Ec6501 - Notes - Rejinpaul PDF

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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 Low
Signal Source Channel
Pass Sampler Quantizer Multiplexer
Analog/ Encoder Encoder
Filter
Digital
Carrier
Pulse
Line
To Channel Modulator Shaping
Encoder
Filters
De- Receiver
From Channel Detector
Modulator Filter
Carrier Ref.
Signal
Digital-to-Analog Channel De-
at the
Converter Decoder Multiplexer
user end
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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

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Back Next
Fig.5

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Fig.6 Back Next

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Back Next
Fig.7

• 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
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

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Back Next
Fig.9

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Fig.10 Back Next
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
2 Example: Nonuniform 3 bit quantizer
-8 -6 -4 -2 2 4 6 8
-2
Input sample
X
-4
-6

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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-to-
quantization 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
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• A particular form of compression law : μ-law
log(1 + µ m )
v= (5.23)
log(1 + µ )
d m log(1 + µ )
= (1 + µ m ) (5.24)
dv µ
• μ-law is neither strictly linear nor strictly logarithmic
• A-law :
 Am 1
1 + log A , 0 ≤ m ≤
 A
v = (5.25)
1 + log( A m ) , 1 ≤ m ≤ 1
 1 + log A A
1 + log A 1
, 0 ≤ m ≤
d m  A A
= (5.26)
dv  1 Fig. 5.11
(1 + log A) m , A ≤ m ≤ 1
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Fig.11 Back Next

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

Eeng 360 47
A-law and µ−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 doesn’t 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 ?
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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 doesn’t
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 Uniform
Expander
Function Quantization
Function
(-1)
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Pulse-Code Modulation
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• 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 input-
output 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.
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Fig.11 Back Next

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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

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• Regeneration Along the Transmission Path
– 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
– Decision-making device Fig. 13
• 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.

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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
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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 k6
• Used in 2nd generation
c
Frequency
f
t
• Advantages: Time
– 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 intersymbol-
interference.
• 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.
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Conceptual diagram of multiplexing-demultiplexing.
PAM TDM System

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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
0 Ts 2Ts

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TDM-PAM for 4 signals.
4
4
4
1 1 1
2 2 2
3 3 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 Δ-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) = Σj=0Q gjZ-j, A(Z) = Σi=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 εodels” 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) = Σ (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 non-
negative 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) = Σi=1P’ ai x(n-i)
• The predicted error
• (n) = x(n)-x(n) = x(n) - Σi=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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• σ2 = Σn 2(n) (time average replaced means)

• It could be proved that if x(n) is generated by “full
poles” model : x(n) = -Σi=1P ai x(n-i) + Ge(n) and
optimized P’ = P, optimized ai = ai, σ2 is minimal.
• σ2 = Σn [x(n) -Σi=1P ai x(n-i)]2
• ={Σn x2(n)}-2Σi=1P ak{Σn x(n-k)x(n)}+
• Σk=1PΣi=1P akai{Σn x(n-k)x(n-i)}
• By setting (σ2 )/ ak = 0 we can get
• -2 {Σn x(n-k)x(n)}+2Σi=1P ai{Σn x(n-k)x(n-i)}=0
• Or Σi=1P aiφ(k,i) = φ(k,0)
• if φ(k,i) =Σn x(n-k)x(n-i) 1<=i<=P and 1<=k<=P
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• Σi=1P aiφ(k,i) = φ(k,0), k=1~P is called δPC

canonical equations. There are some different
algorithms to deal with the solution.
• [σ2]min = Σk=0P ak φ(k,0) with a0 = 1
• So if we have x(n), φ(k,i) could be calculated,
and equations could be solved to get ai and
[σ2]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)

• φ(k,i) = Σn xw(n-k)xw(n-i)
• If window length is N-1 then the summation
range will be 0~N+P-1
• φ(k,i) = Σm xw(m+(i-k))xw(m) = R(i-k) if n-i
=m
• φ(k,i) = R(i-k) = R(k-i) = R(|i-k|) <= R(0)
• The equations became Σi=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)| |a1| | R(1) |

• |R(1) R(0) …… R(P-2)| |a2| | R(2) |
• |………………………………….| |...| = …...
• |R(P-1) R(P-2) … R(0) | |ap| | R(P) |
• 6.2.1 Durbin Algorithm

• 1. E(0) = R(0)
• 2. Ki = [ R(i) - Σ 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
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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) –Σk=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
Get useful 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
• 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 :
• σ2 = Σn=0N-1 2(n) n=0~N-1
• φ(k,i) = Σn=0N-1 x(n-k)x(n-i) k=1~p, i=0~p
• = Σm=-iN-i-1 x(m+(i-k))x(m) let n-i=m, m=-i~N-i-1
• The equations will be like following :
• |φ(1,1) φ(1,2) …… φ(1,p)| |a1| |φ(1,0)|
• |φ(2,1) φ(2,2) …… φ(2,p)| |a2| |φ(2,0)|
• .………………………………………………=…………
• |φ(p,1) φ(p,2) …… φ(p,p)| |ap| |φ(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 :
• aj(p) = aj 1<=j<=p
• 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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function
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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.
• ki = (1-exp(gi))/(1+exp(gi)) i=1~p
• (3) Cepstrum Coefficients

• cn = an + Σk=1n kckan-k/n, 1<=n<=p+1
• = an + Σk=n-pn-1 kckan-k/n, n>p+1
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• (4) The Roots of Predictor

• A(z) = 1 – Σk=1p akz-k = Πk=1p (1-zk z-1) = 0
Σ
• Transfer to S-plane: zi = exp(siT)
• Suppose si = σi + jΩi , zi = zir + jzii , then
• Ωi = tan(zii/zir)/T andσi= log(zii2 + zir2 )/(2T)
• (5)The impulse response of full poles system

• h(n) = Σk=1p akh(n-k)+σ(n) n>=0
• = 0 Get useful
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• (6) Auto-correlated Coefficients of impulse

response of the full poles system
• H(z) = S(z)/U(z) = G/(1- Σk=1p akz-k)
• The auto-correlated coefficients of h(n) is :
• R(i) = Σn=0 h(n)h(n-i) = R(-i)
• It could be proved that :
• R(i) = Σk=1p ak R(|i-k|) 1<=i<=p
• And R(0) = Σk=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 - Σ akz-k
• The impulse response is :
• a(n) = (n) - Σk=1p ak (n-k)
• = 1, n = 0; an , 0<n<=p; 0, otherwise
• Its auto-correlated function is :
• R(i) = Σk=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-Σk=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 expjω:

• P(expjω)=|A(P)(expjω)|expjφ(ω)[1+exp[-j((p+1)ω+
2φ(ω))]
• Q(expjω)=|A(P)(expjω)|expjφ(ω)[1+exp[-j((p+1)ω+
2φ(ω)+π)]
• If the roots of A(P)(z) are inside the unit circle, whenωis
0~π, φ(ω) changes from 0 and returns to 0, the amount
[(p+1)ω+2φ(ω)] will be 0~(p+1)π
• P(expjω)=0 : [(p+1)ω+2φ(ω)]=kπ, k=1,3,…P+1
• Q(expjω)=0 : [(p+1)ω+2φ(ω)]=kπ, k=0,2,…P
• The roots of P and Q : Zk = expjωk [(p+1)ω+2φ(ω)]=kπ,
k=0,1,2,…P+1
• And ω0 < ω1 < ω2 < … < ωP < ωP+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 µp(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 
 , w 

P ( w)   sin( wT / 2) T
 0, 
 w 
 T
1  /T ( wt / 2)
p(t )   e jwt d w
2  / T sin (wT / 2)
2 n1T
1, n0
A  2 n1 p(t )dt  
2
2
T
0, n0
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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
1st Nyquist:
 1st Nyquist:

2nd Nyquist:

2nd Nyquist:

1st Nyquist
1st Nyquist:

1st Nyquist:

2nd Nyquist:

2nd 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
N
0  t  T 
si (t )   sij j (t ),   4.1
(5.5)
j 1 i==1,2,....,M 
coefficient
Energy signal
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• Coefficients:
T i=1,2,....,M 
sij   si (t ) j (t )dt ,   (5.6)
0
 j=1,2,....,M 
• Real-valued basis functions:
T
1 if i  j 
0 i (t ) j (t )dt  ij  0 if i  j  (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    , i  1,2,....,M (5.8)
. 
. 
 
 siN 

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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)
 siT si
2 Matrix
si Transposition
N
=  sij2 , 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 T

definition of average Ei   si2 (t )dt (5.10)
energy in a signal… 0
• Where si(t) is N
si (t )   sij j (t ), (5.5)
j 1

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N T
 N 
• After substitution: Ei    sij j (t )   sikk (t )  dt
0  j 1   k 1 
N N T
• After regrouping: Ei  
j 1
 s s   (t ) (t )dt
k 1
ij ik j k (5.11)
0
• Φj(t) is orthogonal, so N
  2 2
finally we have: E i sij = s i (5.12)
j 1
The energy of a
signal is equal to the
squared length of its
vector
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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 T
i k (5.13)
0
Inner product is invariant

to the selection of basis
Inner product of the functions
signals is equal to the
inner product of their
vector representations
[0,T]
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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:
N
si  s k =  (sij -skj ) 2
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(2f c  2f )t 0  t  Tb (bit  1)

s2 (t )  A cos(2f c  2f )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)
= 2 Eb
cos(2f H t  1 ) 0  t  Tb
Tb
binary 0 sBFSK(t)= vL(t)

2 Eb
= cos(2f Lt   2 ) 0  t  Tb
Tb
• results in phase discontinuities
• discontinuities causes spectral spreading & spurious transmission
Get useful
• not suited for tightly studysystems
designed materials from www.rejinpaul.com
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Continuous Phase FSK

single carrier that is frequency modulated using m(t)
2 Eb
sBFSK(t) = cos(2f ct   (t ))
Tb
2 Eb  t

= cos 2f c t  2k FSK  m( )d 
Tb   
t
where (t) = 2k 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 1 0 1
FSK
Signal
0 1 1
x
a0 0 VCO modulated composite
a1 1 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  nf , n is an integer
PSD decay rate (inversely proportional to spectrum)

1
• PSD decay rate for CP-BFSK 
f 4
1
• PSD decay rate for non CP-BFSK  f 2
f = frequency offset from fc

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Spectrum & Bandwidth of BFSK Signals
Transmission Bandwidth of BFSK Signals (from Carson’s Rule)

• B = bandwidth of digital baseband signal
• BT = transmission bandwidth of BFSK signal
BT = 2f +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 = 2f +(1+) Rb
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General FSK signal and orthogonality
• Two FSK signals, VH(t) and VL(t) are orthogonal if

T
0
 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
T
 VH (t )VL (t )dt  0 V H (t )VH (t )dt  0 ?
0 0

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An FSK signal for 0 ≤ t ≤ Tb
vH(t) = 2 Eb
and vL(t) = 2 Eb
cos( 2 ( f c  f )t ) cos( 2 ( f c  f )t )
Tb Tb
2Eb
then vH(t) vL(t) = cos( 2 ( f c  f )t ) cos( 2 ( f c  f )t )
Tb
=
Eb
cos(2 (2 f c )t )  cos(2 (2f )t )
Tb
T Tb
0 Tb cos(4f ct )  cos(4ft )dt

Eb
and  VH (t )VL (t )dt 
0
Tb
Eb  sin( 4f c t ) sin( 4ft )  Eb  sin( 4f cTb ) sin( 4fTb ) 
=   
4f  0  
=
Tb  4f c Tb  4f c 4f 
vH(t) vL(t) are orthogonal if Δf sin(4πfcTb) = -fc(sin(4πΔf 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
Q
I I
Carrier phases
Carrier phases
{0, /2, , 3/2}
{/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 Q Q
I I I
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 non-
coherent 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
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materials from www.rejinpaul.com
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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 signal’s carrier wave and the
receiver’s 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 Add s(t)

filter
-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,
 t   f  t  cos c t  a sin ct
then the pilots is
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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 = 180º 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

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Co t…
Property 2
2. Entropy H= log2 K when the symbols are equally likely for K
symbols PK= 1/K

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Co t…
Property 3
3. Maximum upper bound on entropy is

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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 η is represented as
η= 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

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Co t…
• 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 Modulator
Digital Source Line X(w)
Control (Transmit
Source Encoder Coding
Coding Filter, etc)
Hc(w) Channel
Transmitter
N(w) Noise
+
Error Demod
Digital Source Line
Control (Receive
Sink Decoder Decoding Y(w)
Decoding Filter, etc)
Receiver

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Error Models
• 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 0 p is the probability of bit
p
error or the Bit Error Rate
IN OUT (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

Source code Codeword
101101 1000
data chopped (n bits)
into blocks Channel
1000 coder
Dataword
(k bits)
Codeword +
Dataword possible errors
(k bits) Channel (n bits)
Channel
decoder
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 X is a valid codeword
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 X is a valid codeword
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 X
This code corrected this way
11011011 O
11010011 O This code corrected this way
11010010 X
X is a valid codeword
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 
 2 
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 0 0 0 1 1 0
0 1 0 0 1 0 1
0 0 1 0 0 1 1
0 0 0 1 1 1 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 
G 
 . . ... .   . 
   
 ak1Get useful akn fromwww.rejinpaul.com
... materials
ak 2 study ak 
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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 k k k
c3   d 3i a i   (d1i  d 2i )a i  d1i a i   d 2i a i
i 1 i 1 i 1 i 1
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 a1 = [1011]
G    a2 = [0101]
 0 1 0 1
• For d = [1 1], then;
1 0 1 1
 0 1 0 1
c 
_ _ _ _
 1 1 1 0

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Linear Block Codes – example 2
• A (6,5) code with

1 0 0 0 0 1
0 1 0 0 0 1
 
G  0 0 1 0 0 1
 
0 0 0 1 0 1

0 0 0 0 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 R R=n-k
1 0 .. 0 p11 p12 .. p1R 

0 1 .. 0 p21 p22 .. p2 R 
G  I | P
.. .. .. .. .. .. .. .. 
 
0 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 Data
000000 0
000001 1
000010 1
000011 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 Get useful study materials from www.rejinpaul.com
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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 is a codeword, then
cn-1 c0 c1 …. cn-3 cn-2
cn-2 cn-1 c0 …. cn-4 cn-3
: : : : :
c1 c2 c3 …. cn-1 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 0 0 0 0 Thus, it is not a cyclic code

1 0 1 1 1 since, for example, the cyclic
C shift of [10111] is [11011]
0 1 1 0 1
   C
1 1 0 1 0
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Example 3
• The (7,4) Hamming code discussed before is
cyclic:
1010001 1110010 0000000 1111111

1101000 0111001
0110100 1011100
0011010 0101110
0001101 0010111
1000110 1001011
0100011 1100101

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Generator matrix of a non-systematic (n,k) cyclic

codes
• The generator matrix will be in this form:
g0 g1  gnk 1 gnk 0 0  0 

0 g0 g1  gnk 1 gnk 0  0 

G0 0 g0 g1  gnk 1 gnk 0 0 
 
  
 0 0 0  0 g0  gnk 1 gnk 
notice that the row are merely cyclic shifts of the r n
basis vector 
g  g0g1 gnk 1gnk 000 
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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 . m g

• 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 0’s and 1’s.
• 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

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The addition and multiplication are as follow:
ax j  bx j  (a  b) x j Where (a+b) and (a.b) are under GF(2). But
(ax j ).(bx k )  (a . b) x j  k j+k is integral addition
Example:
m( x )  m0  m1 x  m2 x 2
g ( x )  g 0  g1 x
   m( x )  g ( x )  (m0  g 0 )  (m1  g1 )x  (m2  0)x 2
addition
   m( x )g ( x )  m0g 0  (m0 g1  m1g 0 )x  (m1g1  m2 g 0 )x 2  m2 g1 x 3

Multipliation
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 (cont’d)

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 (cont’d)

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-r-
1g(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 (cont’d)

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

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(7,4) Code Generated by 1+X+X3
(Cont’d)
Infor. Code Code polynomials
1001 1100101 1 + X + X4 + X6 = (1 + X3) . g(X)
0101 0111001 X+ X2 + X3 + X6 = (X+ X3) . g(X)
1101 1010001 1 + X2 + X6 = (1 + X + X3) . g(X)
0011 0010111 X2 + X4 + X5 + X6 = (X2 + X3). g(X)
1011 1111111 1 + X + X2 + X 3 + X 4 + X 5 + X 6
= (1 + X2 + X3) . g(X)
0111 0100011 X + X5 + X6 = (X + X2 + X3). g(X)
1111 1001011 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) (cont’d)

(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
+ + + + Output
gr gr-1 gr-2 g1 g0
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)
+ + + +
Output
g0 g1 g2 gr-1 gr
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 (cont’d)

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 1 1 0 1
Register : 000 110 101 100 100
initial 1st shift 2nd shift 3rd shift 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 Input Register contents
0 0 0 (initial state) • What is g(x)?
1 0 000 • Find the syndrome using
2 1 100 long division.
3 1 110 • Find the syndrome using
4 0 011 the shortcut for the
remainder.
5 1 011
6 0 111
7 0 1 0 1 (syndrome s)
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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 (cont’d)

(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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(cont’d)
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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(cont’d)
Gate
+
r = 0010110 Gate +
Shift Input Register contents

0 0 0 (initial state)
1 0 000
2 1 100
3 1 110
4 0 011
5 1 011
6 0 111
7 0 1 0 1 (syndrome s)
8 (input gate off) - 1 0 0 (syndrome s (1) )
9 - 0 1 0 (syndrome s (2) )
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(cont’d)
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 Syndrome polynomial.

X6 1 + X2 101
X 1 + X + X2 111
X4 X + X2 011
X3 1+X 110
X2 X2 001
X1 X 010
X0 1 100
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 2 j mod n 
a  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 Corresponding
polynomial
a0) f1X1 + X
(a, a2 , a4 , a8) f2X 1 + X + X4
(a3 , a6 , a9 , a12) f3X 1 + X + X2 + X3 +
X4
(a5 , a10) f4X 1 + X + X2
(a7, a14 , a13 , a11) f5X 1 + X3 + X4
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Design Example (cont’d)

• 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) = f2X f3X = 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 BCH Codes
Linear
Block Codes Hamming Codes
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 burst-
correcting 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 parity-
check 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 k d (designed) d ( actual) g(X)*
7 4 3 3 13
15 11 3 3 23
15 7 5 5 721
15 5 7 7 2463
31 26 3 3 45
31 16 5 7 107657
31 11 7 11 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
by the Rieger bound. r  2b
• 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 k b g(X) (octal)
7 3 2 35 (try to find dmin!)
15 10 2 65
15 9 3 171
31 25 2 161
63 56 2 355
63 55 3 711
511 499 4 10451
1023 1010 4 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
Error Correction Error Correction CRC
CRC To
Encoder Decoder Syndrom
To Info Sink
Encoder e Checker
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, λ, 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 Code Coverage
• where r is the number of check bits
• For some near-optima CRC codes, CRC-12 0.999756
see table 5.6.5 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 (cont’d)

• 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.

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The Viterbi Algorithm:
• 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]

Compute the two possible paths at
Add the weight of the each state and select the one This is called the
path at each state with less cumulative Hamming survival path
Getweight
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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 (cont’d)

• 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 1’s) of the
output corresponding to that branch.
The power of N is the Hamming weight (# of 1’s) 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 (cont’d)

.

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Properties of the Path
Sequence 2:
code sequence: .. 00 11 10 11 00 ..
state sequence: a0 b c a1
Labeled: (D2LN)(DL)(D2L) = D5L3N
Prop. : w =5, dinf =1, diverges from the allzero path by 3
branches.
Sequence 3:
code sequence: .. 00 11 01 01 00 10 11 00 ..
state sequence: a0 b d c b c a1
Labeled: (D2LN)(DLN)(DL)(DL)(LN)(D2L) = D7L6N3
Prop. : w =7, dinf =3, diverges from the allzero path by 6
branches.

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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 (cont’d)

• 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?
y

  mdH y,
   
dH y,
c
  c  
c p
Pr y |   


.(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:
max
cCm
  
c  min
Pr y, 

  
d y, c
cCm H  
 Choose the code sequence through the trellis which has the
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
 There are several code segments
ci
 of n bits which lead
into state j at time i+1. We wish to find the most likely one.
 Let sc i  be the state from which the code segment
ci

emerged
 For each state j, we assume that
ci
 is the path leading
into
Z s 
c i i   d  
c ,y
H i 
state j if:
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
Z i  1  Z
j s
ci






H
 
i   d c i , y
• 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 01 00 10 10 1 1) is sent
 y
 = (0 1 1 1 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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UNIT II WAVEFORM CODING 9

UNIT III BASEBAND TRANSMISSION 9

UNIT IV DIGITAL MODULATION SCHEME 9

UNIT V ERROR CONTROL CODING 9

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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

Sampling (in time) is

 Measure amplitude at regular intervals

 How many times should we sample?

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 where fs is sampling frequency and fm is the

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 low-pass filter removes higher-frequencies

 (e.g.) If max sampling frequency is 22kHz, the it

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• Pulse-Amplitude Modulation (PAM)

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I k : {mk < m ≤ mk +1}, k = 1,2,..., L (17) Fig. 9

– 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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 Many signals such as speech have a nonuniform distribution.

2 Example: Nonuniform 3 bit quantizer

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u-LAW and A-LAW definitions

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Companded Signal -0.5

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