Principles of

Communication Systems
Chapter No. 6
Introduction to Digital Communications
Formatting of Digital Data

Formatting of Analog Information

Sampling

PCM

Sources of Corruption in quantized Signal

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

Good processing techniques are available for digital signals, such

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

Easy to mix signals and data using digital techniques

Performance Metrics
Analog Communication Systems
(t ) m(t )
Metric is fidelity: want m

SNR typically used as performance metric

Digital Communication Systems

Metrics
P pare
b b) rate (R bps) and probability of bit error
( b data
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.
Formatting
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

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)
Transmission of Analog Signals

Structure of Digital Communication Transmitter

Analog to Digital Conversion

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

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

a
time 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 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

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
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 f must be greater than
s
twice the highest signal frequency fmax of the signal
Types of sampling
Ideal Sampling

Natural Sampling

Flat-Top Sampling
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

n
(t nTs )
Ts
e
n
jns t
, s
Ts
Ideal Sampling (or Impulse Sampling)
Therefore, we have: 1
xs (t ) x(t ) e jnst
Ts n

Take Fourier Transform (frequency convolution)

1
1
Xs ( f ) X ( f )*
Ts
e jn s t
X ( f ) * e jn st
Ts

n n

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

1 1 n
Xs( f )
Ts

n
X ( f nf s )
Ts

n
X(f )
Ts
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
Ideal Sampling ( or Impulse Sampling)

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

2Ts
x(t ) Ts x(nTs )
n (t nTs )


T s x(nTs ) sin c(2 f s (t nTs ))
1 1
n Ts
provided that => fs 2 fm
 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
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
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
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
 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
Aliasing is prevented by forcing the bandwidth of the
sampled signal to satisfy the requirement of the
Sampling Theorem
 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.
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?
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
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
See Figure 2.16 (Page 80)
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
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)

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
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

Error of a uniform quantizer is bounded byq q
e
2 2
 Signal to Quantization Noise Ratio

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

is given by:

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

q / 2 q / 2 q / 2

1 e q /2
q 2
Nq q
3

3 q /2 12
 The peak power of the analog signal (normalized to 1 Ohms ) can
be expressed as:

L2 q 2
V p2
P
1 4


Therefore the Signal to Quatization Noise Ratio is given by:

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

 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
Statistics of speech Signal Amplitudes

Figure 2.17: Statistical distribution of single talker speech signal

magnitudes (Page 81)
Nonuniform quantization using
Companding is a method of reducing the number of bits
companding
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
 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

The output of the A/D converter is a set of binary bits
PCM
But Waveform
binary Types entities that have no physical
bits are just abstract
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
 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.,

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
Summary of Major Line Codes
Categories of Line Codes
Bipolar - Send pulse or negative of pulse

Unipolar - Send pulse or a 0

AMI (alternate mark inversion, pseudoternary)

Represent 1 by alternating signed pulses

Generalized Pulse Shapes
NRZ -Pulse lasts entire bit period

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

Unipolar 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
 When the category and the generalized shapes are combined, we have
the following:
Bipolar NRZ:
Wireless, radio, and satellite applications primarily use bipolar

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

 AMI RZ
A unipolar line code, except now we alternate
between positive and negative pulses to send a 1
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)
Commonly Used Line Codes
Bipolar line codes use the antipodal mapping
A, when X n 1
an
A, when X n 0
Bipolar NRZ uses NRZ pulse shape
Bipolar RZ uses RZ pulse shape

Bipolar NRZ

Bipolar RZ
Unipolar NRZ Line Code
Unipolar non-return-to-zero (NRZ) line code is defined by
unipolar mapping
A, when X n 1
an Where Xn is the nth data bit
In addition, the pulse 0, shape
when
for X n 0
unipolar NRZ is:

where Tb is the bit period t

f (t ) , NRZ Pulse Shape
Tb
AMI Line Codes
With AMI line A, codes
when aXspace is mapped to zero and a mark
n 1 and last mark A

isaalternately mapped to 1-Aand
andlast
+Amark A
n A, when X n
0, when X n 0

It is also called pseudo-ternary signaling or alternate mark inversion
(AMI)

RZ-AMI
Manchester Line Codes
Manchester line codes use the antipodal mapping and the
following split-phase pulse shape:
Summary of Line Codes
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

Bipolar 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
Bipolar codes perform better (are more energy efficient) than

Unipolar or AMI 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
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
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, bipolar NRZ, and unipolar RZ all have DC

components
AMI RZ and Manchester NRZ do not have DC components
 Bits per PCM word and M-ary Modulation
Bits per PCM Word and Bits per Symbol
Number of L=2l Number of bits for each
quantization levels quantization level
M-ary Pulse Modulation Waveforms
M = 2k Number of bits for M-ary

Example: The information in an analog waveform, whose maximum

frequency fm=4000Hz, is to be transmitted using a 16-level PAM
system. The quantization distortion 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?
 q
pV pp
Peak-to-Peak Solution to Example 2
V pp
Vpp = 2mp q
L
q
| e | pV pp | e |max
V pp
pV pp
Quantization
2 2L
1
distortion V pp 1 L
2p
V pp Lq q 2 L
l

L 2p
p = 0.01
1
l log 2 l log 2 (50) 6
2 p
fs 8000 Rs 48000 M 16
R 48000
R2 12000symbols / sec
log 2 ( M ) 4
Analog Transmission of
Digital Data
ASK
FSK
PSK
QPSK
QAM
Signal Constellation
Acknowledgement

This material is collection from:

CSE 3213, Fall 2010 Instructor: N. Vlajic