Digital Communication

Lecture-1
INTRODUCTION
Course Books
Text: Digital Communications: Fundamentals and Applications,
By Bernard Sklar, Prentice Hall, 2nd ed, 2001.

Taubs Principles of Communication Systems

3rd edition
Herbert Taub , donald L Schilling , Goutam Saha

Digital Communication
R.N. Mutagi ( Oxford Press )

Probability and Random Signals for Electrical Engineers, Neon Garcia

References:
Digital Communications, Fourth Edition, J.G. Proakis, McGraw Hill, 2000.
Course Outline
Review of Probability
Signal and Spectra (Chapter 1)
Formatting and Base band Modulation (Chapter 2)
Base band Demodulation/Detection (Chapter 3)
Channel Coding (Chapter 6, 7 and 8)
Band pass Modulation and Demod./Detect.
(Chapter 4)
Spread Spectrum Techniques (Chapter 12)
Synchronization (Chapter 10)
Source Coding (Chapter 13)
Fading Channels (Chapter 15)
Todays Goal

Review of Basic Probability

Digital Communication Basic
Question Bank :

Q-1 Explain basic communication system

5
Communication

Main purpose of communication is to transfer information

from a source to a recipient via a channel or medium.

Basic block diagram of a communication system:

Source Transmitter Channel Receiver

Recipient
Brief Description

Source: analog or digital

Transmitter: transducer, amplifier, modulator, oscillator, power
amp., antenna
Channel: e.g. cable, optical fibre, free space
Receiver: antenna, amplifier, demodulator, oscillator, power
amplifier, transducer
Recipient: e.g. person, (loud) speaker, computer
 Types of information
Voice, data, video, music, email etc.

Types of communication systems

Public Switched Telephone Network (voice,fax,modem)
Satellite systems
Radio,TV broadcasting
Cellular phones
Computer networks (LANs, WANs, WLANs)
Information Representation
Communication system converts information into electrical
electromagnetic/optical signals appropriate for the transmission
medium.
Analog systems convert analog message into signals that can
propagate through the channel.

Digital systems convert bits(digits, symbols) into signals

Computers naturally generate information as characters/bits

Most information can be converted into bits
Analog signals converted to bits by sampling and quantizing
(A/D conversion)
Q 2 Why we use Digital Signal ?

10
 Why digital?
Digital techniques need to distinguish between discrete symbols
allowing regeneration versus amplification

Good processing techniques are available for digital signals, such

as medium.
Data compression (or source coding)
Error Correction (or channel coding)(A/D conversion)
Equalization
Security

Easy to mix signals and data using digital techniques

Q- 3 Draw the complete Digital communication system block diagram
and explain each blocks in brief.

13
 Basic Digital Communication Transformations
Formatting/Source Coding
Transforms source info into digital symbols (digitization)
Selects compatible waveforms (matching function)
Introduces redundancy which facilitates accurate decoding
despite errors
It is essential for reliable communication
Modulation/Demodulation
Modulation is the process of modifying the info signal to
facilitate transmission
Demodulation reverses the process of modulation. It
involves the detection and retrieval of the info signal
Types
Coherent: Requires a reference info for detection
Noncoherent: Does not require reference phase information
Basic Digital Communication Transformations
Coding/Decoding
Translating info bits to transmitter data symbols
Techniques used to enhance info signal so that they are
less vulnerable to channel impairment (e.g. noise, fading,
jamming, interference)
Two Categories
Waveform Coding
Produces new waveforms with better performance
Structured Sequences
Involves the use of redundant bits to determine the
occurrence of error (and sometimes correct it)
Multiplexing/Multiple Access Is synonymous with resource
sharing with other users
Frequency Division Multiplexing/Multiple Access
(FDM/FDMA
Q -4 Which are the basic performance metrics for digital communication
system ? ( Advantages )

19
Performance Metrics

Analog Communication Systems

Metric is fidelity: want m (t ) m(t )
SNR typically used as performance metric

Digital Communication Systems

Metrics are data rate (R bps) and probability of bit error
Pb p( b b)
Symbols already known at the receiver
Without noise/distortion/sync. problem, we will never
make bit errors
Main Points
Transmitters modulate analog messages or bits in case of a DCS
for transmission over a channel.
Receivers recreate signals or bits from received signal (mitigate
channel effects)
Performance metric for analog systems is fidelity, for digital it is
the bit rate and error probability.
Q 5 Why we prefer Digital communication system ?

22
Why Digital Communications?
Easy to regenerate the distorted signal
Regenerative repeaters along the transmission path can
detect a digital signal and retransmit a new, clean (noise
free) signal
These repeaters prevent accumulation of noise along the
path
This is not possible with analog communication
systems
Two-state signal representation
The input to a digital system is in the form of a
sequence of bits (binary or M_ary)
Immunity to distortion and interference
Digital communication is rugged in the sense that it is more
immune to channel noise and distortion
Why Digital Communications?
Hardware is more flexible
Digital hardware implementation is flexible and
permits the use of microprocessors, mini-processors,
digital switching and VLSI
Shorter design and production cycle
Low cost

The use of LSI and VLSI in the design of components

and systems have resulted in lower cost
Easier and more efficient to multiplex several digital
signals
Digital multiplexing techniques Time & Code
Division Multiple Access - are easier to implement
than analog techniques such as Frequency Division
Multiple Access
Why Digital Communications?
Can combine different signal types data, voice, text, etc.
Data communication in computers is digital in nature
whereas voice communication between people is analog in
nature
The two types of communication are difficult to combine
over the same medium in the analog domain.
Using digital techniques, it is possible to combine
both format for transmission through a common
medium
Encryption and privacy techniques are easier to
implement
Better overall performance
Digital communication is inherently more efficient than
analog in realizing the exchange of SNR for bandwidth
Digital signals can be coded to yield extremely low rates
and high fidelity as well as privacy
Q-6 Explain disadvantages of Digital communication system

26
Why Digital Communications?
Disadvantages
Requires reliable synchronization
Requires A/D conversions at high rate
Requires larger bandwidth
Nongraceful degradation
Performance Criteria
Probability of error or Bit Error Rate
Q-7 Why designing digital communication system what care we must take ?

28
Goals in Communication System Design

To maximize transmission rate, R

To maximize system utilization, U
To minimize bit error rate, Pe
To minimize required systems bandwidth, W
To minimize system complexity, Cx
To minimize required power, Eb/No
Q-8 Compare Digital and analog communication systems.

Q- 9 What is the advantage of digital over analog communication system ?

Comparative Analysis of Analog and

Digital Communication
Digital Signal Nomenclature

Information Source
Discrete output values e.g. Keyboard
Analog signal source e.g. output of a microphone
Character
Member of an alphanumeric/symbol (A to Z, 0 to 9)
Characters can be mapped into a sequence of binary digits
using one of the standardized codes such as
ASCII: American Standard Code for Information Interchange
EBCDIC: Extended Binary Coded Decimal Interchange Code
Digital Signal Nomenclature

Digital Message
Messages constructed from a finite number of symbols; e.g., printed
language consists of 26 letters, 10 numbers, space and several
punctuation marks. Hence a text is a digital message constructed from
about 50 symbols
Morse-coded telegraph message is a digital message constructed from
two symbols Mark and Space
M - ary
A digital message constructed with M symbols
Digital Waveform
Current or voltage waveform that represents a digital symbol
Bit Rate
Actual rate at which information is transmitted per second
Digital Signal Nomenclature

Baud Rate
Refers to the rate at which the signaling elements are

transmitted, i.e. number of signaling elements per

second.

Bit Error Rate

The probability that one of the bits is in error or simply

the probability of error

Q-10 Classification of signals.

34
1.2 Classification Of Signals
1. Deterministic and Random Signals
A signal is deterministic means that there is no uncertainty with
respect to its value at any time.

Deterministic waveforms are modeled by explicit mathematical

expressions, example:
x(t) = 5Cos(10t)
A signal is random means that there is some degree of
uncertainty before the signal actually occurs.

Random waveforms/ Random processes when examined over a

long period may exhibit certain regularities that can be described
in terms of probabilities and statistical averages.
2. Periodic and Non-periodic Signals

A signal x(t) is called periodic in time if there exists a constant

T0 > 0 such that

x(t) = x(t + T0 ) for - < t < (1.2)

t denotes time
T0 is the period of x(t).
3. Analog and Discrete Signals

An analog signal x(t) is a continuous function of time; that is, x(t)

is uniquely defined for all t

A discrete signal x(kT) is one that exists only at discrete times; it

is characterized by a sequence of numbers defined for each time,
kT, where
k is an integer
T is a fixed time interval.
4. Energy and Power Signals

The performance of a communication system depends on the

received signal energy; higher energy signals are detected more
reliably (with fewer errors) than are lower energy signals

x(t) is classified as an energy signal if, and only if, it has nonzero
but finite energy (0 < Ex < ) for all time, where:
T/2

lim
2
Ex = x (t) dt = x 2 (t) dt (1.7)
T T / 2

An energy signal has finite energy but zero average power.

Signals that are both deterministic and non-periodic are classified

as energy signals
4. Energy and Power Signals

Power is the rate at which energy is delivered.

A signal is defined as a power signal if, and only if, it has finite
but nonzero power (0 < Px < ) for all time, where
T/2
1

2
Px = lim
T T T / 2
x (t) dt (1.8)

Power signal has finite average power but infinite energy.

As a general rule, periodic signals and random signals are

classified as power signals
5. The Unit Impulse Function

Dirac delta function (t) or impulse function is an abstractionan

infinitely large amplitude pulse, with zero pulse width, and unity
weight (area under the pulse), concentrated at the point where its
argument is zero.

(t) dt = 1

(1.9)

(t) = 0 for t 0 (1.10)

(t) is bounded at t 0 (1.11)

Sifting or Sampling Property

x(t ) (t-t 0 )dt = x(t 0 ) (1.12)
Q -11 Define Spectral Density

41
1.3 Spectral Density

The spectral density of a signal characterizes the distribution of

the signals energy or power in the frequency domain.

This concept is particularly important when considering filtering in

communication systems while evaluating the signal and noise at
the filter output.

The energy spectral density (ESD) or the power spectral density

(PSD) is used in the evaluation.
1. Energy Spectral Density (ESD)

Energy spectral density describes the signal energy per unit

bandwidth measured in joules/hertz.
Represented as x(f), the squared magnitude spectrum
x( f ) X ( f )
2
(1.14)
According to Parsevals theorem, the energy of x(t):

x 2 (t) dt =
2
Ex = |X(f)| df (1.13)
Therefore: - -

Ex =
-
x (f) df (1.15)
The Energy spectral density is symmetrical in frequency about
origin and total energy of the signal x(t) can be expressed as:

E x = 2 x (f) df (1.16)
0
2. Power Spectral Density (PSD)

The power spectral density (PSD) function Gx(f ) of the periodic

signal x(t) is a real, even, and nonnegative function of frequency
that gives the distribution of the power of x(t) in the frequency
domain.
PSD is represented as:
G x (f ) =
|Cn |2 ( f nf 0 )
n=-
(1.18)
Whereas the average power of a periodic signal x(t) is
represented as: 1 0
T /2
Px
T0 x 2 (t) dt |C |
n=-
n
2
(1.17)
T0 / 2
Using PSD, the average normalized power of a real-valued
signal is represented as:

Px G

x (f) df 2 G x (f) df
0
(1.19)
Q-12 Explain Autocorrelation

45
1.4 Autocorrelation
1. Autocorrelation of an Energy Signal

Correlation is a matching process; autocorrelation refers to the

matching of a signal with a delayed version of itself.
Autocorrelation function of a real-valued energy signal x(t) is
defined as:

R x ( ) =

x(t) x (t + ) dt for - < < (1.21)

The autocorrelation function Rx() provides a measure of how

closely the signal matches a copy of itself as the copy is shifted
units in time.
Rx() is not a function of time; it is only a function of the time
difference between the waveform and its shifted copy.
1. Autocorrelation of an Energy Signal

The autocorrelation function of a real-valued energy signal has

the following properties:

R x ( ) =R x (- ) symmetrical in about zero

R x ( ) R x (0) for all maximum value occurs at the origin

R x ( ) x (f) autocorrelation and ESD form a

Fourier transform pair, as designated
by the double-headed arrows
value at the origin is equal to
the energy of the signal
R x (0)

x 2 (t) dt
2. Autocorrelation of a Power Signal

Autocorrelation function of a real-valued power signal x(t) is

defined as:
T /2
1
R x ( ) lim
T

T T / 2
x(t) x (t + ) dt for - < < (1.22)

When the power signal x(t) is periodic with period T0, the
autocorrelation function can be expressed as
T0 / 2
1
R x ( )
T0
T0 / 2
x(t) x (t + ) dt for - < < (1.23)
2. Autocorrelation of a Power Signal

The autocorrelation function of a real-valued periodic signal has

the following properties similar to those of an energy signal:

R x ( ) =R x (- ) symmetrical in about zero

R x ( ) R x (0) for all maximum value occurs at the origin

R x ( ) Gx (f) autocorrelation and PSD form a

Fourier transform pair
T0 / 2
1 value at the origin is equal to the
R x (0)
T0
T0 / 2
x 2 (t) dt average power of the signal
Digital Communication Systems
Lecture-2

PULSE MODULATION AND DIGITAL

TRANSMISSION OF ANALOG SIGNAL

50
Q-13 What is formatting ?

51
Formatting

52
Example 1:
In ASCII alphabets, numbers, and symbols are encoded using a 7-
bit code

A total of 27 = 128 different characters can be represented using

a 7-bit unique ASCII code (see ASCII Table, Fig. 2.3)

53
Formatting
Transmit and Receive Formatting
Transition from information source digital symbols
information sink

54
 Character Coding (Textual Information)
A textual information is a sequence of alphanumeric characters

Alphanumeric and symbolic information are encoded into digital bits

using one of several standard formats, e.g, ASCII, EBCDIC

55
Q-13 Explain process of sampling with neat sketches.

56
Transmission of Analog Signals

Structure of Digital Communication Transmitter

Analog to Digital Conversion

57
Sampling
Sampling is the processes of converting continuous-time analog
signal, xa(t), into a discrete-time signal by taking the samples at
discrete-time intervals
Sampling analog signals makes them discrete in time but still
continuous valued
If done properly (Nyquist theorem is satisfied), sampling does not
introduce distortion
Sampled values:
The value of the function at the sampling points

Sampling interval:
The time that separates sampling points (interval b/w samples), Ts

If the signal is slowly varying, then fewer samples per second will
be required than if the waveform is rapidly varying
So, the optimum sampling rate depends on the maximum
frequency component present in the signal

58
 Analog-to-digital conversion is (basically) a 2 step process:
Sampling

Convert from continuous-time analog signal xa(t) to discrete-

time continuous value signal x(n)
Is obtained by taking the samples of xa(t) at discrete-time

intervals, Ts

Quantization
Convert from discrete-time continuous valued signal to discrete

time discrete valued signal

59
Sampling

Sampling Rate (or sampling frequency fs):

The rate at which the signal is sampled, expressed as the
number of samples per second (reciprocal of the sampling
interval), 1/Ts = fs

Nyquist Sampling Theorem (or Nyquist Criterion):

If the sampling is performed at a proper rate, no info is lost about
the original signal and it can be properly reconstructed later on
Statement:

If a signal is sampled at a rate at least, but not exactly equal to

twice the max frequency component of the waveform, then the
waveform can be exactly reconstructed from the samples
without any distortion

f s 2 f max

60
Sampling

If Rs < 2B, aliasing (overlapping of the spectra) results

If signal is not strictly bandlimited, then it must be passed through

Low Pass Filter (LPF) before sampling

Fundamental Rule of Sampling (Nyquist Criterion)
The value of the sampling frequency fs must be greater than twice

the highest signal frequency fmax of the signal

Types of sampling
Ideal Sampling

Natural Sampling

Flat-Top Sampling

61
Q-14 Describe different sampling techniques with neat sketches.

62
Ideal Sampling ( or Impulse Sampling)

Is accomplished by the multiplication of the signal x(t) by the uniform

train of impulses (comb function)
Consider the instantaneous sampling of the analog signal x(t)

Train of impulse functions select sample values at regular intervals

xs (t ) x(t ) (t nTs )
n

Fourier Series representation:

1
2 Ref: ex 2.12 BPL

n
(t nTs )
Ts
e
n
jns t
, s
Ts
Ch .2

63
Ideal Sampling ( or Impulse Sampling)
1 jnst
Therefore, we have: xs (t ) x(t ) e
Ts n
Take Fourier Transform (frequency convolution)

jnst 1
1
X s ( f ) X ( f )* e X ( f )* e jn s t

Ts n Ts n

1
s
X s ( f ) X ( f )* ( f nf s ), f s
Ref :

2
next
Ts n slide

1 1 n
Xs ( f )
Ts

n
X ( f nf s )
Ts

n
X( f )
Ts

64
65
Ideal Sampling ( or Impulse Sampling)

This shows that the Fourier Transform of the sampled signal is the
Fourier Transform of the original signal at rate of 1/Ts

66
Ideal Sampling ( or Impulse Sampling)

This shows that the Fourier Transform of the sampled signal is the
Fourier Transform of the original signal at rate of 1/Ts

69
Ideal Sampling ( or Impulse Sampling)

This means that the output is simply the replication of the original
signal at discrete intervals, e.g

70
Ideal Sampling ( or Impulse Sampling)

As long as fs> 2fm,no overlap of repeated replicas X(f - n/Ts) will

occur in Xs(f)
Minimum Sampling Condition: fs fm fm fs 2 fm
Sampling Theorem: A finite energy function x(t) can be completely
reconstructed from its sampled value x(nTs) with
2 f (t nTs )
sin

x(t ) Ts x(nTs )
2Ts

n (t nTs )


T s x(nTs ) sin c(2 f s (t nTs ))
1 1
n Ts
provided that => fs 2 fm

71
 Ts is called the Nyquist interval: It is the longest time interval that can
be used for sampling a bandlimited signal and still allow
reconstruction of the signal at the receiver without distortion

72
Practical Sampling

In practice we cannot perform ideal sampling

It is practically difficult to create a train of impulses
Thus a non-ideal approach to sampling must be used
We can approximate a train of impulses using a train of very thin
rectangular pulses:

t nTs
x p (t )
n

Note:
Fourier Transform of impulse train is another impulse train
Convolution with an impulse train is a shifting operation

73
Natural Sampling
If we multiply x(t) by a train
of rectangular pulses xp(t),
we obtain a gated waveform
that approximates the ideal
sampled waveform, known
as natural sampling or
gating (see Figure 2.8)
xs (t ) x(t ) x p (t )

x(t )
n
cn e j 2 nf s t

X s ( f ) [ x(t ) x p (t )]


n
cn [ x (t )e j 2 nf s t ]

c
n
n X [ f nf s ]

74
 Each pulse in xp(t) has width Ts and amplitude 1/Ts
The top of each pulse follows the variation of the signal being
sampled
Xs (f) is the replication of X(f) periodically every fs Hz
Xs (f) is weighted by Cn Fourier Series Coeffiecient
The problem with a natural sampled waveform is that the tops of the
sample pulses are not flat
It is not compatible with a digital system since the amplitude of each
sample has infinite number of possible values
Another technique known as flat top sampling is used to alleviate
this problem

75
 Flat-Top Sampling

Here, the pulse is held to a constant height for the whole

sample period
Flat top sampling is obtained by the convolution of the signal
obtained after ideal sampling with a unity amplitude
rectangular pulse, p(t)
This technique is used to realize Sample-and-Hold (S/H)
operation
In S/H, input signal is continuously sampled and then the
value is held for as long as it takes to for the A/D to acquire
its value
Effect of the hold operation is the significant attenuation of
the higher frequency spectral replicates.

76
 Flat top sampling (Time Domain)
x '(t ) x(t ) (t )
xs (t ) x '(t )* p(t )


p(t )* x(t ) (t ) p(t )* x(t ) (t nTs )
n
77
 Taking the Fourier Transform will result to

X s ( f ) [ xs (t )]


P( f ) x(t ) (t nTs )
n
1

P( f ) X ( f ) * ( f nf s )
Ts n

1
P( f )
Ts
X ( f nf )
n
s

where P(f) is a sinc function

78
 Flat top sampling (Frequency Domain)

Flattop sampling becomes identical to ideal sampling as the

width of the pulses become shorter

79
Q 15 Explain the method of reconstruction of signal from sampled signal.

Q- 16 What is aliasing ? How we can avoid aliasing ?

80
Recovering the Analog Signal
One way of recovering the original signal from sampled signal Xs(f)
is to pass it through a Low Pass Filter (LPF) as shown below

If fs > 2B then we recover x(t) exactly

Else we run into some problems and signal
is not fully recovered

81
 Undersampling and Aliasing
If the waveform is undersampled (i.e. fs < 2B) then there will be

spectral overlap in the sampled signal

The signal at the output of the filter will be

different from the original signal spectrum

This is the outcome of aliasing!

This implies that whenever the sampling condition is not met, an
irreversible overlap of the spectral replicas is produced

82
 This could be due to:
1. x(t) containing higher frequency than were
expected
2. An error in calculating the sampling rate
Under normal conditions, undersampling of signals causing
aliasing is not recommended
83
 Solution 1: Anti-Aliasing Analog Filter

All physically realizable signals are not completely bandlimited

If there is a significant amount of energy in frequencies above
half the sampling frequency (fs/2), aliasing will occur
Aliasing can be prevented by first passing the analog signal
through an anti-aliasing filter (also called a prefilter) before
sampling is performed
The anti-aliasing filter is simply a LPF with cutoff frequency
equal to half the sample rate

84
 Case 2

Aliasing is prevented by forcing the bandwidth of the sampled

signal to satisfy the requirement of the Sampling Theorem

Case 1 85
 Solution 2: Over Sampling and Filtering in the Digital
Domain
The signal is passed through a low performance (less costly)
analog low-pass filter to limit the bandwidth.
Sample the resulting signal at a high sampling frequency.

The digital samples are then processed by a high

performance digital filter and down sample the resulting
signal.

The alias frequency is given by

fa = fs / 2 ( f fs / 2 ) = fs - f

86
Example : A 5.5 kHz tone is sampled at 8 KHz . Find the alias frequency
generated.

Solution :

Here,

f = 5.5 kHz and fs = 8 kHz

From

fa = f s / 2 ( f fs / 2 ) = fs f

fa = 8000 / 2 ( 5500 8000 / 2 ) = 8000-5500 = 2500 = 2.5 kHz

87
Q-17 Explain the method of sampling band pass signals.

88
Sampling of Bandpass Signals
X(f) (a)

X(f) f

(b)
fs 2fs f
X(f)
(c)

X(f)
(d)
f
(a) band-pass signal (b) signal sampled at fs>2f2
(c) Signal sampled at fs>2(f2-f1) (d) Band-pass filter required for signal recovery
89
 A band-pass signal occupies a frequency band from f1 to f2.
It has one sided spectrum as shown in figure (a).
Sampling a band-pass signal at Nyquist rate spectrum is shown in fig (b).
Clearly there are gaps in this spectrum.
To avoid the spectrum overlap we can reduce the sampling frequency.
Sampling frequency so arranged avoid overlapping of the spectrum fig(c).
Minimum sampling frequency rate for a band-pass signal from the f1 to f2 ,
with bandwidth B=f2-f1 is given by

Min fs = 2f2 / N where N is integer part of the ratio f2/B

FDM signals and sub-band signals used in speech coding are examples of
band-pass signals.

90
Example : a FM signal at 10.7 MHz IF needs to be digitized for demodulation
in a digital domain. If the bandwidth of the signal is 200 kHz, find the
minimum usable sampling frequency

Solution :

The Bandwidth B= f2-f1= 200 kHz.

The center frequency , fIF = (f1+f2)/2 = 10,700 kHz.
Therefore the highest frequency f2 = 10,700 + 100 = 10,800 kHz
N = f2/B = 10,800/200 = 54
Minimum sampling frequency fs = 2*f2/N = 2 * (10,800 / 54 ) = 400 kHz

91
Example : A triangular waveform with 10 ms period is to be digitized. If the
waveform fidelity is to be maintained up to its 10th harmonic, what should be
the sampling frequency ?

Solution :

The fundamental frequency of the triangular waveform is

f0 = 1/T = 1 / ( 10 x 10-3 ) = 100 Hz

The 10th harmonic is , therefore

fm = 10 * f0 = 10 x 100 = 1000 Hz

Hence the sampling frequency is

fs = 2 x 1000 = 2000 Hz

92
Summary Of Sampling

Ideal Sampling xs (t ) x(t ) x (t ) x(t ) (t nTs )
(or Impulse Sampling) n

x(nT ) (t nT )
n
s s

Natural Sampling

(or Gating)
xs (t ) x(t ) x p (t ) x(t ) cn e j 2 nf s t

Flat-Top Sampling


xs (t ) x '(t )* p(t ) x(t ) (t nTs ) * p(t )
n
For all sampling techniques
If fs > 2B then we can recover x(t) exactly

If fs < 2B) spectral overlapping known as aliasing will occur

93
Example 1:
Consider the analog signal x(t) given by
x(t ) 3cos(50 t ) 100sin(300 t ) cos(100 t )
What is the Nyquist rate for this signal?
Example 2:
Consider the analog signal xa(t) given by
xa (t ) 3cos 2000 t 5sin 6000 t cos12000 t
What is the Nyquist rate for this signal?
What is the discrete time signal obtained after sampling, if
fs=5000 samples/s.
What is the analog signal x(t) that can be reconstructed from the
sampled values?

94
Practical Sampling Rates

Speech
- Telephone quality speech has a bandwidth of 4 kHz
(actually 300 to 3300Hz)
- Most digital telephone systems are sampled at 8000
samples/sec
Audio:
- The highest frequency the human ear can hear is
approximately 15kHz
- CD quality audio are sampled at rate of 44,000
samples/sec
Video
- The human eye requires samples at a rate of at
least 20 frames/sec to achieve smooth motion

95
Q-18 Explain PCM communication system with necessary blocks.

96
Pulse Code Modulation (PCM)

Pulse Code Modulation refers to a digital baseband signal that is

generated directly from the quantizer output
Sometimes the term PCM is used interchangeably with quantization

97
Q-19 What is quantization ?

98
 See Figure 2.16 (Page 80)
Natural samples, quantized samples and pulse code modulation 99
100
Each quantized sample is represented by a word
consisting of three bits in the example. Space between
words (i.e. samples) allow multiplexing.

101
Q-20 Explain PCM system and mention its advantage

102
 Pulse Code Modulation

Figure The basic elements of a PCM system.

EE 541/451 Fall 2006
Advantages of PCM:
Relatively inexpensive

Easily multiplexed: PCM waveforms from different

sources can be transmitted over a common digital
channel (TDM)
Easily regenerated: useful for long-distance
communication, e.g. telephone
Better noise performance than analog system

Signals may be stored and time-scaled efficiently (e.g.,

satellite communication)
Efficient codes are readily available

Disadvantage:
Requires wider bandwidth than analog signals

104
Q- 21 Mention different source of corruption in sampling and quantization
process.

105
2.5 Sources of Corruption in the sampled,
quantized and transmitted pulses

Sampling and Quantization Effects

Quantization (Granularity) Noise: Results when quantization
levels are not finely spaced apart enough to accurately
approximate input signal resulting in truncation or rounding error.

Quantizer Saturation or Overload Noise: Results when input

signal is larger in magnitude than highest quantization level
resulting in clipping of the signal.

Timing Jitter: Error caused by a shift in the sampler position. Can

be isolated with stable clock reference.
Channel Effects
Channel Noise

Intersymbol Interference (ISI)

106
 Channel Noise : Thermal noise , interference from other users and
interference from circuit switching transients can cause errors in detecting the
pulses carrying the digitized samples.

Channel induced error degrades the reconstructed signal quality.

Rapid degradation of output signal quality with channel induced error is called
threshold effect.

Intersymbol Interference : The channel is always band limited .

A band limited channel disperses or spreads a pulse waveform passing
through it.
Channel BW > Pulse BW then spreading will slight

When channel BW close to signal BW the spreading will exceed a symbol

duration and cause signal pulses to overlap.

This is called Inter symbol interference , ISI which degrades the system
performance

Rising the signal power can not overcome the error performance
107
Signal to Quantization Noise Ratio
The level of quantization noise is dependent on how close any
particular sample is to one of the L levels in the converter

For a speech input, this quantization error resembles a noise-

like disturbance at the output of a DAC converter

108
Q- 22 Explain uniform quantization

Q- 23 Explain non uniform quantization

Q-24 Compare Uniform and non uniform quantization

109
Uniform Quantization

A Quantizer with equal quantization level is a Uniform Quantizer

Each sample is approximated within a quantile interval
Uniform Quantizer are optimal when the input distribution is
uniform
i.e. when all values within the range are equally
likely

Most ADCs are implemented using uniform Quantizer

q
Error of a uniform quantizer is bounded by e
q
2 2
110
Q 25 What is quantization noise ?

Q-26 What is quantization error ?

Q-27 Derive the equation for Quantization error

111
 Figure illustrates L- level
quantizer for a signal having
peak to peak voltage range Vpp
= Vp (- Vp) = 2Vp volts.

The step size between

quantization levels, called the
QUANTILE interval , is denoted
by q volts.

Each sample value is

approximated with a quantized
pulse ; the approximation will
result in error no larger than q/2
or q/2 in positive and negative
direction respectively.

112
 Signal to Quantization Noise Ratio

A useful figure of merit for the uniform quantizer is the quantizer

variance.
The mean-squared value (noise variance) of the quantization error is
given by:

2
q/2 1 q/2q/2
2 e p(e)de e
2
q de
1
e 2
de
q / 2 q / 2 q q / 2

1 e q 2
q/2
q
3

3 q / 2 12
Where p(e) =1/q is the uniform probability density function of the quantization
error.
113
Q-28 Derive the equation for signal to quantization noise ratio or SQNR

114
 The variance 2 , corresponds to the average quantization noise
power.
The peak power of the analog signal (normalized to 1 )can be
expressed as:
L x q = 2 Vp= Vpp
2
V pp L2 q 2
2
Vp
P Vp = L x q / 2
1 2 4
Therefore the Signal to Quantization Noise Ratio is given by:

L2 q 2 / 4
SNRq 2 3L2 L=2n
q /12
L is no of quantization levels.

For perfect understanding refer next few slides 115

Average signal power of the applied input signal is given by

Vp
1
S i m (t ) m (t )
2 2
dm
V p
2V p

1 m 3 Vp
1 V
3
V
2

2
p p
(1)
2V p 3 V 2V p
3 3
p

The quantization noise is

L2
NQ (2)
12
The number of quantization levels is L then L x q = 2Vp

So that (3)
Vp = L x q / 2

116
Q-29 Derive the equation for signal to quantization noise ratio or SQNR

Ref : Taub and Shilling ( Principles of communication systems)

117
Using (1) , (2) and (3)

Vp
2
Lq
2
L q
2 2

2 4 4
Si
32 2
4V p
2
4
L2

NQ q q q2 q2
12

Since L = 2n Si
22n
NQ

Si
in dB 10 log 10 2 2 n 6n

NQ
dB

118
If q is the step size, then the maximum quantization error that can
occur in the sampled output of an A/D converter is q
V pp
q Vpp= 2V= qL
L
where L = 2n is the number of quantization levels for the converter.
(n is the number of bits).

Since L = 2n, SNR = 22n or in decibels

S


10log (22n ) 6n dB
N dB 10
log 10 2 0.3

119
Q-30 Explain PWM and PPM modulation and demodulation with necessary
sketches.

Q-31 Explain different pulse modulation techniques with necessary

sketches.

120
 Pulse Modulation
Recall that analog signals can be represented by a sequence of discrete
samples (output of sampler)
Pulse Modulation results when some characteristic of the pulse (amplitude,
width or position) is varied in correspondence with the data signal

Two Types:
Pulse Amplitude Modulation (PAM)
The amplitude of the periodic pulse train is varied in proportion to the
sample values of the analog signal
Pulse Time Modulation
Encodes the sample values into the time axis of the digital signal
Pulse Width Modulation (PWM)
Constant amplitude, width varied in proportion to the signal

Pulse Duration Modulation (PDM)

sample values of the analog waveform are used in determining the
width of the pulse signal

121
122
Pulse Width Modulation (PWM) and
Pulse Position Modulation (PPM) - 1
In PWM, message modulates the width of the pulse and in
PPM, message modulates position of the fixed width pulse.
These are not suitable for TDM.
Pulse Width Modulation (PWM) and
Pulse Position Modulation (PPM) - 2
Pulse Width Modulation (PWM) and Pulse Position Modulation (PPM)

PWM the message modulates the width of the pulse .

PPM the position of the arrival of a fixed width pulse in each sample period
is modulated by the message signal.

Disadvantage : the randomness of the width in PWM and randomness in

position in PPM , not suitable for time division multiplexing scheme.

As a consequence their contribution to communication is limited.

PWM finds its application in motor control, in delivery of power which is

precisely regulated by regulating the width of the pulse. ( Power electronics ,
Chopper )

Together PWM and PPM are known as Pulse Time Modulation or PTM.

125
 Pulse Width Modulation (PWM)

We have a comparator , one input of which is fed by input message signal and
the other by a sawtooth signal which operated at a carrier frequency.

The maximum of the input signal ( both +ve and ve side) should be less than
that of sawtooth signal.

126
 Output of comparator will be PWM wave

PWM pulses occur at regular interval, its rising edge coinciding with the falling
edge sawtooth signal.

127
 When sawtooth signal as its minimum, which is always less than
the minimum of input signal the +Ve input of the comparator is at
higher potential and the comparator output is positive.

When the sawtooth signal rises with a fixed slope and crosses input
signal value the Ve input of comparator is at higher potential and the
comparator output will be Ve.

The duration for which the comparator stays at high is thus

dependent on input signal magnitude and this decides the width of the
pulse generated.

Message information gets reflected in the time during comparator

output is at HIGH ( +Ve) or the width of the pulse generated at its
output which is directly proportional to amplitude of the message signal
at that instant.

128
Pulse Position Modulation (PPM)

PPM generation is usually a post processing of PWM signal and is shown

in figure (b)

PWM signal generated as above is sent to an inverter which reveres polarity of

the pulses.

it is followed by a differentiator , we will have +Ve spike where in original PWM

signal pulse was going from HIGH to LOW and Ve spikes where LOW to HIGH.

129
 these spikes are then fed to a +Ve edge triggered fixed width
pulse generator which generates pulses of fixed width when a +Ve
spike appeared , coinciding with the falling edge of original PWM
signal.

130
 the occurrence of these falling edges were dependent
( proportional to amplitude of message) on input message and hence
the delay in occurrence of these fixed width pulses are proportional to
the amplitude of the message at that instant.

IC 710 Comparator, The inverter differentiator block of fig(b)

designed using op-amp and RC components.

The fixed width pulse generator can be a monoshot device like IC

74121 , IC 555 etc.

Demodulation of PWM
For PWM demodulation , start a ramp at the positive edge and stop it when
the negative edge comes.

Since the widths are different these ramps will reach different heights in
each cycle which is directly proportional to pulse width and in turn the
amplitude of the modulating signal.

131
 This when passed through a low pass filter will follow the envelope i.e.
message signal and the demodulation is done.

PPM Demodulation

Similar scheme is employed , now the ramp starts at one positive edge of
the pulse and stops at the positive edge of the next pulse.

Thus the delay between the pulses decides the height of the ramp
generated and in turn closely follows the modulating message amplitude.

Low pass filter after that filters out the envelope information as
demodulated signal.

Transistor and RC combination can be used both for ramp generation and
filtering to implement a demodulator circuit.

If synchronous clock is available then PPM can be converted to PWM ( fig)

and then PWM demodulator can be used to get the message signal back.

Between PWM and PPM , the latter gives better performance in a noisy
system
132
 SR edge triggered flip-flop is set by +Ve edge of the clock.

It remains set so that output Q is High, till a +ve edge from PPM resets it.

The more the delay in arrival , the longer the duration Q remains high.

It is again set in the next clock period by the rising edge of clock pulse.

Thus the output of the flip-flop is a train of pulses , the width of which is
decided by how late PPM pulses arrive in a particular clock period in which
again the message information is contained.

133
Pulse Width Modulation (PWM) and
Pulse Position Modulation (PPM) - 3
Q-32 Explain quantization and quantization error. Derive the necessary
question for quantization error. ( Ref : Taub and Schilling )

135
Quantization for Digital Representation-1
Understanding of quantization from Taub and Schilling
Quantization error : the quantized signal and the original signal from which it was
derived differ from one another in a random manner.

This difference or error may be viewed as a noise due to the quantization process
and is called QUANTIZATION ERROR.
2
Calculation of Quantization error : e where e is the difference between
original and quantized signal voltage.

Let us divide total peak-to-peak range of the message signal m(t) into M equal
voltage intervals, each of magnitude S volts.

At the center of the each voltage interval we locate a quantization level

m1,m2,m3,., mM as shown in figure (a) ( next slide).

Dashed level instantaneous value of the message signal m(t) at a time t.

-- m(t) happens to be closest o the level mk , the quantizer output will be mk, the
voltage corresponding to that level.

The error is e = m(t) mk

Quantization for Digital Representation-2

(a) Right : Range of voltage over which

signal m(t) makes excursion is divided into
M quantization range each of size S.

(b) Left : The error voltage e(t) as a function

of the instantaneous value of he signal
m(t).
Quantization for Digital Representation-3

Quantization error :
f(m) is pdf and considered
constant for each quantization
level

Substituting x=(m-mk)

f(x) S is the probability that the signal voltage m(t) will

be in the all quantization range so the sum of terms in
the parenthesis has a total value of unity
Q-33 Explain COMPANDING ? What is the advantage ?

Q- 34 Explain different companding techniques .

140
 COMPANDING
The dynamic range can be improved by companding i.e.
by first compressing and then expanding. A small
amplitude signal will range through more quantization
region.

141
Nonuniform Quantization
Nonuniform quantizers have unequally spaced levels
The spacing can be chosen to optimize the Signal-to-Noise Ratio

for a particular type of signal

It is characterized by:
Variable step size

Quantizer size depend on signal size

142
 Many signals such as speech have a nonuniform distribution
See Figure on next page (Fig. 2.17)
Basic principle is to use more levels at regions with large probability
density function (pdf)
Concentrate quantization levels in areas of largest pdf
Or use fine quantization (small step size) for weak signals and
coarse quantization (large step size) for strong signals

143
Statistics of speech Signal Amplitudes

Figure 2.17: Statistical distribution of single talker speech signal

magnitudes (Page 81)
144
145
Nonuniform quantization using companding
Companding is a method of reducing the number of bits required in
ADC while achieving an equivalent dynamic range or SQNR
In order to improve the resolution of weak signals within a converter,
and hence enhance the SQNR, the weak signals need to be
enlarged, or the quantization step size decreased, but only for the
weak signals
But strong signals can potentially be reduced without significantly
degrading the SQNR or alternatively increasing quantization step size
The compression process at the transmitter must be matched with an
equivalent expansion process at the receiver

146
 The signal below shows the effect of compression, where the
amplitude of one of the signals is compressed
After compression, input to the quantizer will have a more uniform
distribution after sampling

At the receiver, the signal is

expanded by an inverse
operation
The process of COMpressing
and exPANDING the signal is
called companding
Companding is a technique
used to reduce the number of bits
required in ADC or DAC while
achieving comparable SQNR

147
 Basically, companding introduces a nonlinearity into the signal
This maps a nonuniform distribution into something that more

closely resembles a uniform distribution

A standard ADC with uniform spacing between levels can be used

after the compandor (or compander)

The companding operation is inverted at the receiver

There are in fact two standard logarithm based companding

techniques
US standard called -law companding

European standard called A-law companding

148
Input/Output Relationship of Compander

Logarithmic expression Y = log X is the most commonly

used compander
This reduces the dynamic range of Y

149
Types of Companding
-Law Companding Standard (North & South
America, and Japan)

log e 1 (| x | / xmax
y ymax sgn( x)
log e (1 )

where
x and y represent the input and output voltages

is a constant number determined by experiment

In the U.S., telephone lines uses companding with = 255

Samples 4 kHz speech waveform at 8,000 sample/sec
Encodes each sample with 8 bits, L = 256 quantizer levels
Hence data rate R = 64 kbit/sec ( 8,000 x 8 = 64,000)
= 0 corresponds to uniform quantization

150
A-Law Companding Standard (Europe, China, Russia, Asia,
Africa)
| x|
A
xmax | x| 1
ymax sgn( x), 0
(1 A) xmax A
y ( x)
| x|
1 log e A
xmax 1 | x|
ymax sgn( x), 1
(1 log e A) A xmax
where
x and y represent the input and output voltages

A = 87.6

A is a constant number determined by experiment

151
152
153
154
Q- Explain different line codes

155
 PCM Waveform Types
The output of the A/D converter is a set of binary bits
But binary bits are just abstract entities that have no physical definition
We use pulses to convey a bit of information, e.g.,

In order to transmit the bits over a physical channel they must be

transformed into a physical waveform
A line coder or baseband binary transmitter transforms a stream of bits
into a physical waveform suitable for transmission over a channel
Line coders use the terminology mark for 1 and space to mean 0
In baseband systems, binary data can be transmitted using many kinds of
pulses

156
 There are many types of waveforms. Why? performance criteria!
Each line code type have merits and demerits
The choice of waveform depends on operating characteristics of a
system such as:
Modulation-demodulation requirements

Bandwidth requirement

Synchronization requirement

Receiver complexity, etc.,

157
Goals of Line Coding (qualities to look for)
A line code is designed to meet one or more of the following goals:

Self-synchronization

The ability to recover timing from the signal itself

That is, self-clocking (self-synchronization) - ease of clock lock

or signal recovery for symbol synchronization
Long series of ones and zeros could cause a problem

Low probability of bit error

Receiver needs to be able to distinguish the waveform associated

with a mark from the waveform associated with a space
BER performance

relative immunity to noise

Error detection capability

enhances low probability of error

158
 Spectrum Suitable for the channel
Spectrum matching of the channel

e.g. presence or absence of DC level

In some cases DC components should be avoided

The transmission bandwidth should be minimized

Power Spectral Density

Particularly its value at zero

PSD of code should be negligible at the frequency near zero

Transmission Bandwidth
Should be as small as possible

Transparency
The property that any arbitrary symbol or bit pattern can be
transmitted and received, i.e., all possible data sequence should
be faithfully reproducible

159
Line Coder

The input to the line encoder is

the output of the A/D converter
or a sequence of values an that
is a function of the data bit
The output of the line encoder
is a waveform:

s(t ) a
n
n f (t nTb )

where f(t) is the pulse shape and Tb is the bit period (Tb=Ts/n for n
bit quantizer)
This means that each line code is described by a symbol mapping
function an and pulse shape f(t)
Details of this operation are set by the type of line code that is
being used

160
Summary of Major Line Codes
Categories of Line Codes
Polar - Send pulse or negative of pulse

Unipolar - Send pulse or a 0

Bipolar (a.k.a. alternate mark inversion, pseudoternary)

Represent 1 by alternating signed pulses

Generalized Pulse Shapes

NRZ -Pulse lasts entire bit period

Polar NRZ

Bipolar NRZ

RZ - Return to Zero - pulse lasts just half of bit period

Polar RZ

Bipolar RZ

Manchester Line Code

Send a 2- pulse for either 1 (high low) or 0 (low high)

Includes rising and falling edge in each pulse

No DC component
161
 When the category and the generalized shapes are combined, we have
the following:
Polar NRZ:
Wireless, radio, and satellite applications primarily use Polar
NRZ because bandwidth is precious
Unipolar NRZ
Turn the pulse ON for a 1, leave the pulse OFF for a 0

Useful for noncoherent communication where receiver cant

decide the sign of a pulse
fiber optic communication often use this signaling format

Unipolar RZ
RZ signaling has both a rising and falling edge of the pulse

This can be useful for timing and synchronization purposes

162
 Bipolar RZ
A unipolar line code, except now we alternate
between positive and negative pulses to send a 1
Alternating like this eliminates the DC component

This is desirable for many channels that cannot

transmit the DC components
Generalized Grouping
Non-Return-to-Zero: NRZ-L, NRZ-M NRZ-S

Return-to-Zero: Unipolar, Bipolar, AMI

Phase-Coded: bi-f-L, bi-f-M, bi-f-S, Miller, Delay

Modulation
Multilevel Binary: dicode, doubinary

Note:There are many other variations of line codes (see Fig. 2.22,
page 80 for more)

163
Commonly Used Line Codes
Polar line codes use the antipodal mapping
A, when X n 1
an
A, when X n 0
Polar NRZ uses NRZ pulse shape
Polar RZ uses RZ pulse shape

164
Unipolar NRZ Line Code
Unipolar non-return-to-zero (NRZ) line code is defined by
unipolar mapping
A, when X n 1
a Where Xn is the nth data bit
when X n 0
n
0,
In addition, the pulse shape for unipolar NRZ is:
where Tb is the bit period f (t ) t , NRZ Pulse Shape

Tb

165
Bipolar Line Codes
With bipolar line codes a space is mapped to zero and
a mark is alternately mapped to -A and +A
A, when X n 1 and last mark A

an A, when X n 1 and last mark A
0, when X n 0

It
is also called pseudoternary signaling or alternate mark inversion
(AMI)
Either RZ or NRZ pulse shape can be used

166
Manchester Line Codes
Manchester line codes use the antipodal mapping
and the following split-phase pulse shape:
Tb Tb
t 4 t 4
f (t ) T
Tb
b
2 2

167
Summary of Line Codes

168
169
Comparison of Line Codes

Self-synchronization
Manchester codes have built in timing information because they

always have a zero crossing in the center of the pulse

Polar RZ codes tend to be good because the signal level always

goes to zero for the second half of the pulse

NRZ signals are not good for self-synchronization

Error probability
Polar codes perform better (are more energy efficient) than

Unipolar or Bipolar codes

Channel characteristics
We need to find the power spectral density (PSD) of the line

codes to compare the line codes in terms of the channel

characteristics

170
Comparisons of Line Codes
Different pulse shapes are used
to control the spectrum of the transmitted signal (no DC value,

bandwidth, etc.)
guarantee transitions every symbol interval to assist in symbol timing

recovery
1. Power Spectral Density of Line Codes (see Fig. 2.23, Page 90)
After line coding, the pulses may be filtered or shaped to further
improve there properties such as
Spectral efficiency

Immunity to Intersymbol Interference

Distinction between Line Coding and Pulse Shaping is not easy

2. DC Component and Bandwidth

DC Components

Unipolar NRZ, polar NRZ, and unipolar RZ all have DC components

Bipolar RZ and Manchester NRZ do not have DC components

171
 First Null Bandwidth
Unipolar NRZ, polar NRZ, and bipolar all have 1st null bandwidths
of Rb = 1/Tb
Unipolar RZ has 1st null BW of 2Rb
Manchester NRZ also has 1st null BW of 2Rb, although the
spectrum becomes very low at 1.6Rb

172
Generation of Line Codes

The FIR filter realizes the different pulse shapes

Baseband modulation with arbitrary pulse shapes can be
detected by
correlation detector

matched filter detector (this is the most common detector)

173
 Bits per PCM word and M-ary Modulation
Section 2.8.4: Bits per PCM Word and Bits per Symbol
L=2l

Section 2.8.5: M-ary Pulse Modulation Waveforms

M = 2k

Problem 2.14: The information in an analog waveform, whose

maximum frequency fm=4000Hz, is to be transmitted using a 16-level
PAM system. The quantization must not exceed 1% of the peak-to-
peak analog signal.
(a) What is the minimum number of bits per sample or bits per PCM
word that should be used in this system?
(b) What is the minimum required sampling rate, and what is the
resulting bit rate?
(c) What is the 16-ary PAM symbol Transmission rate?

174
 Solution to Problem 2.14

q
| e | pV pp | e |max
2
V pp 1
V pp Lq q 2 L
l

L 2p
1
l log 2 l log 2 (50) 6
2p
fs 8000 Rs 48000 M 16
R 48000
R2 12000symbols / sec
log 2 ( M ) 4

175