k : {mk < m mk +1}, k = 1,2,..., L (17)

Representation level (or Reconstruction level)

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Two types of quantization are

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Signal Range is Symmetric

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Example: Nonuniform 3 bit quantizer

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Uniform and Non-uniform Quantization

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u-LAW and A-LAW definitions

-law is neither strictly linear nor strictly logarithmic

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Example: -law Companding

Close View of the Signal

A-law and law Companding

These two are standard companding methods.

PCM (Pulse-Code Modulation)