Digital Communications

Dr J.RAVINDRANADH
Professor
E-mail : [email protected]

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 1

 Test Books & Reference Books
 Text Books
 Simon Haykin, Communication Systems, 3rd Edition, John Wiley
& Sons,
 Leon W Couch II, Digital and Analog Communication Systems, Pe
arson, 2004
 Reference Books
 Taub and Schilling, Principles of Communication Syst 2nd Edition
, TMH, 1986
 J Das,S K Mallik and PK Chatterjee, Principles of
Digital Communication,NAI(P), 2000
 Bernard Sklar, Digital Communication, 2nd Edition, Pearson Educ
ation, 2001

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 2

 Syllabus
Unit-I
PULSE MODULATION:
Quantization Process, Quantization Noise, Pulse Code Modulation
Encoding regeneration, decoding , Delta
Modulation, Differential Pulse Code Modulation (DPCM).

BASE BAND PULSE TRANSMISSION:

Matched filter, Properties, Inter symbol interference,
Correlative level, coding Nyquist’z criterion for
distortionless baseband binary transmission, ideal Nyquist
channel Duobinary signaling, Modified Duobinary signaling.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 3

 Unit-II
DIGITAL PASSBAND TRANSMISSION::

Introduction, Pass band transmission model

Gram Schmidt Orthogonalization procedure,
Geometric interpretation of signals
Coherent detection of signals in noise, Probability of error
Correlation receiver, detection of signals with unknown phase,
Coherent BPSK QPSK, BFSK, Non Coherent BFSK, DPSK

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 4

 Unit-III
INFORMATION THEORY:
Uncertainty, Information, Entropy, Properties of Entropy,
Source Coding Theorem, Shannon Fano Coding, Huffman Coding,
Discrete memoryless channels, Mutual information, Properties,

Channel capacity, Channel coding theorem

Differential entropy and mutual information for continuous
ensembles, Information capacity theorem.

Unit-IV
ERROR CONTROL CODING:

Linear Block Codes, Hamming Codes, Cyclic Codes, Convolution Codes

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 5

  Introduction
Communication systems are designed to transmit the
information generated by a source to some destination.
Information source are different forms. For example, in
Radio broadcasting, the source is generally an audio source
(voice or music). In TV broadcasting, the source is video
(moving image)
Two types of sources: analog source and digital source
The outputs of these are analog signals and hence the
sources are called analog sources.
In computers and storage devices( magnetic or optical disks)
produce discrete outputs( Binary, ASCII) and they are
called discrete sources.
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 6
Block diagram of Digital Communication system

SOURCE ENCODER

The Source encoder converts the input i.e. symbol

sequence into a binary sequence of 0s and 1s by assigning
code words to the symbols in the input sequence. The
important parameters of a source encoder are block size,
code word lengths, average data rate and the efficiency
of the coder
7
CHANNEL ENCODER

Error control is accomplished by the channel coding

operation that consists of systematically adding extra
bits to the output of the source coder.
These extra bits do not convey any information but helps
the receiver to detect and / or correct some of the
errors in the information bearing bits.
MODULATOR
The Modulator converts the input bit stream into an
electrical waveform suitable for transmission over the
communication channel. Modulator can be effectively used
to minimize the effects of channel noise, tomatch the
frequency spectrum of transmitted signal with channel
characteristics, to provide the capability to multiplex
many signals
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 8
 CHANNEL
The Channel provides the electrical connection between
the source and destination. The different channels are:
Pair of wires, Coaxial cable, Optical fibre, Radio channel,
Satellite channel or combination of any of these

The important parameters of the channel are Signal to

Noise power Ratio(SNR), Bandwidth . The signal power
decreases due to the attenuation of the channel. The
signal is corrupted by unwanted, unpredictable electrical
signals referred to as noise.

DEMODULATOR:
The extraction of the message from the information bearing
waveform produced by the modulation is accomplished by the
demodulator. The output of the demodulator is bit stream..

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 9

CHANNEL DECODER

The Channel decoder recovers the information bearing

bits from the coded binary stream. Error detection and
possible correction is also performed by the channel
decoder.

SOURCE DECODER:
At the receiver, the source decoder converts the
binary output of the channel decoder into a symbol
sequence. The decoder for a system using fixed –
length code words is quite simple, but the decoder for
a system using variable – length code words will be very
complex.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 10

Certain Issues in Digital Transmission
 Transmission Rate
 This is a measure of the number of bits that can be transmitted over the
communication channel per unit time

 Bandwidth Requirements
 This is a measure of the spectrum that the communication system
requires to transmit the information at the desired transmission rate.

 Error probability
 This represents the percentage of bits that are in error relative to the
overall number of bits that are transmitted by the communication system

 Transmission Power
 This represents the amount of power of the transmitted signal that would
be required to achieve a particular desired error probability

 System Complexity
 This represents that amount of money that a manufacturer will have to
spend to build the system and the amount of money that a user will have
to pay to use the system

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 11

Advantages of Digital Communication
 Digital circuits are more reliable, fast and cheaper
compared to analog circuits.
 The effect of distortion, noise and interference is less
in a digital communication system.
 Signal processing functions like encryption, compression
can be employed to maintain the secrecy of the
information.
 Error detecting and Error correcting codes improve the
system performance by reducing the probability of
error
 Combining digital signals using TDM is simpler than
combining analog signals using FDM
 We can avoid signal jamming using spread spectrum
technique
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 12
Disadvantages of Digital Communication
 Large System Bandwidth
 Digital transmission requires a large system
bandwidth to communicate the same information in a
digital format as compared to analog format.

 System Synchronization
 Digital detection requires system synchronization
whereas the analog signals generally have no such
requirement

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 13

 Sampling
 Digitizing the coordinates values is called sampling
 The process of converting a continuous time
continuous amplitude into discrete time continuous
amplitude signal is called quantization.
 Sampling-rate = how many samples are taken per unit
of each dimension. e.g., samples per second,
frames per second, etc

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 14

 Pulse Modulation systems
 Sampling of the signals is the fundamental operation of
digital communication.
 A continuous time signal is converted to discrete time
signal by sampling process.
 The data is transmitted is sampled at regular intervals.
Such samples are then transmitted directly or through
the modulation of some carrier is called pulse modulation
 The sufficient number of samples of the signal should
be taken so that the original signal represented in its
samples completely and reconstruct the signal
completely from its samples.
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 15
 Sampling Theorem for Low pass signals
 A continuous time signal can be completely represented
in its samples and recovered back if the sampling
frequency fs ≥ 2 fm. Here fs is sampling frequency and
fm is the maximum present in the signal.
Let us consider an arbitrary signal g(t) of finite energy and
infinite duration and g(t) is strictly band limited

Sampling function samples g(t) signal regularly at rate of fs

samples per second

Ts=1/fs= represents sampling period

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 16
The impulse train of pulse shown in figure having sampling
frequency fs
The impulse train of pulse can be expressed as

 (t )     t - nTs )

Multiplying unit impulse with instantaneous value of g(t)
obtain the sampled version of the g(t)
∞

g  (t ) = ∑ g (nTs )  (t - nTs )
-∞

Let g(nTs) represent the instantaneous amplitude of signal

g(t) at instant t=Ts

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 17

 g(t)

 (t - nTs )

g  (t)

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 18

Fourier transform of impulse train
∞

 Ts ( f )  f s ∑   f - n f s )
-∞

Fourier transform of sampled version of the g(t)

∞

G  ( f ) = G( f ) * f s ∑  ( f - n f s )
-∞

Where * denotes the convolution operation. Interchanging the order

of summation and convolution
∞

G  ( f ) = f s ∑ G( f ) *  ( f - n f s )
-∞

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 19

The convolution of frequency function with a delta function
reproduces the function itself
∞

G  ( f ) = f s ∑ G( f - n f s )
-∞

Fourier transform of ideal sampled signal is periodic with a period

equal to 1/Ts.
∞

Consider the ideal sampling signal g  (t ) = ∑ g (nTs )  (t - nTs )

-∞

Taking Fourier transform both side for the above equation

∞

G(f )= ∑ g ( nT s ) e - j2  fnT s
-∞
this is called Discrete Fourier transform

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 20

The Fourier transform of g(t) is G(f)

G( f ) = 0 for f ≥ W

Let the sampling frequency be exactly equal to twice of the maximum

frequency in g(t)
1
f s = 2W or Ts =
2W

Consider the spectrum of g  (t ) is given by

∞

G  ( f ) = f sG ( f ) ∑ f sG( f - n fs )
n = -∞
n≠ 0

this equation shows that same G(f) will be reproduced at f=0, f=

± f s ,± 2 f s ,± 3 f s etc

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 21

 G(f)

G ( f ) for f s = 2W

G ( f ) for f s < 2W

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 22

 G ( f ) for f s > 2W

Nyquist Rate: When the sampling rate becomes exactly equal to 2W

samples per second, for a signal bandwidth of W Hertz, then it is
called Nyquist Rate

Nyquist rate= 2W

Nyquist interval= 1/2W

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 23

 Quantization of signal
 Digitizing the amplitude values is called quantization.

 The process of transforming the a discrete time

continuous amplitude signal into discrete amplitude
discrete time signal is called quantization.

J. Ravindranadh/ Professor /
ECE Dept / Digital
Communication 24
 Quantization Process
 Convert the continuous of signal in to sampled signal
 Divide the sampled version of the signal into L number
of quantized levels.
 Each sample approximate or round of the nearest
quantized level.
 We divide the total range VH – VL into L equal intervals
each of size Δ. Accordingly Δ is called step size and is
given by
VH - VL
=
L

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 25

 gq(t)
3
g(t)

0
n
 gq(t)
Quantized Signal

0
t
It is quite apparent that the quantized signal is not exactly the same
as the original analog signal. There is a fair degree of quantization
error here. However; as the number of quantization levels is
increased the quantization error is reduced and the quantized signal
gets closer and closer to the original signal
Consider a n-bit PCM system where n= number of bits/sample

L represents the number of quantized level L= 2n

Δ represent the step size

Vmax - Vmin

number of level

Δ= peak to peak/ No of levels

Qe represent the quantization error

Quantization error = sampled value- quantized values

[Qe ]max = Maximum error possible =Δ/2

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 28

 Consider a PCM System with 2 bit A/D converter and assume that
signal amplitude varies from 0 to 8V. This is divided into 4 equal
steps as shown in figure.

8
7
6
5
4
3
2
1
0

Sample value Quantized value Quantized error Qe Encoder o/p

0.7 1 -0.3 00
2.3 3 -0.7 01
5.9 5 0.9 10
7.8 7 0.8 11
29
 If the sampled values are 0.8, 2.7, 7.9, Determine the quantizer
output , encoder output and quantization error for each sample

8
7
6
5
4
3
2
1
0

Sample value Quantized value Quantized error Qe Encoder o/p

0.8 1 -0.2 00
2.7 3 -0.3 01
7 7 0 11
7.9 7 0.9 11
30
 Classification of quantization
Quantization

Uniform Quantization Non Uniform Quantization

Mid tread Mid rise Biased

 Uniform Quantization

 The quantization levels are uniformly spaced ( step size is constant)

 Non uniform Quantization
 Non- uniform type the spacing between the levels will be unequal
and mostly the relation is logarithmic.
31
 Mid tread Quantizer
Transfer characteristics of a Mid tread quantizer is shown in below

Quantization error

32
 when the input is between δ/2 to –δ/2 the quantizer
output is zero
For - δ/2 ≤ x(nTs) < δ/2 , xq(nTs) =0
For δ/2 ≤ x(nTs) < 3δ/2 , xq(nTs) =δ

when x(nTs)=0 , xq(nTs) =0, error is zero at the

origin
when x(nTs) is between δ/2 and 3δ/2, xq(nTs) =δ
error is δ/2 - δ = -δ/2
3δ/2 - δ = δ/2
thus the quantized error lies between -δ/2 and δ/2

the maximum quantized error is  Max =
2
Mid – tread type: Quantization levels – odd number.
33
Mid rise Quantizer
Transfer characteristics of a Mid rise quantizer is shown in below

Quantization error

34
 For 0≤ x(nTs) < δ , xq(nTs) =δ/2
For - δ ≤ x(nTs) < 0 , xq(nTs) =-δ/2

quantized error x(nTs) - xq(nTs)

0 - δ/2 =- δ/2
0 - (- δ/2) = δ/2

thus the quantized error lies between -δ/2 and δ/2


the maximum quantized error is  Max =
2
Mid – Rise type: Quantization levels – even number

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 35

 Biased Quantizer
The Mid tread and Mid rise quantizer are rounding quantizers . Biased
quantizer are truncated quantizers

Quantization error

36
 when the input is between 0 to δ the quantizer output is
zero
For 0 ≤ x(nTs) < δ , xq(nTs) =0
For -δ ≤ x(nTs) < 0 , xq(nTs) =-δ

when x(nTs) is between -δ and 0, xq(nTs) =-δ

quantized error = xq(nTs)- x(nTs)

0 - δ = -δ
thus the quantized error lies between -δ and 0

the maximum quantized error is  Max  

The difference between stair step and dotted line give the
quantized error
37
 If the sampled values are 3.8,2.1,0.5, -1.7,-3.2,-4. Determine
the quantizer output , encoder output and quantization error for
each sample Sketch the transfer characteritics of the quantizer
4
111 3.5
3
110 2.5
2
101 1.5
1
100 0.5
0
011 -0.5
-1
010 -1.5
-2
001 -2.5
-3
000 -3.5
-4

38
Sample value Quantized value Quantized error Qe Encoder o/p
3.8 3.5 -0.3 111
2.1 2.5 -0.4 110
0.5 0.5 0 100
-1.7 -1.5 -0.2 010
-3.2 -3.5 -0.3 000
-4 -3.5 -0.5 000

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 39

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 40
 Signal to Noise Ratio
Let input x(nTs) be the continuous amplitude in the
range –mmax to +mmax
Total amplitude range = mmax- -(mmax) = 2mmax

Total Amplitude range 2 mmax

= = (1)
Number of levels L

Where L is the total number of representation level.

The sinusoidal signal swings between +Am and -Am
2 Am
There fore =
L
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 41
 For midtread and midrise the quantized error lies
between -δ/2 and δ/2 
the maximum quantized error is  Max =
2
The PDF quantized error is for mid tread and mid rise

1 - 
f q (q ) = ≤ qe ≤
 2 2
=0 otherwise
-δ/2 δ/2
The average power of the quantizing noise or variance
or mean square value1

 Q2  ∫ q e2 f q( q ) dq
- 
1 2

1 2
 2
= 2
q dq
Q
 - ∫ e =
12
(2)
2 42
Let n denote the number of bits per sample∴ L = 2 n (3)

equivalently n = log 2 L

Hence substituting eq (3) in eq (1) we get the step size

2 mmax
= n (4)
2
1 2
From eq (4) in eq (2) we get the step size  2
Q  m max 2 - 2n
3
Let denote the average power of the message signal m(t).
The output signal to noise ratio of a uniform quantization

P 3P
( SNR ) O  ( SNR ) O  ( 2 ) 2 2n

 Q2 m max
The output signal to noise ratio of the quantization increases
exponentially with increasing number of bits per samples.
43
Case 1: Sinusoidal Modulating Signal

Sinusoidal modulating signal of amplitude Am .

2
Am
The average signal power is P 
12
The average power of the quantizing noise or variance
1 2 - 2n
 2
Q  Am 2
3
The output signal to noise ratio of a uniform quantization

Normalized signal power

SNR =
Normalized Noise power

A m2 2 3
( SNR ) 0 =  (2 2n )
A m2 2 - 2 n 3 2

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 44

The signal to quantizing noise ratio in decibels

3
( SNR ) dB = 10 log + 20 log L
2
= 1.8 + 20 log L (OR) = 1.8 + 6n
Case 2: average signal power is normalized P=1 therefore Am=1

SNR = 3(2 2n )
( SNR ) dB = 10 log 3 + 20 log L = 4.8 + 20 log L
L=512

( SNR ) dB = 4.8 + 54 = 58.8

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 45

A signal whose amplitude varies from 0 to 10V is band limited to
4K Hz and transmitted through the channel using 5 bit PCM. The
sampling rate is 50% higher than the nyquist rate calculate all
parameters of the PCM

if n=5 L  25  32 levels
Vmax - Vmin 10
  V
number of level 32
 10
Maximum error possible  Max  
2 64
Band limited signal W= 4K
Nyquist rate = 2 W= 8000 samples/sec

Sampling rate fs= 8000 + 8000 X 50/100= 12000 Sample/sec

Bit rate = sampling rate x n= 60000 bits/sec = 6Kbps

Band width =60K Hz
46
A video signal is band limited to 4.5K Hz and transmitted through a
channel using PCM a) determine the sampling rate if the signal is
sampled at a rate of20% higher than the nyquist rate b) Determine
the bit rate if the number of quantization levels are 1024

a) Nyquist rate = 2 W= 2x4.5M =9M samples/sec

Sampling rate fs= 9M + 9M X 20/100= 10.8 M Sample/sec

b) Number of levels 2 n  1024

n  10
Bit rate = sampling rate x n=108M sample/sec

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 47

 Bit duration Tb = Ts/ n

Bit rate 1/Tb = 2 W n

Maximum value of message band width = 3.57 M Hz

3
( SNR ) dB = 10 log + 20 log L  1.8  6n  43.8dB
2

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 48

Consider a sinusoidal signal given by s(t) = 3 cos (1000∏t)
(i) Find step size and the signal –to- quantization noise ratio when
the signal is using 10 bit PCM.
(ii) find the minimum number of bits needed to achieve a signal-
to-noise ratio of at least 40dB.

2 Am  5.85  10 3V
(i ) 
L
3
( SNR ) dB = 10 log + 20 log L  1.8  6n  62dB
2
1.8  20 log L  40 L  81.3

2  81.3
n
n=6 , 2n=64 which is less than desire value

n=7 2n=128 which is greater than desire value. Therefore we

required 7 bits
J. Ravindranadh/ Professor /
ECE Dept / Digital
Communication 49
 Pulse Code Modulation(PCM)
 PCM system consist of PCM encoder (transmitter) ,
transmission path and PCM decoder (receiver).
 The output of PCM is in coded digital form of pulse
of constant amplitude, width and position..

Transmitter

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 50

 The signal is first passing through the low pass filter
the cut off frequency W Hz. The LPF blocks all
frequency components above W Hz. This means the
signal is band limited to the W Hz.
 The sample circuit sample the signal at the rate fs ≥
2W to avoid the aliasing. The output of sampling
circuit is denoted x(nTs) which is discrete time and
continuous in amplitude.
 Divide the sampled version of the signal into L number
of quantized levels. Each sample approximate or round
of the nearest quantized level. Thus output of
quantizer is digital level is called xq(nTs).
 Quantized signal xq(nTs) is given the binary encoder
which convert the quantized signal to n- binary digits.
The encoder is an A to D converter which convert each
quantized value into binary data.
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 51
Transmission path

 The path between the PCM transmitter and the PCM

receiver over which PCM signal is travel called PCM
transmission path.
 The output of encoder is binary data which are
represented in the form of pulses and transmitted
through a channel due to noise amplitude distortion
occurs. But regenerative repeater are used to
eliminate the noise from the signal.
 The imported feature of the PCM system lies in its
ability to control the effect of distortion and noise.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 52

PCM Receiver

 Regenerative circuit is used to eliminate the noise

present at the input of the receiver.
 The decoder is a digital to analog converter which
converts the binary data into samples.
 Using sampled and hold circuit is allowed to pass
through a low pass filter to get the appropriate
signal.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 53

 Application
 In Telephone and space communication
 Voice signal 300-3.5KHz
 Nyquist rate 2fm = 7000 sample/sec
 1/Ts= 8000 sample/sec
 Each sample converted into 8 bits so number of levels
28=256
 Audio-CD recording
 Audio 20-20KHz
 Nyquist rate=40000sample/sec
 1/Ts= 44100 samples/sec
 In this encoder used is 16 bit converter so number of level
216
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 54
 Advantage of PCM
 Very high noise immunity
 Due to digital nature of the signal, repeaters can
be placed between the transmitter and the receiver
 It is possible to store PCM signal
 It is possible to use various coding techniques, this
makes the communication secure
Disadvantage of PCM
 The encoding, decoding and quantizing of PCM is
complex
 PCM requires a large bandwidth as comapared to
the other systems
55
 Electrical representation of binary data
 Unipolar NRZ ( ON-OFF signaling)
Symbol 1 is represented by presence of pulse with
constant amplitude and 0 represented by absence of
pulse. This type of signal is called ON-OFF signal.

0 1 1 0 1 0 0 1

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 56

 Polar NRZ
Symbols 1 and 0 are represented by pulse of
positive and negative amplitudes such as 1 is
represented by +1V and 0 represented by by -1V.
This is referred to polar signal or NRZ.

0 1 1 0 1 0 0 1

 NZ code
A rectangular pulse is used for 1 and no pulse for 0
this type of signal is called Return to Zero.

0
57
 Bipolar return-to-zero signaling
Positive and negative pulses of equal amplitude are
used alternatively symbol 1 and no pulse for symbol
0, it is referred as bipolar signal. This format as
advantage that the power of the transmitter signal
does not have any d.c component.
0 1 1 0 1 0 0 1
.
0

 Split phase signaling (Manchester code)

In this method of signaling, symbol 1 is represented
by positive pulse followed by a negative pulse, with
both pulses being of equal amplitude and half-
symbol width. For symbol 0 the polarities of these
two pulses are reversed.
58
 0 1 1 0 1 0 0 1

 Differential coding
Transition is a binary PCM wave may be used to
designate symbol 0 while no transition is used to
designate symbol1. this representation is called
differential encoding.

0 1 1 0 1 0 0 1

59
 Non uniform quantizer
In non uniform quantizer the step size is not fixed , its varies as
for the input signal.
xq(nTs ) Large step size for high inputs

x( nTs )

small step size for low inputs

Step size is high at higher input levels. Step size is small at
smaller input levels , hence quantization error is small. Therefore
signal to noise is improved at low signal levels.
60
 Necessity of non-uniform quantization

The maximum quantization error q eMax =
2
2x 2
The step size  = max =
L L
L = 2n if n=4 L = 2 4 = 16
2 1 1
Step size  = = q eMax =
16 8 16

i.e quantization error is 1/16 th of full scale voltage, full scale

voltage is 16V maximum error =1V

Thus for low amplitude signals 1V,2V,3V the error may be large
values but for upper range it is very small, it is occurs because of
uniform quantization. To avoid these problem non quantization
used

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 61

Model of non uniform quantizer

 In uniform quantization the step size is fixed, so

quantization noise power is constant. But signal
power is not constant, it is proportional to square of
the signal amplitude.
 Hence Signal power is small for weak signals but
noise power is constant. Therefore signal to noise
ratio for weak signals is very poor which affects
the signal quality.
 The weak signals are amplified and strong signals
are attenuated before applying them to a uniform
quantizer

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 62

  At the receiver, a device with a characteristic
complementary to the compressor called Expander is
used to restore the signal samples to their correct
relative level.
 The Compressor and expander take together
constitute a Compander
 Compander = Compressor + Expander
 Non uniform quantization is achieved using
. companding.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 63

Compressor characteristics Expander characteristics

Compressor provides a high gain to weak signal and small gain to

strong signals. Thus weak signals are boosted to improve the
signal to noise ratio.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 64

Compander Characteristics

Compressor

Expander

Expander

Compressor
Above figure shows the Campander characteristics which
is the combination of compressor and expander
characteristics.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 65

Types of companders
(i) μ- Law compander (ii) A- Law compander

(i) μ- Law compander log(1 +  m )

v=
log(1 +  )
Where m and v are the normalized input and output voltages, and
μ is a positive constant.

J. Ravindranadh/ Professor /
ECE Dept / Digital
Communication 66
Uniform quantization μ =0. for a given value of μ the reciprocal
slope of the compression curve, which defines the quantum steps,
is given by the derivative of m with respect to v

d m log(1 +  m )
= (1 +  m )
dv 
μ –law is neither linear or logarithmic, but it is approximately
linear at low input levels corresponding to μ|m|<<1 approximately
logarithmic at high input levels corresponding to μ|m|>>1

(ii) A- Law compander Am 1

0≤ m ≤
1 + log A A
v=
1 + log( A m ) 1
≤ m≤1
1 + log A A

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 67

This characteristic is also piece wise, made up of linear segment
for low level inputs and logarithmic segment for high level inputs

Uniform quantization corresponds to A=1. The reciprocal slope of

this compression curve is given by the derivative of |m| with
respect |v|

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 68

 1 + log A 1
0≤ m ≤
dm A A
=
dv 1
1 + log( A m ) ≤ m≤1
A

A- law companding is used for PCM telephone systems

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 69

 Differential Pulse Code Modulation
 When the voice or video signal sampled at a rate
slightly higher than the nyquist rate the resulting
sampled signal having a high correlation between
adjacent samples.
 The signal does not change rapidly from one sample
to the next, with the result that the difference
between adjacent samples has variance that is
smaller than the variance of the signal itself.
 When these highly correlated sampled are encoded
by PCM, the resulted encoded signal contains
redundant information.
 To eliminate the redundancy DPCM is used
70
 Continuous time signal m(t) by dotted line, the
samples are encoded by using 3 bit PCM.
 The encoded binary value of each sample is written
on the top of the samples.
 Some of the samples are carrying the same
information mean that it is redundant
71
 Block diagram of the DPCM (Transmitter)

Comparator
Sampled e q (nTs ) DPCM
input e(nTs ) wave
m(nT
m( nTs) Quantizer Encoder

m(nTs )

Prediction
filter
m q (nTs )

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 72

 Differential pulse code modulation works on the
principle of prediction.
 The value of the present sample is predicted from
the past samples.
 The prediction may not be exact but it is very close
to the actual sample value.
 The sampled signal is denoted by m(nTs) and

predicted signal is denoted by m^ (nTs).

 The comparator finds out the difference between
. the actual sample value m(nTs) and prediction
sample value m^(nTs). This is known as prediction
error and it is denoted by e(nTs)

e(nTs ) = m(nTs ) - m(nTs ) (1)

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 73

 Prediction value is produced by using prediction filter
 The quantizer output signal and previous prediction is
added and given as input to the prediction filter.
 This makes the prediction more an more close to the
actual sampled signal.
 Quantized error signal is very small it can encoded
using small number of bits, number of bits reduced in
DPCM
The quantizer output can be written as.

e q (nTs ) = e(nTs ) + q (nTs ) (2)

here q(nTs) is the quantization error
the prediction filter input mq(nTs) is obtain by sum
m^(nTs) and quantizer output
∧

m q (nTs ) = m(nTs ) + eq (nTs ) (3) 74

 substituting the value eq(nTs) in above equation
∧

m q (nTs ) = m(nTs ) + e(nTs ) + q (nTs ) (4)

From equation 1
∧

m(nTs ) = m(nTs ) + e(nTs ) (5)

From equation 4 and 5

m q (nTs ) = m(nTs ) + q (nTs ) (6)

above equation represents the quantized version of
the input signal m(nTs) and quantization error
q(nTs)
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 75
 Block diagram of the DPCM (Receiver)

DPCM output
input Decoder

Prediction
filter

 The decode reconstructs the quantized error signal

from incoming binary signal.
 The prediction filter output and quantized error
signal are summed up to give the quantized version
of the original signal.
 The signal at receiver differs from the actual signal
by quantization error.
76
 Delta Modulation
 Delta modulation transmits only one bit per sample.
 The present sample value is compared with the previous
sample value and this result amplitude is increased or
decreased is transmitted.
 The input signal approximated to step signal in delta
modulation . The step size is kept fixed in delta
modulation.
 The difference between the input signal and staircase
approximate signal is confined to two levels +Δ and -Δ
 If the difference is positive, signal is increase by one
step Δ, the difference is negative signal is reduced by
Δ. 77
When the step is reduced ‘0’ is transmitted and if the step is
increased ‘1’ is transmitted. Hence for each sample one bit is
transmitted.

m(t)

Ts
Staircase u(t)

0 0 1 0 1 1 1 1 1 0 1 0 0 0 0 0

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 78

 Block diagram of the DM (Transmitter)

Comparator
Sampled b( nTs ) DM
input e(nTs ) wave
m(nT
m( nTs) Quantizer Encoder
∧

m(nTs )
u{(n - 1)}Ts

Delay
Ts
u (nTs )

Accumulator 79
The error between the sampled value of m(t) and last approximation
sample is given by
∧

e(nTs ) = m(nTs ) - m(nTs )

If we assume u(nTs) as the present sample approximation of
staircase wave
∧

u{(n - 1)}Ts = m(nTs )

Two quantization levels used are +Δ and –Δ the transfer
characteristic of the quantizer is show in fig
o/p

+Δ
b( nTs ) = ∇ sgn{ e(nTs )}
i/p
-Δ

80
Depending on the sign of error e(nTs) the sign of step size decided
∧

b(nTs ) = +∇ if m(nTs ≥ m( nTs )

∧

-∇ if m(nTs ≤ m(nTs )

The summer in the accumulator adds quantize output with previous

sample approximation

u ( nTs ) = u{( n - 1)}Ts + b( nTs )

The previous sample approximation u(n-1)Ts is restored by delaying
one sample period Ts. If the step size is +Δ then binary ‘1’ is
transmitted and if it is –Δ, then binary ‘0’ is transmitted.

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 81

 Block diagram of the DM (Receiver)
Accumulator Demodulated
output
input LPF

Delay
Ts

 The accumulator generates the staircase approximated

signal output and is delayed by one sampling period Ts.
It is added to the input signal
 If the input is binary 1 then it adds +Δ to the
previous output, input is 0 then Δ is subtracted from
the delayed signal.
 Low pass filter smoothens the staircase signal to
reconstruct the original signal.
82
 Advantage of Delta Modulation
 The delta modulation transmits only one bit for one
sample, therefore the signaling rate and
transmission channel bandwidth is quite small for
delta modulation compared to PCM.
 The transmitter and receiver implementation is very
much simple for delta modulation there is no analog
to digital converter required.

 Disadvantage of Delta modulation

 Slope overload distortion
 Granular noise

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 83

 Slope overload Distortion
 This distortion arise because of large dynamic rang
of the input signal.
 The rate of rise of input signal is so high that the
staircase signal cannot approximated it, the step
size become too small.
 There is large error between the staircase
approximated signal and the original input signal.
This error known as slope overload distortion.
 To reduce this error the step size must be
increased then the slope of the signal is high.

Δ d Slope of the message

< m(t) Δ=
Ts dt Sampling rate
84
 Granular Noise
 Granular noise occurs when the step size is too
large compared to small variation in the input signal.
 When the input signal is almost flat, the staircase
signal keeps on oscillating by ±Δ around the signal.
 the error between the input and approximated
signal is called granular noise

Δ d
> m(t)
Ts dt

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 85

The input the delta modulator is m(t)= 5t and the sampling rate is
5000 samples/sec. Determine the step size

Slope of the signal =d/dt m(t)=5

Slope of the message

Δ= =1mV
Sampling rate

The input to the delta modulator is 5 cos2π(1000t) and pulse rate

56000 pulses per second. Determine the step size

Slope of the message 5(2π)(1000)

Δ= = =0.56V
Sampling rate 56000

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 86

The input to the delta modulator is a sinusoidal signal whose
frequency varies from 200Hz to 4000Hz, the sampling rate is 8
times the nyquist rate. The peak voltage of the signal 1V.
Determine the step size when the signal frequency is 800Hz

Nyquist rate 2 fm= 8000

Sampling rate 8(2 fm)= 64000 sample/sec

1(2π)(800)
Δ= =78mV
64000
The input to the delta modulator =is m(t)= 125t[u(t)-u(t-1)+[250-
125t][u(t-1)-u(t-2)] sampling rate is 32000 samples/sec Determine
the step size

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 87

A TV signal with a bandwidth of 4.2MHz is transmitted using
binary PCM. The number of representation level is 512. calculate
(a) codeword length (b) Final bit rate © transmission bandwidth

L = 2n 512 = 2 n ∴ n=9

Final bit rate =2nW =75.6Mbits/sec

Transmission Band width = 75.6M Hz

A signal band limited within 3.6KHz is to be transmitted via binary

PCM on a channel whose maximum pulse rate is 40,000 pulse/sec.
Design a PCM system and draw the a block diagram showing all
parameters
Avoid distortion fs=fb/n
Sampling frequency fs> 2W = 7.2KHz = 8KHz
Number of bits required n=fb/fs = 5.5 =5
The number of quantization level L=2n =32
88
Consider the following sequences of 1s and 0s
(a) An alternating sequence of 1s and 0s
(b) A long sequence of 1s followed by a long sequence of 0s
(c) A long sequence of 1s followed by a long sequence of 0s and
then a long sequence of 1s
Sketch the wave form (a) On-off signaling (b) Bipolar NRZ

1 0 1 0 1 0 1 0 1

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 89

 1 1 1 1 0 0 0 0

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 90

 A sinusoidal signal with an amplitude of 3.25 Volts is applied to
unifrom quantizer of the midtread type whose output takes on the
values 0, +1, +2, +3 Volts. Sketch the waveform of the resulting
quantizer output for one complete cycle of the input

91
 A sinusoidal signal with an amplitude of 3.25 Volts is applied to
unifrom quantizer of the midrise type whose output takes on the
values 0.5,+1.5, +2.5, +3.5 Volts. Sketch the waveform of the
resulting quantizer output for one complete cycle of the input

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 92

J. Ravindranadh/ Professor / ECE Dept / Digital Communication 93
J. Ravindranadh/ Professor / ECE Dept / Digital Communication 94