Digital Communication

Systems
Lecture-2

Formatting

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

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

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

Formatting

Transmit and Receive Formatting

Transition from information source digital symbols
information sink

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

Transmission of Analog Signals

Structure of Digital Communication Transmitter

Analog to Digital Conversion

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), T
s
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

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

Sampling
Convert from continuous-time analog signal xa(t) to discretetime continuous value signal x(n)
Is obtained by taking the samples of x (t) at discrete-time
a
intervals, Ts

Quantization
Convert from discrete-time continuous valued signal to discrete
time discrete valued signal

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
9

Ideal Sampling ( or Impulse Sampling)

Therefore, we have:

1
xs (t ) x(t )
Ts

e jn st

Take Fourier Transform (frequency convolution)

Xs ( f ) X ( f )*
Ts

jn s t

1
X ( f ) * e jn s t
Ts
n

s
1
X s ( f ) X ( f ) * ( f nf s ), f s
Ts
2
n

1
Xs( f )
Ts

1
X ( f nf s )

Ts
n

n
X(f )

Ts
n
10

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

11

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
(t nTs )

Ts
n

jns t

2
s
Ts
12

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

13

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

2
T

x(t ) Ts x(nTs )

(t nTs )
n

x (nTs ) sin c(2 f s (t nTs ))

provided that =>

1
1
Ts
fs
2 fm
14

Ideal Sampling ( or Impulse Sampling)

This

means that the output is simply the replication of the original

signal at discrete intervals, e.g

15

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

16

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 )

Note:

Fourier Transform of impulse train is another impulse train

Convolution with an impulse train is a shifting operation

17

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 )

cn e j 2 nf s t

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

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

X [ f nf s ]
18

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

19

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

20

x '(t ) x(t ) (t )

Flat top sampling (Time Domain)

xs (t ) x '(t ) * p(t )

p (t ) * x(t ) (t ) p (t ) * x(t ) (t nTs )

n

21

Taking the Fourier Transform will result to

X s ( f ) [ xs (t )]

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

1
P( f ) X ( f ) *
Ts

1
P( f )
Ts

( f nf s )

X ( f nf )

where P(f) is a sinc function

22

Flat top sampling (Frequency Domain)

Flat

top sampling becomes identical to ideal sampling as the

width of the pulses become shorter

23

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
24

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
25

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

26

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

27

Aliasing is prevented by forcing the bandwidth of the sampled

signal to satisfy the requirement of the Sampling Theorem

28

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.

29

Summary Of Sampling

xs (t )
Ideal Sampling
(or Impulse Sampling)

x(t ) x (t ) x(t ) (t nTs )

n

Natural Sampling
(or Gating)

x(nT ) (t nT )

xs (t ) x(t ) x p (t ) x(t ) cn e

j 2 nf s t

Flat-Top Sampling

For all sampling techniques

If fs > 2B then we can recover x(t) exactly
If fs < 2B) spectral overlapping known as aliasing will occur

xs (t ) x '(t ) * p(t ) x(t ) (t nTs ) * p (t )

n

30

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?

31

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
32

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

33

See Figure 2.16 (Page 80)

34

35

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
36

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)
37

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 noiselike disturbance at the output of a DAC converter

38

Uniform Quantization

A quantizer with equal quantization level is a Uniform Quantizer

Each sample is approximated within a quantile interval
Uniform quantizers 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 quantizers

q
q
Error of a uniform quantizer is bounded by 2 e 2
39

Signal to Quantization Noise Ratio

The mean-squared value (noise variance) of the quantization error

is given by:

q/2
1

1
2
2 e p(e)de e
de

e
de

q
q q / 2
q / 2
q / 2

q/2

q/2

2
q
1
e
q

3 q / 2 12
3

q/2

40

The peak power of the analog signal (normalized to 1Ohms )can be

expressed as:
2
V pp
L2 q 2
P

2
4
1

V p2

Therefore the Signal to Quatization Noise Ratio is given by:

L2 q 2 / 4
SNRq 2
3L2
q /12

41

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
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
2n ) 6n dB

10log
(2
10
N dB

42

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

43

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

44

Statistics of speech Signal Amplitudes

Figure 2.17: Statistical distribution of single talker speech signal

magnitudes (Page 81)
45

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

46

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

47

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

48

Input/Output Relationship of Compander

Logarithmic expression Y = log X is the most commonly

used compander
This reduces the dynamic range of Y

49

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

y ymax

log e 1 (| x | / xmax
log e (1 )

sgn( x)

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

= 0 corresponds to uniform quantization

50

A-Law Companding Standard (Europe,

China, Russia, Asia, Africa)

ymax

y ( x)

| x|
A
xmax
sgn( x),
(1 A)

ymax

| x |
1 log e A

x
max

sgn( x),
(1 log e A)

| x| 1
0

xmax A

1 | x|

1
A xmax

where
x and y represent the input and output voltages
A = 87.6
A is a constant number determined by experiment

51

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
52

53

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
54

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.,

55

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

56

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

57

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 )

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
58

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
59

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

60

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)
61

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

62

Unipolar NRZ Line Code

Unipolar non-return-to-zero (NRZ) line code is defined by
unipolar mapping

A,
an
0,

when X n 1

when X n 0

Where Xn is the nth data bit

In addition, the pulse shape for unipolar NRZ is:

t
where Tb is the bit period

f (t )

Tb

NRZ Pulse Shape

63

Bipolar Line Codes

With bipolar line codes a space is mapped to zero and a
mark is alternately mapped to -A and +A

It

A,

an A,
0,

when X n 1 and last mark A

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

is also called pseudoternary signaling or alternate mark inversion

(AMI)
Either

RZ or NRZ pulse shape can be used

64

Manchester Line Codes

Manchester line codes use the antipodal mapping and
the following split-phase pulse shape:

f (t )

Tb
T

b
t
t

4
4
T
Tb
b

2
2

65

Summary of Line Codes

66

67

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

68

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

69

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

70

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)
71

Bits per PCM word and M-ary Modulation

Section 2.8.4: Bits per PCM Word and Bits per Symbol

Section 2.8.5: M-ary Pulse Modulation Waveforms

L=2l

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-topeak 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?

72

Solution to Problem 2.14

| e | pV pp
V pp Lq
1
l log 2

2
p

fs 8000

q
| e |max
2
V pp
q
L

1
2 L
2p
l

l log 2 (50) 6
Rs 48000

M 16

R
48000
R2

12000 symbols / sec

log 2 ( M )
4
73