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Equalization

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Three basic equalization methods

Linear equalization (LE)

Decision feedback equalization (DFE)

Sequence estimation (MLSE-VA)

Example of channel estimation circuit

Linear equalization (LE):

Performance is not very good when the frequency response

of the frequency selective channel contains deep fades.

Zero-forcing algorithm aims to eliminate the intersymbol

interference (ISI) at decision time instants (i.e. at the

center of the bit/symbol interval).

Least-mean-square (LMS) algorithm will be investigated in

greater detail in this presentation.

Recursive least-squares (RLS) algorithm offers faster

convergence, but is computationally more complex than

LMS (since matrix inversion is required).

Decision feedback equalization (DFE):

Performance better than LE, due to ISI cancellation of tails

of previously received symbols.

Decision feedback equalizer structure:

Feed-back

Feed-back

filter

filter(FBF)

(FBF)

Input

Feed-forward

Feed-forward

filter

filter(FFF)

(FFF)

Adjustment of

filter coefficients

Output

+

Symbol

decision

Maximum Likelihood Sequence Estimation using

the Viterbi Algorithm (MLSE-VA):

Best performance. Operation of the Viterbi algorithm can be

visualized by means of a trellis diagram with m K-1 states,

where m is the symbol alphabet size and K is the length of

the overall channel impulse response (in samples).

State trellis

diagram

Allowed transition

between states

State

Sample time

instants

Basic idea:

Raised

Raised

cosine

cosine

spectrum

spectrum

Z f B f H f E f

Transmitted

Transmitted

symbol

symbol

spectrum

spectrum

Channel

Channelfrequency

frequency

response

response

(incl.

(incl.TT&&RRfilters)

filters)

Equalizer

Equalizer

frequency

frequency

response

response

B f

H f

E f

Z f

0

fs = 1/T

Zero-forcing equalizer

Transmitted

impulse

sequence

Communication

channel

Overall

channel

2N+1

coefficients

h k

h k n

fk

2M+1

coefficients

c k

Coefficients of

equivalent FIR filter

Equalizer

z k

Input to

decision

circuit

n N

r k

m M

cm hk m

m M

cm k m

( M k M )

but the equalizer can only handle 2M+1 equations)

Zero-forcing equalizer

We want overall filter response

to be non-zero at decision time

k = 0 and zero at all other

sampling times k 0 :

fk

m M

cm hk m

1, k 0

0, k 0

h0 c M h1c M 1 ... h2 M cM 0

This leads to

a set of 2M+1

equations:

(k = M)

:

hM c M hM 1c M 1 ... h M cM 1

(k = 0)

:

h2 M 1c M h2 M 2c M 1 ... h1cM 0

h2 M c M h2 M 1c M 1 ... h0 cM 0

(k = M)

The aim is to minimize:

J E ek

Input to

decision

circuit

s k

Estimate

of k:th

symbol

Channel

r k

Error

ek

Equalizer

bk

z k

b k

zk

2

J E ek

equalizer coefficient values

J

c2

c1

equalizer coefficients (or one complexvalued coefficient)

or use an algorithm that recursively changes the equalizer

coefficients in the correct direction (towards the minimum

value of J)!

Wiener solution

We start with the Wiener-Hopf equations in matrix form:

Rc opt p

R = correlation matrix (M x M) of received (sampled) signal

values

rk

received signal values

coefficient values

2M+1 taps like in other parts of this presentation)

R E r k r *T k

where

r k rk , rk 1 ,..., rk M 1

*

p E r k bk

M samples

we must know the stochastical properties of the transmitted

signal (and of the channel if it is changing). Usually we do not

have this information => some non-stochastical algorithm like

Least-mean-square (LMS) must be used.

Algorithms

Stochastical information (R and p) is available:

1. Direct solution of the Wiener-Hopf equations:

Rc opt p

c opt R 1p

Inverting a large

matrix is difficult!

3. Method of steepest descent (this iterative algorithm is slow

but easier to implement)

Use an algorithm that is based on the received signal sequence

directly. One such algorithm is Least-Mean-Square (LMS).

Received complex

signal samples

rk M

c M

c1 M

Widrow

with 2M+1 filter taps

cM 1

of equalizer filter

adjustment of

tap coefficients

rk M

ek

cM

zk

bk

after symbol decision

r k

zk

Equalizer

Equalizerfilter

filter

e

Coefficient

Coefficient

updating

updating

bk

Phase

Phase

synchronization

synchronization

ek

Godard

Minimize:

J E ek

cm rk m exp j bk

m M

ek zk bk

(derived from method of steepest descent)

for convergence towards minimum mean square error (MMSE)

Re cn i 1 Re cn i

2

ek ek ek

2 2 M 1 1

equations

Phase:

i 1 i

Iteration index

Re cn

Im cn i 1 Im cn i

ek

ek

ek

Im cn

After some calculation, the recursion equations are obtained in

the form

Re cn i 1 Re cn i 2 Re e

Im cn i 1 Im cn i 2 Im e

r e j

c

r

b

m k m k k n

m M

r e j

c

r

m k m

k k n

m M

j M

i 1 i 2 Im bk e cm rk m

m M

ek

smaller

larger

Slow

Slow acquisition

acquisition

Poor

Poor stability

stability

Poor

Poor tracking

tracking

performance

performance

Large

Large variation

variation

around

around optimum

optimum

value

value

bk 1

q1

qQ 1

c M

c1 M

bk

FBF

rk M

bk Q

?

zk

rk M

cM 1

FFF

cM

ek

LMS

algorithm

for tap

coefficient

adjustment

The purpose is again to minimize J E ek

where

ek zk bk

m M

ek

cm rk m qn bk n bk

n 1

tap spacing smaller than symbol interval is allowed

=>

fractionally spaced equalizer

=> oversampling by a factor of 2 or 4 is common

Feedback filter (FBF) is used for either reducing or canceling

(difference: see next slide) samples of previous symbols at

decision time instants

tap spacing must be equal to symbol interval

The coefficients of the feedback filter (FBF) can be

obtained in either of two ways:

Recursively (using the LMS algorithm) in a similar

fashion as FFF coefficients

Proakis, Ed.3, Section 11-2

By calculation from FFF coefficients and channel

coefficients (we achieve exact ISI cancellation in

this way, but channel estimation is necessary):

qn

m M

cm hn m

n 1, 2, K , Q

Proakis, Ed.3, Section 10-3-1

Proakis, Ed.3, Section 11-3

Estimated

symbols

bk

LMS

algorithm

T

c0

c1 cM 1

cM

rk

rk

+

bk M

hm cm

1. Acquisition phase

Uses training sequence

Symbols are known at receiver, bk bk .

2. Tracking phase

Uses estimated symbols (decision directed mode)

Symbol estimates are obtained from the decision

circuit (note the delay in the feedback loop!)

Since the estimation circuit is adaptive, time-varying

channel coefficients can be tracked to some extent.

Alternatively: blind estimation (no training sequence)

Mandatory for MLSE-VA, optional for DFE

b k

Training

symbols

(no

errors)

r k

Estimated channel coefficients

Channel

Channel

estimation

estimation

circuit

circuit

h m

Equalizer

Equalizer

&& decision

decision

circuit

circuit

b k

(extension of DFE, for simulation of matched filter bound)

bk P K bk 1

bk 1 K bk Q

Precursor

Precursorcancellation

cancellation

of

offuture

futuresymbols

symbols

Postcursor

Postcursorcancellation

cancellation

of

ofprevious

previoussymbols

symbols

rk P

Filter

Filtermatched

matchedto

to

sampled

sampledchannel

channel

impulse

impulseresponse

response

bk

+

(impossible in a practical system), matched filter performance

can be achieved.

r t

Matched

Matched

filter

filter

NW

NW

filter

filter

y k

MLSE

MLSE

(VA)

(VA)

b k

f k

f k

Channel

Channel

estimation

estimationcircuit

circuit

before it is available for channel estimation

=> channel estimates may be out-of-date

(in a fast time-varying channel)

The probability of receiving sample sequence y (note: vector form)

of length N, conditioned on a certain symbol sequence estimate and

overall channel estimate:

Since we have AWGN

p y b, f p y k b , f

exp 2

N 2

N

k 1

2

2

Objective:

find symbol

sequence that

maximizes this

probability

noise samples are

uncorrelated due to NW

(= noise whitening) filter

Length of f (k)

y f b

k 1

K 1

n 0

k n

Metric to be

minimized

(select best b

..

using VA)

We want to choose that symbol sequence estimate and overall

channel estimate which maximizes the conditional probability.

Since product of exponentials <=> sum of exponents, the

metric to be minimized is a sum expression.

If the length of the overall channel impulse response in

samples (or channel coefficients) is K, in other words the time

span of the channel is (K-1)T, the next step is to construct a

state trellis where a state is defined as a certain combination

of K-1 previous symbols causing ISI on the k:th symbol.

f k

0

including response of matched

filter and NW filter

K-1

At adjacent time instants, the symbol sequences causing ISI

are correlated. As an example (m=2, K=5):

At time k-3

1 0 0 1 0

At time k-2

1 0 0 1 0 0

At time k-1

1 0 0 1 0 0 1

At time k

1 0 0 1 0 0 1 1

Bits

causing ISI

:

16 states

State trellis diagram

Number

of states

sequence is

estimated by means

of Viterbi algorithm

(VA)

m K 1

Alphabet

size

k-3

k-2

k-1

k+1

time instant, the VA selects the transition associated with

highest accumulated probability (up to that time instant) for

further processing.

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