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You are on page 1of 59

(a) Formatting

(b) Sources of Corruption

(c) PCM

(d) Uniform and Non-Uniform

Quantization

Sheraz Alam Khan

Asst. Professor, Department of Engineering

Sheraz.alam@gmail.com

National University of Modern Language

1

Outline

Formatting of Analog Information

Sampling

Sources of Corruption in quantized Signal

PCM

2

Formatting and Source encoding

The goal of the first essential signal step FORMATTING is to ensure

that the message (or source signal) is compatible with digital

processing.

Textual information is transformed in to binary digits by use of a

coder

Analog info in formatted using three separate processes: sampling,

Quantization and coding

Transmit formatting is a transformation from source information to

digital symbols. When data compression in addition to formatting

is employed, the process is termed as source coding.

In all cases the result of formatting is a sequence of binary digits

function 3

Formatting..

Transmit and Receive Formatting

Transition from information source digital symbols information

sink

4

2.2 Formatting Textual Data

A textual information is a sequence of alphanumeric characters. It is

transformed into binary digits by use of a coder.

Alphanumeric and symbolic information are encoded into digital bits

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

When digitally transmitted, the characters are first encoded into, a

sequence of bits, called a bit stream or baseband signal

Group of k bits is called symbol M = 2k

A system using a symbol set size of M is referred to as M-ary system

For example: K=1 means binary, K=2 means quaternary

The value of k or M represents an important initial choice in the

design of any DCS

5

Example of Messages, characters, and symbols

8-ary example

6

Example of Messages, characters, and symbols

32-ary example

7

2.4 Formatting Analog Information

Structure of Digital Communication Transmitter

8

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 9

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 10

Sampling

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

Ifsignal 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 (Impulse) Sampling

Natural Sampling

Flat-Top Sampling

11

1. Ideal Sampling ( or Impulse Sampling)

Consider the case of ideal sampling with a sequence of unit

impulse function.

12

Ideal Sampling..

It is accomplished by the multiplication of the analog signal x(t) by

the uniform train of impulses (comb function)

Consider the instantaneous sampling of the analog signal xs(t), which

is the product of x(t) with a periodic train of unit impulse function

x(t)

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

n

Where Ts is the sampling period and (t) is the unit impulse or Dirac

delta function.

13

Ideal Sampling..

1

2

The Fourier series

n

( t nT s )

Ts

n

e jn s t , s

Ts

1 jnst

xs (t ) x(t ) e

Therefore Ts n

Take Fourier Transform (frequency convolution) which is zero outside the

interval (-fm < f < fm).

jn s t 1

1

Xs( f ) X ( f )* e X ( f )* e s

jn t

Ts n Ts 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

)

14

Ideal Sampling..

This shows that the Fourier Transform of the sampled signal

is the Fourier Transform of the original signal at rate of 1/Ts

15

Ideal Sampling..

This means that the output is simply the replication of the

original signal at discrete intervals, e.g.

16

Ideal Sampling..

will occur in Xs(f)

fs fm fm fs 2 fm

Minimum Sampling Condition:

completely reconstructed from its sampled value x(nTs) with

2 f (t nTs )

sin

x(t ) Ts x(nTs )

2Ts

n (t nT s )

T

n

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

1 1

provided that => Ts 17

fs 2 fm

2. Natural or Practical Sampling

In practice we cannot perform ideal sampling

It is not practically possible 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

18

Practical 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)

x s (t ) x (t ) x p (t )

x (t )

n

c n e j 2 nf s t

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

n

c n [ x ( t ) e j 2 n f s t ]

n

c n X [ f n19f s ]

Practical Sampling..

The sampling here is termed as natural sampling, since the top of each

pulse in the xs(t) sequence retains the shape of its corresponding analog

segment during the pulse interval.

Each pulse in xp(t) has width Ts and amplitude 1/Ts

Xs (f) is the replication of X(f) periodically every fs Hz

Xs (f) is weighted by Cn (Fourier Series Coefficients) of the pulse train,

compared with a constant value in the impulse sampled case.

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

20

3. Flat-Top Sampling (Sample-and-Hold)

Simplest and most popular method

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

21

Flat-Top Sampling.

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

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

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

n 22

Flat-Top Sampling.

Taking the Fourier Transform will result to

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

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

n

1

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

Ts n

1

P( f )

Ts

n

X ( f nf s )

Flat-Top Sampling.

Flat top sampling becomes identical to ideal sampling as the

width of the pulses become shorter(see figure 2.8f)

The most obvious effect of the hold operation is the

significant attenuation of the higher-frequency spectral

replicates (compare figure 2.8f to 2.6f), which is a desired

effect

Additional analog post-filtering is usually required to finish

the filtering process by further attenuating the residual

spectral components located at multiples of the sample rate.

24

Aliasing

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

Else we run into some problems and

signal is not fully recovered

25

Aliasing

Undersampling and Aliasing

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

spectral overlap in the sampled signal

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 26

Solution 1: Anti-Aliasing Analog Filter

bandlimited

Ifthere is a significant amount of energy in frequencies

above half the sampling frequency (fs/2), aliasing will

occur

Aliasingcan be prevented by first passing the analog signal

through an anti-aliasing filter (also called a pre-filter)

before sampling is performed

The anti-aliasing filter is simply a LPF with cutoff

frequency equal to half the sample rate

28

29

Pre-filtering:

Aliasing is

prevented by

forcing the

bandwidth of the

sampled signal to

satisfy the

requirement of

the Sampling

Theorem

Small transition B.W means sharp cut-offs, but trade-off is filter complexity

and cost. Practically transition B.W is 10 to 20% of signal B.W:

30

Engineering version of Nyquist rate:

Aliasing filters results in a loss of some of the signal information

For this reason, the sample rate, cut-off B.W, and the filer type

selected for a particular signal B.W are all interrelated

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.

Why Oversample?

Most economic solution for the task of ADC or DAC

High performance analog equipment is much more costly than DSP

equipment to perform same task

Due to transition period of analog filter, the B.W of O/P signal is

increased by some amount ft

Consequently the sampling rate must be increased to (2fm+ ft )

E.g. for a CD signal having a two sided B.W of 40 kHz, sampled at 44.1

kHz

our intuition is to use analog filters with small transition B.W(sharp

cut-off)but this induces distortion and high cost because of higher

order filters (required for sharp cut-off)

The solution is to use Oversampling

E.g. rather than sampling at 44 kHz (transition B.W 4.1 kHz) implemented

with 10th order elliptic filter.we might choose 176.4 kHz with a

transition B.W of 136.4 kHz implemented with 4rth order elliptic filter

33

Why Oversample?

34

Summary Of Sampling

Ideal Sampling

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

(or Impulse Sampling) n

n

x ( nTs ) (t nTs )

Natural Sampling

(or Gating)

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

n

Flat-Top Sampling

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

For all sampling techniques n

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

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

Example 1:

Consider the analog signal x(t) given by

What is the Nyquist rate for this signal?

Example 2:

Consider the analog signal xa(t) given by

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

Signal interface for Digital System

Not compatible

with digital system

38

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.

Sources of Corruption

Channel Effects

Channel Noise:

Thermal noise, interference from users and circuit switching

transients etc. may induce errors that degrade

reconstructed signal quality

The rapid degradation of the o/p signal quality with channel

induced errors is called threshold effect

A large difference in behavior can occur for a very small

changes in a channel noise level

Intersymbol Interference (ISI):

Channel is always bandlimited thus spreading or dispersing the

signal waveforms passing through it

When the channel B.W is close to the signal B.W, the spreading

will exceed a symbol duration and cause signal pulses to

overlap.this overlapping is termed as ISI

ISI causes system degradationerrors in detection

40

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

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

likely

q q

e

2 2

Most ADCs are implemented using uniform quantizers

Error of a uniform quantizer is bounded by

Signal to Quantization Noise Ratio

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

is given by: q/2 q/2

1 1 q/2

2 e 2 p(e)de e 2 de e 2 de

q / 2 q / 2 q q q / 2

Corresponds to the avg. q/2 2

quantization to noise 1q e q

3

power 3 q / 2 12

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

as: V p2 V pp L2 q 2

2

P

1 2 4

Therefore the Signal to Quantization Noise Ratio is given by:

Ratio of peak signal 2 2

power to avg. S N R q L 2q / 4 3 L 2

quantization noise power q /12

L= number of quantization levels

SNRq improves as a function of the number of quantization levels squared

Signal to Quantization Noise Ratio

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

q pp

L

converter. (n is the number of bits).

S

10 log (2 2 n ) 6 n dB

N dB 10

2.6 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

First, the quantile intervals between the levels must be equal

It is convenient to choose levels that have zero mean

(symmetrical around zero)

45

PCM.

46

47

PCM..

Binary codes are assigned to each sample according to gray

codes ( Single bit change per adjacent symbol)

Increasing the no. of levels reduces the quantization noise

but at what cost?

Delay is avoided for real-time communication thus the

transmission time of each symbol must be same regardless of

how many bits are used to represent the symbol

Hence for more bits per sample, the bits have to move

faster; in other words they must be replaced by skinnier bits

That is; The increased data rate is at the cost of more B.W

since

48

PCM..

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

49

Non-uniform Quantization

Non-uniform 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 non-uniform

distribution, See Figure on next page (Fig. 2.17)

50

Non-uniform Quantization.

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

51

Non-uniform 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

52

Non-uniform quantization using companding

amplitude of one of the signals is compressed

After compression, input to the quantizer will have a more

uniform distribution after sampling

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

53

Non-uniform quantization using companding

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

techniques

US standard called -law companding

European standard called A-law companding

54

Input/output Relationship of Compander

used compander

This reduces the dynamic range of Y

55

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

= 0 corresponds to uniform quantization

56

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 eA

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

57

Questions?

Life.Answers are not!

Thank You!

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