3F4 Equalisation

Dr. I. J. Wassell
 Introduction
• When channels are fixed, we have seen that it is
possible to design optimum transmit and receive
filters, subject to zero ISI
• In practice, this is not usually possible,
– Ideal filters cannot be realised
– The channel responses can be unknown and/or time
varying
– The same transmitter may be used over many
different channels
 Introduction
• We can improve the situation by including an
additional filtering stage at the receiver. This is
known as an equalisation filter and usually it is
designed to reduce ISI to a minimum
• Equalisers may be categorised as,
– Fixed- The optimal equalisation filter is calculated
for a fixed (known) received pulse shape
– Adaptive- The filter is adapted continuously to the
changing characteristics of the channel
 Introduction
• Equalisation may be implemented using,
– Analogue filters- A traditional technique
mainly confined to fixed channels. Now
superseded by,
– Digital filters- Have all usual advantage of
digital systems, e.g. flexibility, reliability etc.
May be either fixed or adaptive. We will
consider fixed equalisers implemented as
digital filters
 Digital Filters
• An analogue signal x(t) is sampled at times
t=nT to give a ‘digital’ signal xn
xn  x(nT ), n  0,1,...... 
• The Z-transform of xn is defined analogously
to the Laplace transform of a continuous
signal as,

X ( z )   xn z  n
n 0
 FIR Filter
• A Finite Impulse Response (FIR) filter generates
a new digital signal yn from xn using delay,
multiply and addition operations
xn xn-1 xn-2 xn-q
D D D D
b0 b1 b2 bq
X X X X
yn
+
q
yn  xnb0  xn1b1  xn2b2  ........xnqbq   xni bi
i 0
Where bi are known as the filter coefficients and
delay D is equal to the sample (symbol) period T
 FIR Filter
• Taking the Z transform yields,
Y ( z )  X ( z )b0  X ( z ) z 1b1  X ( z ) z 2b2  .....  X ( z ) z  q bq


 X ( z ) b0  z 1b1  z  2b2  .....  z  q bq 
q
 X ( z ) z i bi
i 0
Where z-n may be taken to mean a delay of n sample periods
• Now,
Y ( z)  X ( z)H ( z)
• Hence the transfer function H(z) is,
q
Y ( z)
H ( z)    z i bi
X ( z ) i 0
 IIR Filters
• A recursive Infinite Impulse Response Filter
generates a new digital signal yn from the input xn
as follows, yn
xn yn-1 yn-2 yn-p
+ D D D D
a1 a2 ap
X X X

+
p
yn  xn  yn1a1  yn 2 a2  ........ yn  p a p  xn   yni ai
i 1
Where ai are known as the filter coefficients and
delay D is equal to the sample (symbol) period T
 IIR Filters
• Taking Z transform yields,
Y ( z )  X ( z )  Y ( z ) z 1a1  Y ( z ) z 2 a2  .....  Y ( z ) z  p a p
• Rearranging,
X ( z )  Y ( z )  Y ( z ) z 1a1  Y ( z ) z 2 a2  .....  Y ( z ) z  p a p

 Y ( z ) 1  z 1a1  z  2 a2  .....  z  p a p 
 p

 Y ( z )1   z ai 
i

 i 1 
• Now,
Y ( z) 1
H ( z)  
X ( z)  p
i 
1   z ai 
 i 1 
 Zero-Forcing Equalisers
• Suppose the received pulse in a PAM system is
p(t), which suffers ISI
• This signal is sampled at times t=nT to give a
digital signal pn=p(nT)
• We wish to design a digital filter HE(z) which
operates on pn to eliminate ISI
• Zero ISI implies that the filter output is only non-
zero in response to pulse n at sample instant n, i.e.
the filter output is the unit pulse n in response to pn
 Zero-Forcing Equalisers
• Note that the Z transform of n is equal to 1, so,
P( z ) H E ( z )  1
1
H E ( z) 
P( z )
• Now,
P( z )  p0 z 0  p1 z 1  p2 z 2  ...... Where pi are the

sample values of the
  pi z i isolated received pulse
• So, i 0

1 1 1
H E ( z)   1 2
 
P ( z ) p0 z  p1 z  p2 z  .....
0

 i
p
i 0
z i
 Zero-Forcing Equalisers
• We see that this expression has the form of an IIR
filter,
1
H E ( z) 
p0 z 0  p1 z 1  p2 z  2  .....
1

1  z 1a1  z 2 a2  .....  z  p a p
If,
That is we define the amplitude of the isolated
p0  1,
pulse at the optimum sampling point to be unity
a1   p1
a 2   p2 etc.
 FIR Approximations to ZFE
• IIR filters are difficult to deal with in practice
– stability is not guaranteed
– adaptive methods are difficult to derive
– Their recursive nature makes them prone to
numerical instability
• The simplest solution is to use an FIR
approximation to the ideal response
 Truncated Impulse Response
• A simple way to create an FIR
approximation is simply to truncate the
ideal impulse response
• However, this can give rise to significant
errors in the filter response
 Truncated Impulse Response
• The IIR response has the form,
1
H E ( z) 
p0 z 0  p1 z 1  p2 z  2  .....
• The FIR response has the form,
 
q
H ( z )  b0  z 1b1  z  2b2  .....  z  q bq   z i bi
i 0

• Thus we must perform polynomial division

to calculate the coefficients of H(z)
 Truncated Impulse Response
• Example
– The unequalised pulse response at the receiver in
response to a single unit amplitude transmitted pulse
at sample times k = 0, 1 and 2 is, p0= 1, p1= - 0.4 and
p2= - 0.2
Now,
1 1
H E ( z)  
P ( z ) p0 z 0  p1 z 1  p2 z  2  .....
So in this example,
1
H E ( z) 
1  0.4 z 1  0.2 z  2
 Truncated Impulse Response
• Performing the polynomial division,
1  0.4 z 1  0.36 z 2  0.224 z 3  0.1616 z 4  0.10944 z 5
1  0.4 z 1  0.2 z 2 1
1  0.4 z 1  0.2 z 2
0.4 z 1  0.2 z 2
0.4 z 1  0.16 z 2  0.08 z 3
0.36 z 2  0.08 z 3
0.36 z 2  0.144 z 3  0.072 z 4
0.244 z 3  0.072 z 4
0.224 z 3  0.0896 z 4  0.0448 z 5
0.1616 z 4  0.0448 z 5
0.1616z 4  0.06464z 5  0.03232z 6
Now, 0.10944 z 5  0.03232 z 6
H E ( z )  b0  z 1b1  z 2b2  .....  z  q bq
 1  0.4 z 1  0.36 z 2  0.224 z 3  0.1616z 4  .....  z  q bq
• Truncating to 5 terms gives FIR filter with the
coefficients: 1, 0.4, 0.36, 0.224, 0.1616
 Direct Zero Forcing
• The FIR filter equaliser output in the time
domain is,
q
yn  pnb0  pn1b1  pn2b2  ........ pnqbq   pni bi
i 0

• In the time domain, the zero forcing

constraint is yn= 1 for one value of n and
yn= 0 otherwise
 Direct Zero Forcing
• This constraint implies an infinite set of
simultaneous equations corresponding to,
n   ,....., 
• However, we only have q+1 filter coefficients,
so we set up q+1 equations in q+1 unknowns
and solve for the coefficients
 Direct Zero Forcing-Example
• The sampled received pulse in response to a
single binary ‘1’ is,
p0  0.8, p1  0.6, p2  0.3, 0 otherwise
• Design a 3-tap FIR equaliser to make the
response at n=0 equal to 1, and equal to
zero for n=1 and n=2
• The FIR equaliser filter output is,
2
yn  pnb0  pn1b1  pn2b2   pnibi
i 0
 Direct Zero Forcing-Example
• The zero forcing constraint is,
y0 = 1, y1 = 0, y2 = 0
• Write out previous equation for n=0, 1 and 2,
0.8b0  0b1  0b2  1 n0
0.6b0  0.8b1  0b2  0 n 1
0.3b0  0.6b1  0.8b2  0 n  2
• Solving these equations gives,
b0  1.25, b1  0.9375, b2  0.234
 Error Rates and Noise
• Equalisation is designed to reduce ISI and
hence increase the eye opening
• However, channel noise also passes through
the equaliser and must be handled carefully
to predict performance
• The frequency response of a digital filter
may be obtained by substituting,
j T
ze
 Error Rates and Noise
• The ideal ZFE has a response,
1
H E ( z) 
P( z )
• So in the frequency domain,
j T 1
H E (e )
P ( e j T )
• Thus at frequencies where P(ejwT) is small,
large noise amplification will occur.
 Error Rates and Noise
Received pulse spectrum Equaliser spectral response
P() HE() = 1/P()

0 0
 
• In this example the low pulse spectrum
response near zero will give rise to high
gain and noise enhancement by the
equaliser in this region.
 Error Rates and Noise
• What is the mean-square value (w)2 of the
noise at the equaliser output?
• Suppose the equaliser filter has impulse
response bn, (n=0,..,q).
• Consider the response of the equaliser to
noise alone, q
wn   bi vn i
i 0
 Error Rates and Noise
• The mean-squared value is,
   
q q
 w  E wn  E  bi1vn i1  bi 2 vn i 2 
2 2

i10 i 20 
q q
   bi1bi 2 E  vn i1vn i 2 
i1 0 i 2  0

• Assume that vn has a mean-squared value,

 
E vn2   v2
And that vn is uncorrelated white noise. Then
all the terms E[vnvm] will be zero except when
m=n, so,
 Error Rates and Noise
q
 w2   bi2 v2
i 0
q
  v2  bi2
i 0

• That is, the mean square noise at the filter

output is that at the input multiplied by the
sum squared of the filter impulse response
 Error Rates and Noise
• Hence the worst case BER may be calculated as
follows,
– Calculate the eye opening h for the equalised pulse
– Calculate the mean-squared noise power  w2
 h 
– Substitute into the BER expression, Q 
 2w 
 Error Rates and Noise- Example
• Returning to the previous example, calculate the
worst case BER after equalisation if unipolar line
coding with transmit levels of 1V and 0V is
employed.
• The sampled received pulse in response to a
single binary ‘1’ is,
p0  0.8, p1  0.6, p2  0.3, 0 otherwise
• The direct zero forcing solution is an FIR filter
with the following coefficient values,
b0  1.25, b1  0.9375, b2  0.234
Error Rates and Noise- Example
• We now need to calculate the worst case
eye opening for the equalised pulse.
• To do this we need to calculate the
‘residual’ values at the output of the
equaliser in response to a single received
pulse, pn
• From the earlier equations the FIR filter
(equaliser) output is given by,
q
yn  xnb0  xn1b1  xn2b2  ........xnqbq   xni bi
i 0
 Error Rates and Noise- Example
• In the example, the input sequence xn is the
single pulse pn and q = 2. In this case we have,
2
yn   pni bi  pnb0  pn1b1  pn2b2
i 0
• This direct convoulution yields,
y0  p0b0  0.8 1.25  1
y1  p1b0  p0b1  (0.6 1.25)  (0.8  0.9375 )  0
y2  p2b0  p1b1  p0b2  (0.3 1.25)  (0.6  0.9375 )  (0.8  0.234 )  0
y3  p2b1  p1b2  (0.3  0.9375 )  (0.6  0.234 )  0.141
y4  p2b2  (0.3  0.234 )  0.0702
• Thus the equalised pulse response is,
 ‘residuals’
 Error Rates and Noise- Example
• So, remembering that for a unipolar scheme
only ‘1’s give rise to residuals, the worst case
received ‘1’ is,
1 - 0.141 = 0.859 i.e., 1 other ‘1’ contributing
• The worst case ‘0’ is,
0 + 0.0702 = 0.0702 i.e., 1 other ‘1’ contributing
• The minimum eye opening h is,
h = 0.859 - 0.0702 = 0.789
 Error Rates and Noise- Example
• To calculate the rms noise at the output of
the equaliser we utilise,
q
Where v is the rms noise at the
w v i
b 2

i 0 equaliser input
• For our example,
2
w v i v 0 1 2 v
b 2

i 0
  b 2
 b 2
 b 2
  1 . 25 2
 0 .9375 2
 0 .234 2

 w  2.01 v
Showing that the noise power has been increased
 Error Rates and Noise- Example
• The probability of bit error is given by,
 h 
Pe  Q 
 2 w 
• Substituting for h and w gives,
 0.789   0.2 
Pe  Q   Q 
 2  2.01 v   v 
 Error Rates and Noise- Example
• Note that instead of performing the
convolution to give the equaliser output in
response to a single received pulse (and
hence determine the residuals), an
alternative is to multiply the pulse response
and equaliser response in the z domain, so
Y ( z )  P( z ) H E ( z )
 1
 p0  p1 z  p2 z 2
b  b z
0 1
1
 b2 z 2

 Error Rates and Noise- Example
Y ( z )  p0b0   p0b1  p1b0  z 1   p0b2  p1b1  p2b0  z 2   p1b2  p2b1  z 3  p2b2 z 4

Equating this to the expansion for Y(z),

Y ( z )  y0  y1 z 1  y2 z 2  y3 z 3  y4 z 4

Which yields the same expressions for the output sample

values yn obtained previously by direct convolution
 Other Equalisation Methods
• We have seen that with ZFEs, noise can be
amplified leading to poor BER performance
• Alternative design approaches take into account
noise as well as signal propagation through the
equaliser
• The Minimum Mean Squared Error (MMSE)
equaliser is one such approach
 MMSE Equaliser
• The MMSE explicitly accounts for the
presence of noise in the system
• Assuming a similar model to that used
previously, then in Z transform notation,
Y ( z )  H E ( z )( X ( z )  V ( z ))
X(z) Y(z)
+ HE(z)

V(z)
Where X(z) is the Z transform of the sampled received
signal xn, and V(z) is the Z transform of the noise vn
 MMSE Equaliser
• Ideally, the equalised output yn depends only on
the transmitted symbols ak. This is not possible
owing to the random noise, hence we choose to
minimise the total expected mean square error
(MSE) between yn and an with respect to the
equaliser HE(z), i.e.,
E[  y n  a n  ]
2
 MMSE Equaliser
MMSE equaliser formulation
xn yn
HE(z) - E[(.)2]
minimise
an

From data source

For a ‘fixed’ equaliser E[(.)2] is minimised by adjusting the

coefficients of HE(z). Effectively we have a trade off between
noise enhancement and ISI.
 MMSE Equaliser
• The solution has the form,
1
H E ( z) 
P( z )  N o
Where P(z) is the Z transform of the channel pulse
response and No is the noise PSD
• Note,
– The equaliser needs knowledge of the noise PSD
– If No=0, the solution is the same as the ZFE
– When noise is present the ZFE solution is modified to
make a trade-off between ISI and noise amplification
 Non-Linear Equalisation
• The equalisers considered so far are linear,
since they simply involve linear filtering
operations
• An alternative we consider now is non-
linear equalisation
• An example is the Decision Feedback
Equaliser (DFE)
 DFE
• The DFE is a non-linear filter.
• Again, its purpose is to cancel ISI.
• The non-linearity allows some of the noise
problems associated with linear equalisers
to be overcome
 DFE
• The structure is,
xn + ân
Slicer
pn (sampled - Detected
values of one symbols
+
pulse)
ap a2 a1
X X X

D D D

Where ai are known as the filter coefficients (not to

be confused with the transmitted symbols an) and
delay D is equal to the sample (symbol) period T
 DFE
• The DFE has almost the same structure as the
standard IIR filter based equaliser
• In the following development this relationship
is demonstrated
• We see that the only significant difference is
the position of the data slicer (decision block)
• A minor difference is the subtract function at
the DFE input. Its only effect is to alter the
sign of the DFE coefficients compared with
those in the IIR filter
 DFE Development
yn ân
IIR Structure
Slicer
xn
+ D D D D
a1 a2 ap
X X X

+
xn yn ân
+ Slicer
D D D D
ap a2 a1
X X X IIR

+
xn
DFE Development yn ân
+ Slicer
+
IIR
X ap X a2 X a1

D D D D
xn yn ân
+ Slicer
+ DFE

X ap X a2 X a1

D D D D
 DFE
• The DFE is almost a standard IIR filter
• For this structure we know that the ZFE
solution is,
1
H E ( z) 
p0 z 0  p1 z 1  p2 z  2  .....
Where p0, p1, etc. is the sampled response at the equaliser input
received in response to one transmitted symbol of unity
amplitude. Because we define the amplitude of the isolated
pulse at the optimum sampling point to be unity, then po = 1.
Comparing with the previous ZF solution where ai = -pi, this
time ai = pi owing to the subtract function at the DFE input.
• The outputs of this filter with no channel noise
are unit pulses, weighted by the transmitted
symbol amplitudes, an
 DFE Example
• The sampled response to an isolated received pulse
pn is given by,
p0 = 1, p1 = 0.5, p2 = -0.25
• Design a suitable DFE
• From the earlier work we see that the DFE
coefficients are given by ai = pi, so,
a1 = p1 = 0.5 and a2 = p2 = -0.25
• Assuming polar data pulses the effect is to
– add (subtract) 0.25 (if previous but one bit is a binary
one (zero)) to the current input value to remove its effect.
– subtract (add) 0.5 (if previous bit is a binary one (zero))
to the current input value to remove its effect.
• Thus the effect of the previous pulses is eliminated
 DFE Example
1 p(nT)

0.5 Sampled isolated pulse

0
T 2T 3T nT
-0.5
p0 = 1, p1 = 0.5, p2 = -0.25
xn + ân
Slicer
-
+
a2= p2 = -0.25 a1= p1 = 0.5
X X

D D
 DFE
• In noise, we have seen that noise amplification
occurs in the IIR filter approach
• To overcome this, the decision slicer is moved
inside the filter loop in the DFE
• The slicer outputs the symbol estimate âk which
is closest to the value at its input
• With no noise, this change makes no difference
because the ISI is cancelled perfectly by the IIR
filter, so the slicer input is ak anyway
 DFE
• However, in noise, the slicer acts to ‘clean-up’ the
signal, giving a noise free decision at its output.
• For example, the slicer input becomes ak+vk,
where vk is the noise value
• Provided that vk is small enough the slicer still
outputs the correct decision ak
• So error free cancellation continues without
problems of noise amplification
 DFE-Problems
• Consider what happens when vk is large enough
to cause an error in the slicer decision
• The error feeds back around the loop and so the
ISI is no longer cancelled.
• Often a long run or ‘burst’ of errors will then be
experienced- known as error propagation
• The length of the burst is of the order of 2N bits,
where N is the number of taps in the feedback
filter
 Automatic Equalisers
• In practical communication systems the channel
is often unknown and/or time varying
• To overcome this problem so called Automatic
Equalisers are employed
• Two approaches are used
– Preset equaliser: The channel is measured
periodically by sending some known data. The
equaliser coefficients are re-calculated and
subsequent data is equalised using the new
coefficients.
 Automatic Equalisers
–Adaptive equaliser: The coefficients are adapted
continuously based on the received data. A simple
approach uses the Least Mean Squares (LMS) algorithm
to adjust the coefficient values based on an error criterion.
This approach requires that the equaliser is initially
trained, so that the coefficients are initialised with
approximately the correct values

An alternative adaptation algorithm is known as

Recursive Least Square (RLS). This has the advantage of
faster training but at the cost of higher complexity
 Summary
• In this section we have seen
– That in practical systems it is difficult to arrange optimum TX
and RX filters subject to zero ISI
– That additional filters (equalisers) can be added to reduce ISI
and improve BER performance
– Equalisers can be implemented in an analogue or digital manner
and may be fixed or adaptive
– Digital implementations are fundamentally of IIR structure but
may be approximated by a truncated FIR filter
– The zero-forcing criterion may lead to noise enhancement and
poor performance
– MMSE and non-linear (DFE) approaches can reduce the
problem of noise enhancement