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Quantization

Introduction

• Digital representation of analog signals

Digital

Analog Waveform

signal

signal Coding

destination

source (Codec)

Analog-to-Digital

Digital Encoding

Advantages of Digital Transmissions

Noise immunity

Error detection and correction

Ease of multiplexing

Integration of analog and digital data

Use of signal regenerators

Data integrity and security

Ease of evaluation and measurements

More suitable for processing ……..

Disadvantages of Digital Transmissions

Additional hardware for encoding/decoding

A Typical Digital Communication Link

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 retrie

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

Practical Aspects of Sampling

1. Sampling Theorem

2 .Methods of Sampling

4. Anti-aliasing Filter

Sampling

continuous analog signal, xa(t), into a discrete-time si

by taking the “samples” at discrete-time

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 wavef

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

time analog signal xa(t) to discrete-time continuous value signal x

– Is obtained by taking the “samples” of xa(t) at discrete-time

discrete intervals, Ts

Quantization

– Convert from discrete-time

time continuous valued signal to discrete time discrete valued signal

Sampling

he rate at which the signal is sampled, expressed as the number of samples per second

eciprocal of the sampling interval), 1/Ts = fs

the sampling is performed at a proper rate, no info is lost about the original signal and it can be

operly reconstructed later on

atement:

“If a signal is sampled at a rate at least, but not exactly equal to twice the max frequency

onent of the waveform, then the waveform can be exactly reconstructed from the samples

without any distortion”

f s 2 f max

….. Sampling Theorem

Sampling Theorem for Bandpass Signal - If an analog information

signal containing no frequency outside the specified bandwidth W

Hz, it may be reconstructed from its samples at a sequence of points

spaced 1/(2W)

W) seconds apart with zero-mean

zero squared error.

The minimum sampling called the Nyquist interval, that is, Ts =

rate of (2W) samples per 1/(

/(2W).

second, for an analog The phenomenon of the presence of

signal bandwidth of W Hz, high

high-frequency component in the

is called the Nyquist rate. spectrum of the original analog signal is

called aliasing or simply foldover.

Sampling Theorem

Sampling Theorem for Baseband Signal - A baseband signal having

no frequency components higher than fm Hz may be completely

recovered from the knowledge of its samples taken at a rate of at

least 2 fm samples per second, that is, sampling frequency fs ≥ 2 fm.

The minimum sampling components higher than fm Hz is

rate fs = 2 fm samples per completely described by its sample

second is called the values at uniform intervals less than or

equal to 1/(2fm) seconds apart, that is,

Nyquist sampling rate. the sampling interval Ts ≤ 1/(2fm)

seconds.

Methods of Sampling

Ideal

sampling -

an

impulse at

each

sampling

instant

Ideal 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)

x s (t ) x (t )

n

(t n Ts )

1 jn s t 2

n

(t n T s )

Ts

n

e , s

Ts

Ideal Sampling ( or Impulse Sampling)

his shows that the Fourier Transform of the sampled signal is the Fourier Transform of the original

nal at rate of 1/Ts

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

ampled value x(nTs) with

2 f (t n Ts )

s in

2 T s

(t ) Ts x ( n Ts )

n (t n Ts )

T s x ( n T s ) s in c ( 2 f s ( t n T s ) )

1 1

n Ts

provided that => fs 2 fm

Ideal Sampling ( or Impulse Sampling)

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

Ts is called the Nyquist interval: It is the longest time interval that can be used for sampling a ban

signal and still allow reconstruction of the signal at the receiver without distortion

….. Methods of Sampling

Natural

sampling - a

pulse of

short width

with varying

amplitude

with natural

tops

Natural Sampling

Natural Sampling

pulses xp(t), we obtain a gated wavefo

that approximates the ideal sampled

waveform, known as natural sampling

gating (see Figure 2.8)

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

j 2 nfs

x (t )

n

cne

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

j2

n

c n [ x (t )e

n

cn X [ f nfs

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

possible values

Another technique known as flat top sampling is used to alleviate this problem

….. Methods of Sampling

Flat-top

sampling - a

pulse of

short width

with varying

amplitude

with flat

tops

top Sampling

Flat-top

Flat-Top

Top Sampling

ere, the pulse is held to a constant height for the whole sample period

at top sampling is obtained by the convolution of the signal obtained after ideal

mpling with a unity amplitude rectangular pulse, p(t)

is technique is used to realize Sample-and

and-Hold (S/H) operation

S/H, input signal is continuously sampled and then the value is held for as long as i

kes to for the A/D to acquire its value

Flat top sampling (Time Domain)

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

x s ( t ) x '( t ) * p ( t )

p ( t ) * x ( t ) ( t ) p ( t ) * x ( t ) ( t n T s )

n

Taking the Fourier Transform will result to

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

P ( f ) x (t ) (t n T s )

n

1

P( f ) X ( f )*

Ts

( f nfs )

n

1

P( f )

Ts

n

X ( f nfs )

Flat top sampling (Frequency Domain)

Flattop sampling becomes identical to ideal sampling as the width of the pulses become

shorter

Recovering the Analog Signal

One way of recovering the original signal from sampled signal Xs(f) is to pass it through a Low Pass

ilter (LPF) as shown below

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

Significance of Sampling Rate

spectral

components of

adjacent samples

will overlap,

known as aliasing

An Illustration of Aliasing

Undersampling and Aliasing

– If the waveform is undersampled (i.e. fs < 2B)) then there will be spectral overlap in the sample

signal

different from the original signal spectrum

his implies that whenever the sampling condition is not met, an irreversible overlap of the spectral

licas 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

– 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 (als

called a prefilter)) before sampling is performed

– The anti-aliasing

aliasing filter is simply a LPF with cutoff frequency equal to half the sample rate

Antialiasing Filter

An anti-aliasing

filter is a low-pass

filter of sufficient

higher order

which is

recommended to

be used prior to

sampling.

Minimizing Aliasing

• 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

low filter to lim

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.

Summary Of Sampling

Ideal Sampling

(or Impulse Sampling)

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

n

(t n T s )

n

x ( n T s ) ( t n T

Natural Sampling

(or Gating)

j 2 n

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

n

cne

Flat-Top Sampling

x s ( t ) x '( t ) * p ( t ) x ( t ) ( t n T s ) * p

For all sampling techniques n

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

– If fs < 2B) spectral overlapping known as aliasing

liasing will occur

Quantization

Quantization is a non linear transformation which maps elements from a continuous

o a finite set. It is also the second step required by A/D conversion.

ontinuous time - Discrete time

ontinuous value - Discrete time - Discrete value

- Continuous value

Uniform Quantization

output w2(t)

V

-V V

input w1(t)

-V

on of operation

For M=2n levels, step size :

= 2V /2n = V(2-n+1)

Figure 3.10 Two types of quantization: (a)

( midtread and (b) midrise.

Quantization Error, e

output w2(t)

V

-V V

input w1(t)

-V

Error, e

/2

-/2 input w1(t)

Error is symmetric

around zero. 0

3

1

V

2

22

2

2 2 2

V 2 n 1

2

V 2 2n

2 V V

e ( s ) ds

0

x dx

3 12 12

3

2

Suppose the input signal is a triangula r wave between V and V .

V 2

Then the average signal power is .

3

S

2 2n

N out

Definition. The dynamic range of an input signal is the ratio of the largest to

the smallest power levels which the input signal can take on and be reproduced

with the acceptable signal distortion.

The dynamic range of the quantizer input in the PCM system is 6n dB.

Nonuniform Quantizer

Used to reduce quantization error and increase the dynamic range when input signa

not uniformly distributed over its allowed range of values.

owed

lues input

for

t

pressing-and-expanding”

expanding” is called “companding.”

Nonuniform quantizer

ples Compressor Quantizer

Channel

••••

ved

al signals

Line codes:

1. Unipolar nonreturn-to-zero

zero (NRZ) Signaling

2. Polar nonreturn-to-zero(NRZ)

zero(NRZ) Signaling

3. Unipor nonreturn-to-zero

zero (RZ) Signaling

4. Bipolar nonreturn-to-zero

zero (BRZ) Signaling

5. Split-phase

phase (Manchester code)

Figure 3.15 Line codes for the electrical representations of binary data.

(a) Unipolar NRZ signaling. (b)) Polar NRZ signaling.

(c) Unipolar RZ signaling. (d)) Bipolar RZ signaling.

(e) Split-phase or Manchester code.

Application of Sampling Theorem –

PAM/TDM

Design of

PAM/TDM

System

UNIT-II

II DIGITAL MODULATION

Pulse Code Modulation (PCM)

2 PCM Sampling

Pulse Code Modulation

PCM Sampling

he Process

of Natural

Sampling

Quantization of Sampled Signal

s(t)

sq(t)

VH

s7 Δ7

L67

s6 Δ6

L56

s5 Δ5

s(t)

L45 sq(t)

peration of L34

s4 Δ4

uantization L23

s3 Δ3

s2 Δ2

Δ/2

L12

s1 Δ1

L01

Δ s0 Δ0

VL

t

Quantization Error and Classification

conversion of an analog difference between rounding off sample

values of an analog signal to the nearest

sample of the

permissible level of the quantizer during

information signal into the process of quantization.

discrete form. Thus, an

infinite number of

Classification of Quantization

possible levels are Process

converted to a finite

Uniform Non-uniform

number of conditions. quantization quantization

Characteristics of Compressor, Uniform

and Non-

Non-uniform Quantizer

μ-law and A-

A-law Compression

Characteristics

Encoding of Quantized Sampled Signal

PCM System Parameters

frequency component of information signal

DELTA MODULATION

Essence of Delta Modulation (DM)

Delta modulation (DM) uses a single-bit

single DPCM code to achieve

digital transmission of analog signals

DM system. (a)) Transmitter. (b)

( Receiver.

lator consists of a comparator, a quantizer, and an accumulator

utput of the accumulator is

n

m q n sgn( e i )

i 1

n

e q i (3.55)

i 1

ope overload distortion and granular noise

a-Sigma modulation (sigma-delta

delta modulation)

modulation

which has an integrator can

eve the draw back of delta modulation (differentiator

differentiator)

eficial effects of using integrator:

Pre-emphasize the low-frequency

frequency content

Increase correlation between adjacent samples

educe the variance of the error signal at the quantizer input )

Simplify receiver design

ause the transmitter has an integrator , the receiver

sists simply of a low-pass filter.

differentiator in the conventional DM receiver is cancelled by the integrator )

ivalent versions of delta-sigma modulation system.

Differential Pulse-Code

Code Modulation (DPCM)

PCM has the sampling rate higher than the Nyquist rate .The encode signal contains redundant information. D

ciently remove this redundancy.

Adaptive Differential Pulse--Code Modulation (ADPCM)

eed for coding speech at low bit rates , we have two aims in mind:

Remove redundancies from the speech signal as far as possible.

Assign the available bits in a perceptually efficient manner.

Comparison of PCM and DM Techniques

S. No. Parameter PCM DPCM DM ADM

1. Number of bits per 4/8/16 bits More than one bit but One bit One bit

sample less than PCM

2. Number of levels Depends on number of bits Fixed number of levels Two levels Two levels

4. Transmission bandwidth More bandwidth needed Lesser than PCM Lowest Lowest

6. Quantization Quantization noise depends Quantization noise & slope overload & Quantization noise only

noise/distortion on number of bits slope overload granular noise

implementation

UNIT-III

UNIT

Basband Pulse Transmission

Transmit and Receive Formatting

Sources of Error in received Signal

Major sources of errors:

– Thermal noise (AWGN)

• disturbs the signal in an additive fashion (Additive)

• has flat spectral density for all frequencies of interest (White)

• is modeled by Gaussian random process (Gaussian Noise)

– Inter-Symbol Interference (ISI)

• Due to the filtering effect of transmitter, channel and receiver, symbols are

“smeared”.

Receiver Structure

AW GN

DETECT

DEM ODULATE & SAM PLE

SAM PLE

at t = T

R ECEIV E D

W AVEFO RM FR EQ U EN C Y

R E C E IV IN G E Q U A L IZ IN G

DOWN

F IL T E R F IL T E R TH R ESH O LD M ESSA

A N S M ITT ED C O N V E R S IO N

AVEFORM C O M P A R IS O N SYM BO

OR

CHANN

FOR CO M P EN S A T IO N

SYM BO

BAN DPASS FO R CH ANN EL

SIG N A L S IN D U CED ISI

O P T IO N A L

E S S E N T IA L

Receiver Structure contd

The digital receiver performs two basic functions:

functions

– Demodulation

– Detection

Why demodulate a baseband signal???

– Channel and the transmitter’s filter causes ISI which “smears” the

transmitted pulses

– Required to recover a waveform to be sampled at t = nT.

Detection

– decision-making

making process of selecting possible digital symbol

Steps in designing the receiver

Find optimum solution for receiver design with the following goals:

1. Maximize SNR

2. Minimize ISI

Steps in design:

– Model the received signal

– Find separate solutions for each of the goals.

Detection of Binary Signal in Gaussian Noise

ilter

• Reduces the received signal to a single variable z(T)

• z(T) is called the test statistics

Detector (or decision circuit)

• Compares the z(T) to some threshold level 0 , i.e.,

H 1

z (T ) H1 and H0 are the two

where

possible binary hypothesis

0

H 0

Finding optimized filter for AWGN channel

function

Detection of Binary Signal in Gaussian Noise

• For any binary channel, the transmitted signal over a symbol interval (0,T)

( is:

s 0 (t ) 0 t T for a binary 0

si (t )

s1 ( t ) 0 t T for a binary 1

• The received signal r(t) degraded by noise n(t) and possibly degraded by the impulse resp

the channel hc(t), is

r (t ) s (t ) * h (t ) n (t ) i 1,2

Where n(t) is assumed toi be zero mean

c AWGN process

• For ideal distortionless channel where hc(t) is an impulse function and convolution with

produces no degradation, r(t) can be represented as:

r (t ) si (t ) n (t ) i 1, 2 0 t T

Design the receiver filter to maximize the SNR

t) h c (t ) r (t ) r (t ) si (t ) h c (t ) n (t )

n (t )

AWGN

– Ideal

Received

channels

signal in AWGN

hc (t ) (t )

si (t ) r (t ) r (t ) si (t ) n (t )

n (t )

AWGN

Find Filter Transfer Function H0(f)

sing signal ai(t) at filter output in terms of filter transfer function H(f)

j 2 ft

a i (t ) H ( f ) S( f ) e df

the filter transfer funtion and S(f) is the Fourier transform of input signal s(t)

ded PSD of i/p noise is N0/2

t noise power can be expressed as:

2 N 0

0 | H ( f ) | 2 df

2

(S/N)T : 2

j 2 fT

H ( f ) S( f ) e df

S

N T N 0

| H ( f ) | 2 df

2

• For H(f) = Hopt (f) to maximize (S/N)T use Schwarz’s Inequality:

Inequality

2 2 2

f 1 ( x ) f 2 ( x ) dx f 1 ( x ) dx f 2 ( x ) dx

• Equality holds if f1(x) = k f*2(x) where k is arbitrary constant and * indicates complex conjugate

• Associate H(f) with f1(x) and S(f) ej2 fT with f2(x) to get:

2 2 2

j 2 fT

H ( f ) S( f ) e df H ( f ) df S ( f ) df

• Substitute yields to:

2

S 2

S ( f ) df

N T N0

S 2 E

max and energy E of the input signal s(t):

N T N 0 2

E S ( f ) df

s (S/N)T depends on input signal energy

power spectral density of noise and

T on the particular shape of the waveform

holds 2E filter transfer function H0(f)

max

N T N 0

that: H ( f ) H 0 ( f ) kS * ( f ) e j 2 fT

h (t ) 1

kS * ( f(3.55)

)e j 2 fT

eal valued s(t):

kS ( T t ) 0 t T

h (t )

0 else where

mpulse response of a filter producing maximum output signal-to-noise

signal ratio is the

or image of message signal s(t), delayed by symbol time duration T.

filter designed is called a MATCHED FILTER

kS ( T t ) 0 t T

h (t )

0 else where

ned as:

a linear filter designed to provide the maximum

signal-to-noise

noise power ratio at its output for a given

transmitted symbol waveform

Matched Filter Output of a rectangular Pulse

Replacing Matched filter with Integrator

Implementation of matched filter receiver

Bank of M matched filters

*

z 1 (T )

s1 (T t ) z1 Matched filter output:

r (t ) z Observation

z vector

* z M

s M (T t ) zM (T

T)

z i r ( t ) s i (T t ) i 1 ,..., M

z ( z 1 ( T ), z 2 ( T ),..., z M ( T )) ( z 1 , z 2 ,..., z M )

Detection

Probability of Error

Detection

ed filter reduces the received signal to a single variable z(T), after which the detection of symbol is carried out

ncept of maximum likelihood detector is based on Statistical Decision Theory

ws us to

mulate the decision rule that operates on the data

imize the detection criterion

H 1

z (T )

0

H 0

Probabilities Review

hese probabilities are known before transmission

), p(z|s1)

nditional pdf of received signal z, conditioned on the class si

], P[s1|z] a posteriori probabilities

fter examining the sample, we make a refinement of our previous knowledge

0], P[s0|s1]

rong decision (error)

1], P[s0|s0]

rrect decision

How to Choose the threshold?

( criterion:

p ( s 0 | z ) p ( s1 | z ) H 0

p ( s1 | z ) p ( s 0 | z ) H 1

em is that a posteriori probabilities are not known.

on: Use Bay’s theorem:

p (z |s ) p (s )

p (s | z) i i

i p(z)

H H

p ( z | s1 ) P ( s1 )

1

p (z | s0 )P (s0 )

1

p ( z | s1 ) P ( s1 )

p ( z | s0 ) P ( s0 )

P (z) H0

P (z) H0

s means that if received signal is positive, s1 (t) was sent,

sent else

s0 (t) was sent

Likelihood of So and S1

1

MAP criterion:

H1

p ( z | s1 ) P (s0 )

L(z)

likelihood ratio test ( LRT )

p(z |s0 ) H

P ( s1 )

0

When the two signals, s0(t) and s1(t),, are equally likely, i.e., P(s0) = P(s1) = 0.5, then the decision rule

becomes

H 1

p ( z | s1 )

L(z)

1 max likelihood ratio test

p(z |s0) H 0

hypothesis that corresponds to the signal with the maximum likelihood.

erms of the Bayes criterion, it implies that the cost of both types of error is the same

tuting the pdfs

2

1 1 z a0

H 0 : p ( z | s0 ) exp

0 2 2 0

2

1 1 z a1

H1 : p ( z | s1 ) exp

0 2 2 0

H1 1 1 H1

exp 2

z a 1 2

p ( z | s1 ) 0 2 2 o

L(z) 1 1

p ( z | s0 ) 1 1 2

exp 2

z a 0

H 0 0 2 2 0 H 0

Hence:

z (a1 a 0 ) ( a 12 a 02 )

exp 2

1

0 2 02

Taking the log, both sides will give

H1

z ( a1 a 0 ) ( a 12 a 02 )

ln{ L ( z )} 2

0

0 2 02

H 0

H 1

z (a1 a 0 ) ( a 12 a 02 ) ( a 1 a 0 )( a 1 a 0 )

02 2 02 2 02

H 0

• Hence

H1 H1

02 ( a 1 a 0 )( a 1 a 0 )

z ( a1 a 0 )

2 02 ( a 1 a 0 ) z 0

2

H0 H0

where z is the minimum error criterion and 0 is optimum

ptimum threshold

• For antipodal signal, s1(t) = - s0 (t) a1 = - a0

H1

z 0

H0

Probability of Error

will occur if

s sent s0 is received

P(H 0 | s1 ) P ( e | s1 )

0

P ( e | s1 ) p ( z | s 1 ) dz

s sent s1 is received

P (H 1 | s0 ) P (e | s0 )

P (e | s0 ) p ( z | s 0 ) dz

0

2

PB

i 1

P ( e , s i ) P ( e | s1 ) P ( s1 ) P ( e | s 0 ) P ( s 0 )

P (H 0 | s1 ) P ( s1 ) P ( H 1 | s0 ) P (s0 )

nals are equally probable

PB P ( H 0 | s1 ) P ( s1 ) P ( H 1 | s0 ) P (s0 )

1

P ( H 0 | s1 ) P ( H 1 | s0 )

2

e, the probability of bit error PB, is the probability that an incorrect hypothesis is made

rically, PB is the area under the tail of either of the conditional distributions p(z|s1) or p(z|s0)

PB P (H 1 | s 0 ) dz p ( z | s 0 ) dz

0 0

2

1 1 z a

0

exp dz

0 0 2 2 0

Inter-Symbol

Symbol Interference (ISI)

SI in the detection process due to the filtering effects of the

ystem

Overall equivalent system transfer function

H ( f ) H t ( f )H c ( f )H r ( f )

– creates echoes and hence time dispersion

– causes ISI at sampling time

zk sk nk

i k

i si

Inter-symbol

symbol interference

Baseband system model

x2

Channel Rx. filter zk

Tx filter r (t )

ht (t ) hc (t ) hr (t ) Detector

t kT

T H t( f ) Hc( f ) Hr( f )

x3 T n (t )

Equivalent model

x2

Equivalent system z (t ) zk

h (t ) Detector

t kT

H (f )

x3 T nˆ ( t )

filtered noise

H ( f ) H t ( f )H c ( f )H r ( f )

Nyquist bandwidth constraint

Nyquist bandwidth constraint:

• The theoretical minimum required system bandwidth to detect Rs [symbols/s]

without ISI is Rs/2 [Hz].

• Equivalently, a system with bandwidth W=1/2T=Rs/2 [Hz] can support a maxim

transmission rate of 2W=1/T=Rs [symbols/s] without ISI.

1 Rs Rs

W 2 [symbol/s/ Hz]

2T 2 W

Bandwidth efficiency, R/W [bits/s/Hz] :

• An important measure in DCs representing data throughput per hertz of bandw

• Showing how efficiently the bandwidth resources are used by signaling techniq

Ideal Nyquist pulse (filter)

Ideal Nyquist filter Ideal Nyquist pulse

H (f ) h ( t ) sinc( t / T )

T 1

0 f 2T T 0 T 2T

1 1

2T 2T

1

W

2T

Nyquist pulses (filters)

Nyquist pulses (filters):

– Pulses (filters) which results in no ISI at the sampling time.

Nyquist filter:

– Its transfer function in frequency domain is obtained by convolving a

rectangular function with any real even-symmetric

even frequency function

Nyquist pulse:

– Its shape can be represented by a sinc(t/T) function multiply by another

time function.

Example of Nyquist filters: Raised-Cosine

Raised filter

Pulse shaping to reduce ISI

Goals and trade-off in pulse-shaping

shaping

– Reduce ISI

– Efficient bandwidth utilization

– Robustness to timing error (small side lobes)

The raised cosine filter

aised-Cosine Filter

– A Nyquist pulse (No ISI at the sampling time)

1 for | f | 2 W 0 W

| f | W 2W 0

2

H ( f ) cos for 2 W 0 W | f | W

4 W W 0

0 for | f | W

cos[ 2 (W W 0 ) t ]

h ( t ) 2 W 0 (sinc( 2 W 0 t ))

1 [ 4 (W W 0 ) t ] 2

W W0

Excess bandwidth: W W0 Roll-off factor r

W0

0 r 1

The Raised cosine filter – cont’d

| H ( f ) | | H RC (f )| h ( t ) h RC ( t )

1 r 0 1

r 0 .5

0.5 0.5 r 1

r 1 r 0 .5

r

1 3 1 0 1 3 1 3T 2T T 0 T 2T

T 4T 2T 2T 4T T

Rs

Baseband W sSB (1 r ) Passband W DSB (1 r ) R s

2

Pulse shaping and equalization to remove ISI

No ISI at the sampling time

H RC ( f ) H t ( f )H c ( f )H r ( f )H e ( f )

quare-Root

Root Raised Cosine (SRRC) filter and Equalizer

H RC ( f ) H t ( f )H r ( f )

Taking care of ISI

H r( f ) Ht( f ) H (f) H (f) caused by tr. filter

RC SRRC

1

H e( f ) Taking care of ISI

H c( f ) caused by channel

Example of pulse shaping

Square-root Raised-Cosine

Cosine (SRRC) pulse shaping

mp. [V]

Third pulse

t/T

First pulse

Second pulse

Data symbol

Example of pulse shaping …

Raised Cosine pulse at the output of matched filter

Amp. [V]

the matched filter output

(zero ISI)

t/T

Eye pattern

Eye pattern:Display on an oscilloscope which sweeps the system response to

a baseband signal at the rate 1/T (T symbol duration)

istortion

due to ISI

Noise margin

amplitude scale

Sensitivity t

timing erro

Timing jitter

time scale

Example of eye pattern:

Binary-PAM,

PAM, SRRQ pulse

Perfect channel (no noise and no ISI)

Correlative Coding

Transmit 2W

W symbols/s with zero ISI, using the theoretical minimum bandwidth of W Hz, without

infinitely sharp filters.

Correlative coding (or duobinary signaling or partial response signaling) introduces some

controlled amount of ISI into the data stream rather than trying to eliminate ISI completely

Doubinary signaling

Duobinary signaling

Duobinary signaling (class I partial response)

Duobinary signal and Nyguist Criteria

Nyguist second criteria: but twice the bandwidth

Differential Coding

interval.

The response is partial in any signaling interval.

Detection :

– Major drawback : error propagation.

To avoid error propagation, need deferential coding

(precoding).

Modified duobinary signaling

Modified duobinary signaling

– In duobinary signaling, H(f) is nonzero at the origin.

– We can correct this deficiency by using the class IV partial response

Modified duobinary signaling

Spectrum

Modified duobinary signaling

Time Sequency: interpretation of receiving 2, 0, and -2?

Duobinary Transfer Function

Comparison of Binary with Duobinary

Signaling

Binary signaling assumes the transmitted pulse amplitude are independent of one another

Duobinary signaling introduces correlation between pulse amplitudes

Duobinary technique achieve zero ISI signal transmission using a smaller system bandwidth

Duobinary coding requires three levels, compared with the usual two levels for binary coding

Duobinary signaling requires more power than binary signaling (~2.5

(~ dB greater SNR than binary

signaling)

Pass-band

band Data Transmission

Block Diagram

band data transmission system.

Signaling

Illustrative waveforms for the three basic forms of signaling binary information.

(a) Amplitude-shift keying. (b) Phase-shift

shift keying. (c)

( Frequency-shift keying with

continuous phase.

What do we want to study?

techniques in terms of

– Probability of errors

– Power Spectrum

– Bandwidth efficiency

Rb

Bits/s/Hz

B

Coherent PSK

– Consider the system with 2 basis functions

2

1 t cos 2 f c t

Tb

– and

2

2 t sin 2 f c t

Tb

BPSK

If we want to fix that for both symbols (0

( and 1) the transmitted energies are

equal, we have

2

s0

s1

1

s0

We place s0 to minimize

probability of error

BPSK

We found that phase of s1 and s0 are 180 degree difference.

We can rotate s1 and s0

2

s1

1

s0

Rotate

BPSK

2

s1

s0

1

basis function is sufficient to represent the signals

BPSK

Finally, we have

2Eb

s 1 t E b 1 (t ) cos 2 f c t

Tb

2Eb

s 0 t E b 1 (t ) cos 2 f c t

Tb

BPSK

Signal-space

space diagram for coherent binary PSK system. The waveforms depicting

the transmitted signals s1(t) and s2(t),

), displayed in the inserts, assume nc 2.

BPSK

(Pr(m0)=Pr(m1)), we have

1 d ik

Pe erfc

2 2 N

0

1 Eb

erfc

2 N

0

BPSK

Block diagrams for (a)) binary PSK transmitter and (b)

( coherent binary PSK

receiver.

Quadriphase-Shift

Shift Keying (QPSK)

2E

s i t cos 2 f c t 2 i 1 ; 0 t T

T 4

T is symbol duration

E is signal energy per symbol

There are 4 symbols for i = 1, 2, 3,, and 4

QPSK

2 2

s i t E cos 2 i 1 cos 2 f c t E sin 2 i 1 sin 2 f c t

4 T 4 T

E cos 2 i 1 1 t E sin 2 i 1 2 t ; 0 t T

4 4

E cos 2 i 1

si 4

E sin 2 i 1

4

QPSK

QPSK point

signaling si1 si2

1 10 /4

E /2 E /2

2 00 3 / 4 E /2 E /2

3 01 5 / 4 E /2 E /2

4 11 7 / 4

E /2 E /2

QPSK

2

s3 s4

(01) (11)

1

s2 s1

(00) (10

10)

QPSK signals

QPSK

Block diagrams of (a)

QPSK transmitter and

(b) coherent QPSK

receiver.

QPSK: Error Probability QPSK

Consider signal Z3 2 Z4

constellation given in

s3 s4

the figure (10

10) (11)

E /2

E /2 E /2

Z1 1

s2 s1

Z2

(10)

(00) E /2

QPSK

can treat QPSK as the combination of 2 independent

K over the interval T=2Tb

ce the first bit is transmitted by 1 and the second bit is

nsmitted by 2.

bability of error for each channel is given by

1 d 12 1 E

P erfc erf c

2 2 2N 0

2 N0

QPSK

mbol is to be received correctly both bits must be received

ectly.

ce, the average probability of correct decision is given by

2

P

ch gives the probability of errors equal

c 1

to P

E 1 2 E

PC erfc erfc

2N 0 4 2 N 0

E

fc

2 N 0

QPSK

e one symbol of QPSK consists of two bits, we have E = 2Eb.

Eb

Pe per symbol erfc

N0

above probability is the error probability per symbol. The avg.

bability of error per bit

1 1 Eb

it Pe per symbol erfc

N

2 2

ch is exactly the same as BPSK

.0

BPSK vs QPSK

P o w e r s p e c t ru m d e n s it y o f B P S K vs . Q P S K

2

1.8 BPSK

QPSK

1.6

1.4

b

Normalized PSD,Sf/2E

1.2

0.8

0.6

0.4

0.2

0

0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2

N o rm a liz e d fre q u e n c y , fT b

QPSK

Conclusion

– QPSK is capable of transmitting data twice as faster as BPSK with the

same energy per bit.

– We will also learn in the future that QPSK has half of the bandwidth

of BPSK.

OFFSET QPSK

90 degree shift in phase

2

(01) s3 s4 (11)

1

s2 s1

(00) (10

10)

OFFSET QPSK

OFFSET QPSK

Whenever both bits are changed simultaneously, 180 degree

phase-shift occurs.

At 180 phase-shift,

shift, the amplitude of the transmitted signal

changes very rapidly costing amplitude fluctuation.

This signal may be distorted when is passed through the filter

or nonlinear amplifier.

OFFSET QPSK

2

-1

-2

0 1 2 3 4 5 6 7 8

Original Signal

2

1 . 5

0 . 5

- 0 . 5

- 1

- 1 . 5

- 2

0 1 2 3 4 5 6 7 8

Filtered signal

OFFSET QPSK

To solve the amplitude fluctuation problem, we propose the

offset QPSK.

Offset QPSK delay the data in quadrature component by T/2

T/

seconds (half of symbol).

Now, no way that both bits can change at the same time.

OFFSET QPSK

In the offset QPSK, the phase of the signal can change by 90

or 0 degree only while in the QPSK the phase of the signal can

change by 180 90 or 0 degree.

OFFSET QPSK

hase 1

SK 0 . 5

0 1 0

- 0

0

. 5

1

- 1

0 1 2 3 4 5 6 7 8

phase 0 . 5

0

1 0 0 0

PSK - 0 . 5

- 1

0 1 2 3 4 5 6 7 8

PSK 10 00

0

- 1

- 2

0

01

1 2 3 4

10

5 6 7 8

0 . 5

0 0 1 1 0

SK

- 0 . 5

- 1

0 1 2 3 4 5 6 7 8

1 0

0 . 5

SK

0

- 0 . 5

0

- 1

0 1 2 3 4 5 6 7 8

SK

1

10 10

0

- 1

01 00

- 2

0 1 2 3 4 5 6 7 8

Offset QPSK

message points in (a) QPSK and (b)) offset QPSK.

OFFSET QPSK

dwidths of the offset QPSK and the regular QPSK is the

me.

m signal constellation we have that

E

P e erfc

2 N 0

M-array

array PSK

At a moment, there are M possible symbol values being sent

for M different phase values,

i 2 i 1 / M

2E 2

s i t cos 2 f c t i

1 , i 1, 2 , , M

T M

M-array

array PSK

Signal-space diagram for octaphase-

shift keying (i.e., M 8). The decision

boundaries are shown as dashed

lines.

Signal-space diagram illustrating the

application of the union bound for

octaphase-shift keying.

M-array

array PSK

Probability of errors

d 12 d 18 2 E sin / M

E

P e erfc sin / M ; M 4

N0

M-ary

ary PSK

0

10

-1 0

10

Probability of Symbol errors

-2 0

10

-3 0

10

-4 0

10

QPSK

8 -a ry P S K

1 6 -a ry P S K

-5 0

10

0 5 10 15 20 25 30

E /N dB

b 0

M-array

array PSK

Power Spectra (M-array)

S PSK ( f ) 2 E sinc 2 Tf

2 E b log 2 M sinc 2 T b f log 2 M

M=2, we have

S BPSK ( f ) 2 E b sinc 2 T b f

M-array

array PSK

Power spectra of M-ary

ary PSK signals for M 2, 4, 8.

Tbf

M-array

array PSK

Bandwidth efficiency:

– We only consider the bandwidth of the main lobe (or null-to-null

null bandwidth)

2 2 2 Rb

B

T T b log 2 M log 2 M

– Bandwidth efficiency of M-ary

ary PSK is given by

Rb Rb

log 2 M 0 . 5 log 2 M

B 2Rb

M-ary

ary QAM

QAM = Quadrature Amplitude Modulation

Both Amplitude and phase of carrier change according to the

transmitted symbol, mi.

where a

s i t i and

2 E 0b are integers.

ia i cos 2 f c t

2E0

b i sin 2 f c t ; 0 t T

T T

M-ary

ary QAM

Again, we have

2

1 t cos 2 f c t ;0 t T

T

2

2 t sin 2 f c t 0 t T

Tb

M-ary

ary QAM

QAM square Constellation

– Having even number of bits per symbol, denoted by 2n.

– M=L x L possible values

– Denoting

L M

16-QAM

QAM

( 3 ,3 ) ( 1, 3 ) (1 , 3 ) ( 3 ,3 )

( 3 ,1 ) ( 1 ,1 ) (1 ,1 ) ( 3 ,1 )

a i , b i

( 3 , 1) ( 1, 1 ) (1 , 1 ) ( 3 , 1)

( 3, 3) ( 1, 3 ) (1 , 3 ) (3, 3)

L-ary, 4-PAM

16-QAM

16-QAM

QAM

Calculation of Probability of errors

– Since both basis functions are orthogonal, we can treat the 16-QAM

as combination of two 4-ary

ary PAM systems.

– For each system, the probability of error is given by

E0

Pe 1

1

erfc d 1 1

erfc

L 2 N M N0

0

16-QAM

QAM

– A symbol will be received correctly if data transmitted on both 4-ary

PAM systems are received correctly. Hence, we have

Pc symbol 1 P e 2

– Probability of symbol error is given by

1 1 2 Pe Pe 2 2 Pe

16-QAM

QAM

– Hence, we have

1 E0

Pe symbol 21 erfc

M N0

– But because average energy is given by

2E0 L / 2 2 2 M 1 E 0

E av 2 2 i 1

L i 1 3

– We have

1 3 E av

Pe symbol 21 erfc

M 2 M 1 N 0

Coherent FSK

FSK = frequency shift keying

Coherent = receiver have information on where the zero phas

of carrier.

We can treat it as non-linear

linear modulation since information is

put into the frequency.

Binary FSK

Transmitted signals are

2Eb

cos 2 f i t , 0 t Tb

s i t Tb

0, elsewhere

where

nc i

fi ; i 1, 2

Tb

Binary FSK

S1(t) represented symbol “1”.

S2(t) represented symbol “0”.

This FSK is also known as Sunde’s FSK.

It is continuous phase frequency-shift

frequency keying (CPFSK).

Binary FSK

There are two basis functions written as

2

cos 2 f i t , 0 t Tb

i t T b

0, elsewhere

Eb 0

s1 and s2

0 E b

BFSK

From the figure, we have d 12 2 E b

In case of Pr(0)=Pr(1),

), the probability of error is given by

1 Eb

Pe erfc

2 2N

We observe that at a given value

of0 P

e, the BFSK system

requires twice as much power as the BPSK system.

TRANSMITTER

RECEIVER

Power Spectral density of BFSK

Consider the Sunde’s FSK where f1 and f2 are different by 1/Tb. We can write

2Eb t

s i t cos 2 f c t

Tb Tb

2Eb t 2Eb t

cos cos 2 f c t sin sin 2 f c t

Tb Tb Tb Tb

We observe that in-phase

phase component does not depend on mi since

2Eb t 2Eb t

cos cos

Tb T b Tb T

b

Power Spectral density of BFSK

Half of the symbol power

We have

2

2Eb t Eb 1 1

S BI f F cos f f

Tb Tb 2Tb 2Tb 2Tb

2Eb t 8 E b T b cos 2 T b f

g t sin S BQ

Tb Tb

2 4 T b2 f 2 1

2

Power Spectral density of BFSK

Finally, we obtain S B ( f ) S BI ( f ) S BQ ( f )

Phase Tree of BFSK

FSK signal is given by

2Eb t

s t cos 2 f c t

Tb Tb

At t = 0, we have

2Eb 0 2Eb

s 0 cos 2 f c 0 cos 0

Tb T b Tb

Phase Tree of BFSK

At t = Tb, we have

2Eb Tb 2Eb

s T b cos 2 f c T b cos

Tb Tb Tb

for symbol “1” and + for symbol “0

0”

Minimum-Shift

Shift keying (MSK)

MSK tries to shift the phase after one symbol to just half of

Sunde’s FSK system. The transmitted signal is given by

2Eb

cos 2 f 1 t 0 for "1"

2Eb Tb

cos 2 f c t t

Tb 2Eb

cos 2 f 2 t 0 for "0"

Tb

MSK

Where

h

t 0 t

Tb

Observe that

h h

f1 f c and f2 fc

2T b 2T b

1

fc f1 f2

2

MSK

h = Tb(f1-f2) is called “deviation ratio.”

For Sunde’s FSK, h = 1.

For MSK, h = 0.5.

h cannot be any smaller because the orthogonality between cos(2f

cos( 1t)

and cos(2f2t) is still held for h < 0.5

5.

Orthogonality guarantees that both signal will not interfere each other

in detection process.

MSK

Phase trellis diagram for MSK signal 1101000

MSK

Signal s(t)

(t) of MSK can be decomposed into

2Eb

s t cos 2 f c t t

Tb

2Eb 2Eb

cos t cos 2 f c t sin t sin 2 f c t

Tb Tb

s I t cos 2 f c t s Q t sin 2 f c t

where

t 0 t ;0 t T b

2Tb

MSK

0 /2

1

-/2

0 -/2

0

/2

MSK

For the interval –Tb < t 0,, we have

t 0 t ; T b t 0

2T b

Let’s note here that the for the interval -Tb<t 0 and 0< tTb ma

not be the same.

We know that

t t t

cos 0 cos 0 cos sin 0 sin

2T b 2Tb 2Tb

MSK

Since (0) can be either 0 or depending on the past history. We have

t t t

cos 0 cos 0 cos cos

2Tb 2Tb 2T b

“+” for (0) = 0 and “-” for (0) =

Hence, we have

2Eb t

s I (t ) cos ;Tb t Tb

Tb 2T b

MSK

Similarly we can write

t T b t T b

2T b

for 0< tTb and Tb < t2Tb. Note the “+” and “-”

“ may be different

between these intervals.

Furthermore, we have that (Tb) can be /2 depending on the

past history.

MSK

Hence, we have

t T b t T b t T b

sin T b sin T b cos cos T b sin

2 T b 2T b 2T b

t t

sin T b cos cos T b sin

2Tb 2 2Tb 2

t T b t t

sin T b cos

sin

2Tb 2T b 2 2T b

MSK

Hence, we have

2Eb t

s Q (t ) sin ;0 t 2T b

Tb 2T b

“+” for (Tb) = +/2 and “-” for (Tb) = -/2

The basis functions change to

2 t

1 t cos cos 2 f c t ;0 t T b

Tb 2Tb

2 t

2 t sin sin 2 f c t ;0 t T b

Tb 2T b

MSK

We write MSK signal as

2Eb 2Eb

s t cos t cos 2 f c t sin t sin 2 f c t

Tb Tb

2Eb 2 t 2Eb 2 t

cos 0 cos cos 2 f c t sin T b sin sin 2 f c t

Tb Tb Tb Tb

E b cos 0 1 ( t ) E b sin T b 2 ( t )

s 1 1 ( t ) s 2 2 ( t )

Where s1 and

E b cos 0 s2 E b sin T b

MSK

0 Eb /2 Eb

1

Eb -/2 Eb

0 Eb -/2 Eb

0

Eb /2 Eb

1 Eb

Pe erfc

2 N0

Phase: 0 /2 /2 /2 0 -/2

MSK

We observe that MSK is in fact the QPSK having the

t

pulse shape cos

2T b

next two slides.

Tb

x1 x ( t ) 1 ( t ) dt

Tb

2T b

x2 x ( t ) 2 ( t ) dt

0

4

MSK

3.5 BPSK

QPSK

2.5

1.5

0.5

0

0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2

N o rm a liz e d F re q u e n c y , fT b

MSK

Probability of error of MSK system is equal to BPSK and QPSK

This due to the fact that MSK observes the signal for two symbol

intervals whereas FSK only observes for single signal interval.

Bandwidth of MSK system is 50% % larger than QPSK.

2

32 E b cos 2 T b f

S MSK ( f )

2 16 T b2 f 2 1

Noncoherent Orthogonal Modulation

Noncoherent implies that phase information is not available to the

receiver.

As a result, zero phase of the receiver can mean any phase of the

transmitter.

Any modulation techniques that transmits information through the

phase cannot be used in noncoherent receivers.

Noncoherent Orthogonal Modulation

sin(2ft) sin(2ft)

cos(2ft)

cos(2ft)

Receiver

Transmitter

Noncoherent Orthogonal Modulation

where the axes are.

However, we can still determine the distance of the each signal

constellation from the origin.

As a result, the modulation techniques that put information in the

amplitude can be detected.

FSK uses the amplitude of signals in two different frequencies. Hence

non-coherent

coherent receivers can be employed.

Noncoherent Orthogonal Modulation

Consider the BFSK system where two frequencies f1 and f2 are used to

represented two “1” and “0”.

The transmitted signal is given by

2E

s (t ) cos 2 f i t ; i 1, 2 , 0 t T b

T

Problem is that is unknown to the receiver. For the coherent receiver

is precisely known by receiver.

Noncoherent Orthogonal Modulation

Furthermore, we have

2E

s (t ) cos 2 f i t

T

2E 2E

cos cos 2 f i t sin sin 2 f i t

T T

s i1 1 ( t ) s i 2 2 ( t )

( we use the amplitude

Noncoherent Orthogonal Modulation

Where

T T

s i 1 s ( t ) 1 ( t ) dt x 1 x ( t ) 1 ( t ) dt

0 0

T T

s i 2 s ( t ) 2 ( t ) dt x 2 x ( t ) 2 ( t ) dt

0 0

The amplitude of the received signal

1/2

T 2 2

T

l i x ( t ) cos 2 f i t dt x ( t ) sin 2 f i t dt

0 0

Noncoherent Orthogonal Modulation

Probability of Errors

1 E

Pe exp

2 2N 0

Noncoherent: BFSK

For BFSK, we have

2Eb

cos 2 f i t ; 0 t Tb

s i t Tb

0 ; elsewhere

Noncoherent: BFSK

Noncoherent: BFSK

Probability of Errors

1 Eb

Pe exp

2 2N 0

DPSK

Differential PSK

– Instead of finding the phase of the signal on the interval 0<tTb. Thi

receiver determines the phase difference between adjacent time

intervals.

– If “1”” is sent, the phase will remain the same

– If “0”” is sent, the phase will change 180 degree.

DPSK

Or we have

Eb

cos 2 f c t ; 0 t 2Tb

2Tb

s1 ( t )

Eb

cos 2 f c t ; Tb t 2T b

2Tb

and

Eb

cos 2 f c t ; 0 t 2Tb

2T b

s 2 (t )

Eb

cos 2 f c t ; T b t 2T b

2T b

DPSK

In this case, we have T=2Tb and E=2

2Eb

Hence, the probability of error is given by

1 Eb

Pe exp

2 N0

DPSK: Transmitter

d k bk d k 1 bk d k 1

DPSK

b k} 1 0 0 1 0 0 0 1 1

dk-1} 1 1 0 1 1 0 1 0 0

Differential encoded

1 1 0 1 1 0 1 0 0 0

d k}

Transmitted Phase 0 0 0 0 0

DPSK: Receiver

DPSK: Receiver

From the block diagram, we have that the decision rule as

say 1

l x x I 0 x I 1 x Q 0 x Q 1 0

say 0

If the phase of signal is unchanged (send “ the sign (“+” or “-”) of

“1”)

both xi and xQ should not change. Hence, the l(x) should be positive.

If the phase of signal is unchanged (send “0”)

“ the sign (“+” or “-1”) of

both xi and xQ should change. Hence, the l(x) should be negative.

Signal-space

space diagram of received DPSK signal.

Unit-V –Introduction

Introduction to Spread Spectrum Techniques

M-ary signaling scheme:

In this signaling scheme 2 or more bits are grouped

together to form a symbol.

s1(t) ,s2(t),s3(t),……sM(t)

is transmitted during each symbol period

of duration Ts.

where n is an integer.

The symbol values of M for a given value of n:

n M = 2n Symbol

1 2 0, 1

…. …… ……….

• Depending on the variation of amplitude, phase or frequency of the carrier, the modulation scheme is calle

M-ary ASK, M-ary PSK and M-ary FSK.

M-ary

ary Phase Shift Keying(MPSK)

In M-ary

ary PSK, the carrier phase takes on one of the M possible values, namely

i = 2 * (i - 1) / M

where i = 1, 2, 3, …..M.

The modulated waveform can be expressed as

Ts is symbol period = (log2 M) Tb.

he above equation in the Quadrature form is

M-ary signal set can be expressed as

Since there are only two basis signals, the constellation of M-ary

M PSK is two

dimensional.

The M-ary

ary message points are equally spaced on a circle of radius Es, centered

at the origin.

ary PSK signal set is shown in fig.

Fig: Constellation diagram of an M-ary

ary PSK system(m=8)

Derivation of symbol error probability:

Decision Rule:

If a symbol (0,0,0)) is transmitted, it is clear

that if an error occurs, the transmitted signal is most

likely to be mistaken for (0,0,1) and (1,1,11) and the

signal being mistaken for (1,1,0)) is remote.

/8(below 1(t)- axis) to = + /8 ( above 2(t)- axis)

Fig: Probability density function of Phase .

The average symbol error probability of an coherent M-ary

M PSK system in

AWGN channel is given by

M PSK system in

AWGN channel is given by

Fig: The performance of symbol error probability for

-different values of M

M-ary

ary Quadrature Amplitude Modulation

(QAM)

As we allow the amplitude to also vary with the phase, a new modulation scheme

called quadrature amplitude modulation (QAM) is obtained.

ary QAM consists of a square lattice of signal

points.

Fig: signal Constellation of M-ary

ary QAM for M=16

M=

M-ary

ary Frequency Shift

Keying(MFSK)

In M-ary

ary FSK modulation the transmitted signals are defined by:

The M transmitted signals are of equal energy and equal duration, and

the signal frequencies are separated by 1/2Ts Hertz, making the signal

orthogonal to one another.

The average probability of error based on the union bound is given by

Power Efficiency and Bandwidth :

Bandwidth:

ary FSK signal is :

The channel bandwidth of a noncohorent MFSK is :

M FSK signal decreases

with increasing M. Therefore, unlike M--PSK signals, M-FSK signals are

bandwidth inefficient.

Introduction to Spread Spectrum

occur with wireless systems

• Spread spectrum modulation spreads out the modulated signal

bandwidth so it is much greater than the message bandwidth

• Independent code spreads signal at transmitter and despreads sig

at receiver

Multiplexing

channels ki

• Multiplexing in 4 dimensions

– space (si) k1 k2 k3 k4 k5 k6

– time (t) c

– frequency (f) t c

– code (c) t

s1

f

• Goal: multiple use s2

of a shared medium c

t

• Important: guard spaces needed!

s3

f

Frequency multiplex

• Separation of spectrum into smaller frequency bands

• Channel gets band of the spectrum for the whole time

• Advantages:

k3 k4 k5

– no dynamic coordination needed

– works also for analog signals c

• Disadvantages:

– waste of bandwidth

if traffic distributed unevenly

– inflexible

– guard spaces

t

Time multiplex

Channel gets the whole spectrum for a certain amount of time

Advantages:

– only one carrier in the

medium at any time

– throughput high even

for many users k1 k2 k3 k4 k5

c

Disadvantages:

– precise

synchronization

necessary

t

Time and frequency multiplex

• A channel gets a certain frequency band for a certain amount of time (e.g.

GSM)

• Advantages:

– better protection against tapping

– protection against frequency

selective interference

– higher data rates compared to k1 k2 k3 k4 k5

code multiplex c

• Precise coordination

required

t

Code multiplex

k1 k2 k3 k4 k5 k6

c

All channels use same spectrum at same time

Advantages:

– bandwidth efficient

– no coordination and synchronization

– good protection against interference

Disadvantages:

– lower user data rates

– more complex signal regeneration

Implemented using spread spectrum technology

t

Spread Spectrum Technology

• Problem of radio transmission: frequency dependent fading can wipe out

narrow band signals for duration of the interference

• Solution: spread the narrow band signal into a broad band signal using a

special code

interference

spread signal power signal

ower spread

interferenc

detection at

receiver

f f

Spread Spectrum Technology

• Side effects:

– coexistence of several signals without dynamic coordination

– tap-proof

• Alternatives: Direct Sequence (DS/SS), Frequency Hopping (FH/SS)

• Spread spectrum increases BW of message signal by a factor N, Processing

Gain

B ss B ss

P r o c e s s in g G a in N 1 0 lo g 1 0

B B

Effects of spreading and interference

user signal

broadband interference

narrowband interference

P P

i) ii)

f f

P sender P P

iii) iv) v)

f f f

receiver

Spreading and frequency selective fading

channel

quality

2 narrowband channels

1 5 6

3

4

guard space

channel

quality

2

2

2

2

2

1 spread spectrum

channels

spread frequency

spectrum

DSSS (Direct Sequence Spread Spectrum) I

• Advantages

– reduces frequency selective

fading Tb

– in cellular networks user dat

• base stations can use the

same frequency range

0 1 XOR

• several base stations can

Tc

detect and recover the signal chippin

• But, needs precise power control sequen

01101010110101

=

resulting

signal

01101011001010

DSSS (Direct Sequence Spread Spectrum) II

transmitter

Spread spectrum

Signal y(t)=m(t)c(t) transmit

user data signal

X modulator

m(t)

chipping radio

sequence, c(t) carrier

receiver correlator

sampled

received products data

sums

signal demodulator X integrator decision

radio

carrier

Chipping sequence, c(t)

DS/SS Comments III

Pseudonoise(PN) sequence chosen so that its autocorrelation

is very narrow => PSD is very wide

– Concentrated around t < Tc

– Cross-correlation

correlation between two user’s codes is very small

DS/SS Comments IV

Secure and Jamming Resistant

– Both receiver and transmitter must know c(t)

– Since PSD is low, hard to tell if signal present

– Since wide response, tough to jam everything

Multiple access

– If ci(t) is orthogonal to cj(t), then users do not interfere

Near/Far problem

– Users must be received with the same power

FH/SS (Frequency Hopping Spread Spectrum)

• Discrete changes of carrier frequency

– sequence of frequency changes determined via PN sequence

• Two versions

– Fast Hopping:: several frequencies per user bit (FFH)

– Slow Hopping:: several user bits per frequency (SFH)

• Advantages

– frequency selective fading and interference limited to short period

– uses only small portion of spectrum at any time

• Disadvantages

– not as robust as DS/SS

– simpler to detect

FHSS (Frequency Hopping Spread Spectrum) I

Tb

user data

0 1 0 1 1 t

f

Td

f3 slow

f2 hopping

(3 bits/hop)

f1

Td t

f

f3 fast

f2 hopping

(3 hops/bit)

f1

t

Tb: bit period Td: dwell time

FHSS (Frequency Hopping Spread Spectrum) III

signal signal

user data

modulator modulator

frequency hopping

synthesizer sequence

receiver

received data

signal demodulator demodulator

hopping frequency

sequence synthesizer

Applications of Spread Spectrum

Cell phones

– IS-95 (DS/SS)

– GSM

Global Positioning System (GPS)

Wireless LANs

– 802.11b

Performance of DS/SS Systems

Pseudonoise (PN) codes

– Spread signal at the transmitter

– Despread signal at the receiver

Ideal PN sequences should be

– Orthogonal (no interference)

– Random (security)

– Autocorrelation similar to white noise (high at t=0 and low for t not

equal 0)

PN Sequence Generation

• Codes are periodic and generated by a shift register and XOR

• length (ML) shift register sequences, m-stage shift register, length: n = 2m – 1

Maximum-length

bits

R(

R(t)

t

-1/n Tc nTc

-nTc

Output

+

Generating PN Sequences

+ modulo-2 adder

Take m=2 =>L=3 2 1,2

cn=[1,1,0,1,1,0, . . .], usually written

as bipolar cn=[1,1,-1,1,1,-1, . . .] 3 1,3

4 1,4

5 1,4

1 L

m cncnm 6 1,6

L n 1

1 m 0 8 1,5,6,7

1 / L 1 m L 1

Problems with m-sequences

Cross-correlations with other m-sequences

m generated by

different input sequences can be quite high

Easy to guess connection setup in 2m samples so not too

secure

In practice, Gold codes or Kasami sequences which combine

the output of m-sequences

sequences are used.

Detecting DS/SS PSK Signals

transmitter

Spread spectrum

Signal y(t)=m(t)c(t) transmit

Bipolar, NRZ signal

m(t) X X

PN

sequence, c(t) sqrt(2)cos

)cos (wct + )

receiver

received z(t) w(t)

signal

X X LPF integrator decision

x(t)

Optimum Detection of DS/SS PSK

Recall, bipolar signaling (PSK) and white noise give the optimum error

probability

2 Eb

Pb Q

Not effected by spreading

– Wideband noise not affected by spreading

– Narrowband noise reduced by spreading

Signal Spectra

B ss B ss Tb

P r o c e s s in g G a in N 1 0 lo g 1 0

B B Tc

• Effective noise power is channel noise power plus jamming (NB)

signal power divided by N

Tb

Tc

Multiple Access Performance

Assume K users in the same frequency band,

Interested in user 1,, other users interfere

4 6

1

3 2

Signal Model

Interested in signal 1,, but we also get signals from other K-1

users:

xk t 2 mk t k c k t k c o s c t k k

2 mk t k c k t k c o s c t k k k

At receiver,

K

x t x1 t x k t

k 2

Interfering Signal

t 2 m k t k c k t k c1 t c o s c t k c o s c t

After LPF

w k t m k t k c k t k c1 t c o s k 1

After the integrator-sampler

Tb

Ik c o s k 1 mk t k c k t k c1 t

0

At Receiver

(t) =+/-1 (PSK), bit duration Tb

terfering signal may change amplitude at tk

k Tb

c o s k 1 b 1 ck t k c1 t d t b 0 c k

t k c1

0 k

Tb

User 1: I1 m 1 t c1 t c1 t d t

0

eally, spreading codes are Orthogonal:

Tb Tb

0

c1 t c1 t d t A

0

ck t k c1 t d t 0

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