E 5 Eq
E 5 Eq
E 5 Eq
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superseded by,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 xn1b1 xn2b2 ........xnqbq xni 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 yn1a1 yn 2 a2 ........ yn p a p xn yni 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
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
i10 i 20
q q
bi1bi 2 E vn i1vn i 2
i1 0 i 2 0
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
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
D D D
+
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)
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