Equalization and

Interference Cancelation
Kalpana Dhaka
What is the need for equalization?
• Mitigate intersymbol interference (ISI) in Frequency selective channels to
obtain flat received signal response and a linear phase response.
• Equalization enhances the frequency components with small amplitudes and
attenuates the frequency components with high amplitudes
• High data rate systems (eg. TDMA systems) are more sensitive to delay
spread because for them rms delay spread 𝜎𝑟𝑚𝑠 is comparable to symbol
period 𝑇𝑠 or 𝜎𝑟𝑚𝑠 ≥ 𝑇𝑠 (or Coherence BW of channel ≤ BW of symbol).
• In low data rate applications (e.g. cordless phone), delay spread is small and
therefore equalization is not needed.
Equalizer Design: Features required
• Balance ISI mitigation with noise enhancement because both signal and
noise pass through an equalizer
1
Eg. Consider channel with impulse response 𝐻 𝑓 = , for 𝑓 < 𝐵 and noise power
𝑓
spectral density 𝑁0 2. Then noise power for 𝐵 = 30kHz without
𝑁 𝐵
equalizer is 𝑁0 𝐵 = 3 × 104 𝑁0 and with equalizer is 0 −𝐵 |( |𝑓|)2 | 𝑑𝑓 = 0.5𝑁0 𝐵2 =
2
4.5 × 108 𝑁0 . In this case, increase in noise is observed on equalization
• Must estimate channel impulse or frequency response to mitigate ISI
• For time varying channel, equalizer must learn impulse or frequency
response as the channel changes (training) and then update its estimate of
the frequency response as the channel changes (tracking): known as
Adaptive equalizer
Equalizer implementation
• Can be implemented at
• Baseband
• Carrier frequency
• Intermediate frequency
• Most equalizers are implemented at baseband because equalizers
(filters) designed for baseband are small, cheap, easily tunable, and
very power efficient
 End-to-end system Ideally𝑔𝑚 𝑡 = ℎ 𝑡 but c(t) is time varying
therefore it is not possible to have 𝑔𝑚 𝑡 = ℎ 𝑡 .
with equalizer In general, 𝑔𝑚 𝑡 is matched to g 𝑡 . Thus, the
performance is optimal for 𝑐 𝑡 = 𝛿(𝑡) and
suboptimal for 𝑐(𝑡) ≠ 𝛿(𝑡).
n(𝑡)
ℎ(𝑡)
w(𝑡) y(𝑡) y(𝑛)
Pulse ISI 𝑠𝑘 𝑠𝑘
𝑠𝑘 RF Front Matched Equalizer Decision
shape
filter g(t)
Channel
c(t)
+ End ∗ (−𝑡)
Filter 𝑔𝑚
Sampling
𝐻𝑒𝑞 (𝑧) Device

at 𝑇𝑆

- +
𝑝 𝑡 =𝑔 𝑡 ∗ (−𝑡)
∗ 𝑐 𝑡 ∗ 𝑔𝑚
Tap Update Σ
∗ (−𝑡) Algorithm
𝑦 𝑡 =𝑠 𝑡 ∗𝑝 𝑡 + 𝑛 𝑡 ∗ 𝑔𝑚
∞
where 𝑠 𝑡 = 𝑘=−∞ 𝑠𝑘 𝛿(𝑡 − 𝑘𝑇𝑆 )
therefore 𝑠 𝑡 ∗ 𝑝 𝑡 = ∞ 𝑘=−∞ 𝑠𝑘 𝑝(𝑡 − 𝑘𝑇𝑆 )
∞

𝑦 𝑡 = 𝑠𝑘 𝑝(𝑡 − 𝑘𝑇𝑆 ) + 𝑛𝑔 𝑡
𝑘=−∞
ISI Mitigation
∞

𝑦 𝑛 = 𝑦 𝑛𝑇𝑆 = 𝑠𝑘 𝑝(𝑛𝑇𝑆 − 𝑘𝑇𝑆 ) + 𝑛𝑔 𝑛𝑇𝑆

𝑘=−∞

= 𝑠𝑘 𝑝[𝑛 − 𝑘] + 𝑛𝑔 𝑛 = 𝑠𝑛 𝑝 0 + 𝑠𝑘 𝑝[𝑛 − 𝑘] + 𝑛𝑔 𝑛
𝑘=−∞ 𝑘≠𝑛

Desired data ISI Sampled noise

No ISI if 𝑝 𝑛 − 𝑘 = 0 for 𝑘 ≠ 𝑛. That is, 𝑝 𝑘 = 𝛿 𝑘 𝑓 0 .

ISI Mitigation (cont.)
∞

𝑝 𝑡 = 𝑝(𝑘𝑇𝑆 )𝛿(𝑡 − 𝑘𝑇𝑆 )

𝑘=−∞

1 ∞ 𝑘 1
𝑃 𝑓 = 𝑘=−∞ 𝑃 𝑓− is periodic with period
𝑇𝑆 𝑇𝑆 𝑇𝑆

∞ ∞

𝑃 𝑓 = 𝐹𝑇[𝑝(𝑡)] = 𝑝 𝑘𝑇𝑆 𝛿 𝑡 − 𝑘𝑇𝑆 𝑒 −𝑗2𝜋𝑓𝑡 𝑑𝑡

−∞ 𝑘=−∞

For no ISI, 𝑝 𝑘𝑇𝑆 = 0, 𝑘 ≠ 0

∞
𝑃(𝑓) = 𝑝 0 𝛿 𝑡 𝑒 −𝑗2𝜋𝑓𝑡 𝑑𝑡 = 𝑝 0 𝐹𝑇 𝛿 𝑡 =𝑝 0
−∞

1 ∞ 𝑘
Thus, for no ISI
𝑇𝑆 𝑘=−∞ 𝑃 𝑓−
𝑇𝑆
= 𝑝(0) or P 𝑓 = 𝑇𝑆 𝑝 0 𝑟𝑒𝑐𝑡 𝑓𝑇𝑆
ISI Mitigation (cont.)
• For no ISI
𝑡
𝑝 𝑡 = 𝐹 −1 𝑇𝑆 𝑝 0 𝑟𝑒𝑐𝑡 𝑓𝑇𝑆 = 𝑝 0 𝑠𝑖𝑛𝑐
𝑇𝑆
𝑝 𝑛𝑇𝑆 = 𝑝 0 𝑠𝑖𝑛𝑐 𝑛 = 𝑝 0 , for 𝑛 = 0
= 0, otherwise
• In general, 𝑝(𝑡) is not flat therefore equalizer 𝐻𝑒𝑞 (𝑧) is used to reduce
ISI.
Equalizers:
• Linear Equalizer
• No feedback path to adapt the equalizer
• Symbol-by-symbol (SBS) equalization: Remove ISI from each symbol and
detect each symbol individually
• Simple to implement
• Noise enhancement
• Nonlinear Equalizer
• Feedback to change the subsequent outputs of the equalizer
Linear Equalizer
n(𝑡)
ℎ(𝑡)
w(𝑡) y(𝑡) y(𝑛)
Pulse ISI 𝑠𝑘
𝑠𝑘 RF Front Matched Equalizer
shape
filter g(t)
Channel
c(t)
+ End ∗ (−𝑡)
Filter 𝑔𝑚 𝐻𝑒𝑞 (𝑧)
Linear Equalizer (cont.)
• 𝑁-tap transversal filter (𝑁 = 2𝐿 + 1)
𝐿

𝐻𝑒𝑞 𝑧 = 𝑤𝑖 𝑧 −𝑖
𝑖=−𝐿
• 𝑁 is dictated by implementation considerations: Large 𝑁 implies
higher complexity and delay
• Causal linear equalizer have 𝑤𝑖 = 0, for 𝑖 < 0.
• Equalizer design includes
• Tap weights 𝑤𝑖 , 𝑖 = {−𝐿, −𝐿 + 1, … , 0, … . , 𝐿 − 1, 𝐿}
• Algorithm to update tap weights
Linear Equalizer (cont.)
• In wireless system, probability of error and outage probability are
performance metrics which can be used to identify optimum choice of
filter coefficients.
• Above method is difficult to analyze, therefore other methods used are
• Zero-forcing (ZF) equalizer
• Removes all ISI but can lead to noise enhancement
• Minimum mean square error (MMSE) equalizer
• Minimize expected mean-squared error between the transmitted symbol and the symbol
detected at the equalizer output
• Provide better balance between ISI and noise mitigation
• System with MMSE equalizer has better BER performance
Zero-Forcing (ZF) Equalizer
• Z-transform of input at equalizer is
𝑌 𝑧 = 𝑆 𝑧 𝑃 𝑧 + 𝑁𝑔 (𝑧)
• ZF equalizer removes all ISI introduced in 𝑝(𝑡)
1
𝐻𝑍𝐹 𝑧 =
𝑃(𝑧)
• Noise power spectrum on ZF equalization
∗ (1/𝑧 ∗ )| 2 × |𝐻 (𝑧)|2
= 𝑁0 × |𝐺𝑚 𝑍𝐹

2
∗ (1/𝑧 ∗ )| 2
1 𝑁0
= 𝑁0 × |𝐺𝑚 × ∗ =
𝐻(𝑧)𝐺𝑚 (1/𝑧 ∗ ) |𝐻 𝑧 |2
ZF Equalizer (cont.)
• If 𝐻(𝑧) is highly attenuated at any frequency in the bandwidth of interest, noise
power increases significantly in case of ZF equalizer
1
• 𝐻𝑍𝐹 (𝑧) = may not be implementable as a finite-impulse response (FIR)
𝑃(𝑧)
𝐿 −𝑖
filter, 𝐻𝑍𝐹 𝑧 = 𝑖=−𝐿 𝑤𝑖 𝑧
• 𝐻𝑍𝐹 (𝑧) can be implemented as infinite-impulse response (IIR) filter 𝐻𝑍𝐹 𝑧 =
∞ −𝑖
𝑖=−∞ 𝑤 𝑖 𝑧 , where tap weights 𝑤𝑖 minimizes
2
𝐿
1
− 𝑤𝑖 𝑧 −𝑖
𝑃(𝑧)
𝑖=−𝐿
ZF Equalizer (cont.)
• Alternatively, set tap weights to minimize worst case ISI. This is a
convex optimization problem and it can be solved using technique like
steepest descent.
Minimum Mean Square Error (MMSE)
Equalizer
• Minimize the average mean square error (MSE) between the
transmitted symbol 𝑠𝑘 and its estimate 𝑠𝑘 (𝐸[𝑠𝑘 − 𝑠𝑘 ]2 ) at the output
of equalizer. In case of linear equalizer

𝑠𝑘 = 𝑤𝑖 𝑦[𝑘 − 𝑖]
𝑖=−𝐿
• Unlike ZF equalizer, MMSE equalizer is expended into components
• Noise-whitening component
• ISI-removal component
 MMSE Equalizer (cont.)
n(𝑡)
ℎ(𝑡)
w(𝑡) y(𝑡) y(𝑛) 𝐻𝑒𝑞 (𝑧)

Pulse ISI Noise

𝑠𝑘 RF Front Matched 𝑦𝑤 [𝑛] 𝐻𝑒𝑞 (𝑧) 𝑠𝑘
shape
filter g(t)
Channel
c(t)
+ End ∗ (−𝑡)
Filter 𝑔𝑚
Sampling
Whitener
1/𝐺𝑚∗ (1/𝑧 ∗ )

at 𝑇𝑆

Noise power spectrum at output of noise whitener is

2
∗ (1/𝑧 ∗ )| 2 ×
1
𝑁0 × |𝐺𝑚 ∗ = 𝑁0
𝐺𝑚 (1/𝑧 ∗ )
If 𝐻𝑒𝑞 (𝑧) is linear filter
𝐿

𝐻𝑒𝑞 (𝑧) = 𝑤𝑖 𝑧 −𝑖
𝑖=−𝐿
MMSE Equalizer (cont.)
• Output of MMSE equalizer is
𝑠𝑘 = 𝒘𝑇 𝒗 = 𝒗𝑇 𝒘
where
𝑦 … 𝑦 𝑇
𝑦
𝒗 = [ 𝑘+𝐿 𝑦𝑘+𝐿−1 𝑘+𝐿−2 𝑘−𝐿 ]
𝑤 … 𝑤 𝑇
𝒘=[ 𝑘 𝑤 𝑤𝑘−1 𝑘−2 𝑘−𝑁 ]
• In order to find equalizer tap weights, we minimize the average MSE
𝐸[𝑠𝑘 − 𝑠𝑘 ]2 . That is
𝐽 = 𝐸[(𝑠𝑘 −𝑠𝑘 )2 ] = 𝐸[|𝑠𝑘 |2 + 𝒘𝑇 𝒗𝒗𝐻 𝒘∗ − 2Re[𝒗𝐻 𝒘∗ 𝑠𝑘 ]
= 𝐸 |𝑠𝑘 |2 + 𝒘𝑇 𝑹𝒘∗ − 2Re[𝒓𝒘∗ ]
where 𝑹 = 𝐸 𝒗𝒗𝐻 and 𝒓 = 𝐸[𝒗𝐻 𝑠𝑘 ]
MMSE Equalizer (cont.)
• The optimum tap weight can be obtained by setting 𝜵𝒘 𝐽 = 0
𝑇
𝜕𝐽 𝜕𝐽 𝜕𝐽
𝜵𝒘 𝐽 = ,…, ,…, = 2𝒘𝑇 𝑹 − 2𝒓 = 0
𝜕𝑤−𝐿 𝜕𝑤0 𝜕𝑤𝐿

𝒘𝑜𝑝𝑡 = (𝑹𝑇 )−𝟏 𝒓𝑇

• On substituting optimum weights MMSE is
∗
𝐽𝑚𝑖𝑛 = 𝐸 |𝑠𝑘 |2 + ((𝑹𝑇 )−𝟏 𝒓𝑇 )𝑇 𝑹((𝑹𝑇 )−𝟏 𝒓𝑇 )∗ −2Re 𝒓 𝑹𝑇 −𝟏 𝒓𝑇

= 𝐸 |𝑠𝑘 |2 − 𝒓𝑹−1 𝒓𝐻
MMSE Equalizer (cont.)
• For an equalizer of infinite length,
𝒗 = [𝑦𝑘+∞ … 𝑦𝑘 … 𝑦𝑘−∞ ] 𝑇
𝑤 … 𝑤 𝑇
𝑤
𝒘 = [ −∞ … 0 ∞ ]
• 𝒘𝑇 𝑹= 𝒓 can be written as
∞

𝑤𝑖 𝑃 𝑗 − 𝑖 + 𝑁0 𝛿 𝑗 − 𝑖 ∗ −𝑗 ,
= 𝑔𝑚 −∞ ≤ 𝑗 ≤ ∞
𝑖=−∞
• Taking z-transform
∗ (1/𝑧 ∗ )
𝐻𝑒𝑞 𝑧 𝑃 𝑧 + 𝑁0 = 𝐺𝑚
MMSE Equalizer (cont.)
• Thus, Z-transform of MMSE equalizer
𝐻𝑒𝑞 𝑧 1
𝐻𝑒𝑞 𝑧 = =
∗ 1 𝑃 𝑧 + 𝑁0
𝐺𝑚 ∗
𝑧
• MMSE equalizer is same as ZF equalizer for 𝑁0 = 0
Linear Equalizer
• Transversal Structure Equalizer
• LMS
• RLS
• Fast RLS
• Square-Root Least Square
• Lattice Structure Equalizer
• Gradient RLS
Nonlinear Equalizer
• Decision Feedback Equalizer (DFE)
• Transversal Structure Equalizer
• Least Mean Square (LMS)
• Recursive Least Square (RLS)
• Lattice Structure Equalizer
• Gradient RLS
• Maximum Likelihood Sequence Estimator (MLSE)
• Transversal channel estimator
• LMS
• RLS
• Fast RLS
• Square-Root Least Square
DFE
• Relatively simple to implement and performs well
• Symbol-by-symbol (SBS) equalization: Remove ISI from each symbol
and detect each symbol individually
• Does not perform well for channels with low SNR
• It propagates error to subsequent stages on incorrect detection
 DFE
n(𝑡)
ℎ(𝑡)
w(𝑡) y(𝑡) y(𝑛)

𝑠𝑘 Pulse ISI RF Matched Feedforward + 𝑠𝑘 𝑠𝑘−1

shape
filter g(t)
Channel
c(t)
+ Front
End
Filter
∗ (−𝑡)
𝑔𝑚
Filter 𝑊(𝑧)
-
+ Decision
Device
Sampling
at 𝑇𝑆
Feedback
• Input for feedforward filter: Received sequence filter 𝑉(𝑧)
• Input for feedback filter: Previously detected sequence
• Feedback filter is strictly causal for the system to be stable
DFE (cont.)
• DFE determines ISI contribution from the detected symbols 𝑠𝑘−1 by passing them
through a feedback filter 𝑉(𝑧) that approximates the composite channel P(𝑧)
convolved with feedforward filter W(𝑧)
• Resulting ISI is subtracted from the incoming signals. In DFE, the estimate of
transmitted symbol is
0 𝑁2

𝑠𝑘 = 𝑤𝑖 𝑦 𝑘 − 𝑖 − 𝑣𝑖 𝑠𝑘−𝑖
𝑖=−𝑁1 𝑖=1

• Since FBF is in feedback loop, it must be strictly causal for system to be stable.
DFE (cont.)
• FBF of DFE does not suffer from noise enhancement because it
estimates channel frequency response rather than its inverse.
• DFE performs well for channels with deep spectral nulls than linear
equalizers
• The coefficients of FFF and FBF are selected using either ZF (remove
all ISI) or MMSE (minimize expected MSE between the DFE output
and the original symbol)
DFE (cont.)
• Assume MF at receiver is perfectly matched to channel, 𝑔𝑚 𝑡 = ℎ(𝑡)
• In order to obtain a causal filter for V(z), spectral factorization of
∗ 1 ∗ 1
𝐻 𝑧 𝐺𝑚 ∗ = 𝐻 𝑧 𝐻 ∗ is required.
𝑧 𝑧
1 1
• Let 𝐵 𝑧 is a causal filter such that 𝐻 𝑧𝐻∗ 𝐵∗
=𝐵 𝑧 =
𝑧∗ 𝑧∗
𝜆2 𝐵1 (𝑧)𝐵1∗ (1/𝑧 ∗ ), where 𝐵1 (𝑧) is normalized to have a leading
coefficient of 1.
• When 𝑊(𝑧) and 𝑉(𝑧) have infinite duration, then optimal FFF and
FBF for ZF DFE are
𝑊 𝑧 = 1/(𝜆2 𝐵1∗ (1/𝑧 ∗ )) and 𝑉 𝑧 = 1 − 𝐵1 (𝑧)
DFE (cont.)
• For MMSE DFE, E 𝑠𝑘 − 𝑠𝑘 is minimized.
• Let 𝑏𝑛 = 𝑏[𝑛] denotes the inverse z-transform of 𝐵(𝑧).
• Then using MMSE minimization, the coefficients of feedforward filter must
satisfy
0 ∗ 0 ∗
𝑞 𝑤
𝑖=−𝑁1 𝑙𝑖 𝑖 = 𝑏 −𝑙 for 𝑞𝑙𝑖 = 𝑗=−𝑙 𝑗 𝑏𝑗+𝑙−𝑖 + 𝑁0 𝛿[𝑙 − 𝑖]
𝑏
with 𝑖 = −𝑁1 , … , 0
• The coefficients of the feedback filter are then determined from the
feedforward coefficients
𝑣𝑘 = − 0𝑖=−𝑁1 𝑤𝑖 𝑏𝑘−𝑖
• These coefficients completely eliminate ISI if there is no decision error.
MLSE
• Optimum equalization technique
• Sequence estimator: Detect sequence of symbols, therefore effect of
ISI is part of estimation process
• Complexity grows exponentially with the length of the delay spread
• Impractical for most channels of interest because of high complexity
 MLSE (cont.)
n(𝑡)
ℎ(𝑡)
w(𝑡) y(𝑡) y(𝑛)
Pulse ISI 𝑠𝑘
𝑠𝑘 RF Front Matched MLSE
shape
filter g(t)
Channel
c(t)
+ End ∗ (−𝑡)
Filter 𝑔𝑚
Sampling
Algorithm
Delay
at 𝑇𝑆
+

𝑒
+ -
Channel
Estimation
MLSE Algorithm
𝑔(𝑡) and ℎ(𝑡) are known at receiver
MLSE (cont.)
• Using Gram-Schmidt orthonormalization procedure
𝑁

𝑤 𝑡 = 𝑤𝑛 𝜙𝑛 𝑡 , 𝑡𝜖[0, 𝐿𝑇𝑆 ]
𝑛=1
where {𝜙𝑛 𝑡 } is complete set of orthonormal basis
𝐿

𝑤𝑛 = 𝑠𝑘 ℎ𝑛𝑘 + 𝑣𝑛
𝑘=0
𝐿𝑇𝑆 𝐿𝑇𝑆
where ℎ𝑛𝑘 = 0
ℎ 𝑡− 𝑘𝑇𝑆 𝜙𝑛∗ 𝑡 𝑑𝑡 and 𝑣𝑛 = 0
𝑛(𝑡) 𝜙𝑛
∗ 𝑡 𝑑𝑡
∗ 𝑁0
with zero mean and covariance 𝐸 𝑣𝑛 𝑣𝑚 = 𝛿 𝑛 − 𝑚 .
2
MLSE (cont.)
• Therefore, weights of equalizer has multivariate Gaussian distribution
2
𝑁 𝐿
1 1
𝑝 𝒘 𝒔, ℎ 𝑡 = 𝑒𝑥𝑝 − 𝑤𝑛 − 𝑠𝑘 ℎ𝑛𝑘
𝜋𝑁0 𝑁0
𝑛=1 𝑘=0

• Using above equation, ML estimate of weights is

𝒔 = arg max[log 𝑝 𝒘 𝒔, ℎ 𝑡 ]
2
𝑁 𝐿

= arg max − 𝑤𝑛 − 𝑠𝑘 ℎ𝑛𝑘

𝑛=1 𝑘=0
MLSE (cont.)
𝑁 𝑁

𝒔 = arg max − 𝑤𝑛 2
+ 𝑤𝑛∗ 𝑠𝑘 ℎ𝑛𝑘 + 𝑤𝑛 𝑠𝑘∗ ℎ𝑛𝑘
∗

𝑛=1 𝑛=1 𝑘 𝑘
𝑁

− 𝑠𝑘 ℎ𝑛𝑘 𝑠𝑘∗ ℎ𝑛𝑘

∗

𝑛=1 𝑘 𝑘
𝑁 𝑁

= arg max 2 𝑅𝑒 𝑠𝑘∗ ∗

𝑤𝑛 ℎ𝑛𝑘 − ∗
𝑠𝑘 𝑠𝑚 ∗
ℎ𝑛𝑘 ℎ𝑛𝑚
𝑘 𝑛=1 𝑘 𝑚 𝑛=1
MLSE (cont.)
𝑁 ∞
∗
𝑤𝑛 ℎ𝑛𝑘 = 𝑤 𝜏 ℎ∗ 𝜏 − 𝑘𝑇𝑆 𝑑𝜏 = 𝑦[𝑘]
𝑛=1 −∞
and
𝑁 ∞
∗
ℎ𝑛𝑘 ℎ𝑛𝑚 = ℎ 𝜏 − 𝑘𝑇𝑆 ℎ∗ 𝜏 − 𝑚𝑇𝑆 𝑑𝜏 = 𝑓[𝑘 − 𝑚]
𝑛=1 −∞
where 𝑓 𝑡 = ℎ 𝑡 ∗ ℎ∗ (−𝑡)
Thus
𝒔 = arg max 2 𝑅𝑒 𝑠𝑘∗ 𝑦[𝑘] − ∗ 𝑓[𝑘 − 𝑚]
𝑠𝑘 𝑠𝑚
𝑘 𝑘 𝑚
MLSE (cont.)
• MLSE output is based on the channel output 𝑤 𝑡 only, the receiver
matched filter 𝑔𝑚 𝑡 = ℎ(𝑡) is optimal for MLSE detection
• Typically 𝑔𝑚 𝑡 = ℎ(𝑡) is optimal for detecting signals in AWGN, but
for this case it is optimal even in the presence of ISI
• Viterbi algorithm can be used for MLSE to reduce complexity.
However, complexity still grows exponentially with the channel delay
spread.
Fractionally Spaced Equalizer (FSE)
• MF matched to transmitted signal pulse results in significant
degradation in performance
• Such suboptimal filter is extremely sensitive to timing error
• FSE samples at least as fast as the Nyquist rate
• FSE compensates for channel distortion before aliasing effect occurs
due to symbol rate sampling
• FSE incorporates function of a matched filter and equalizer
Adaptive Equalizers
• In all the previous equalizer structure 𝑔 𝑡 and 𝑐(𝑡) are assumed to
known.
• Generally 𝑐(𝑡) is unknown or it changes over time 𝑐 𝜏, 𝑡 . Therefore,
system must periodically estimate the channel and update the equalizer
coefficients (training).
• Equalizer also use the detected data to adjust equalizer coefficients
(tracking).
Adaptive Equalizers (cont.)
• Training: Coefficients of equalizer are updated at time 𝑘 based on
known training sequence of length 𝑀 sent over the channel.
• Length of training sequence is dependent on
• Number of equalizer coefficients to be determined
• Convergence speed of the training algorithm
• Equalizer is retained for the coherence time 𝑇𝐶
• If 𝑀𝑇𝑆 > 𝑇𝐶 , then channel decorrelate before equalizer finish training
and equalization is not effective
Adaptive Equalizers (cont.)
• Let {𝑠𝑘 } be the transmitted training sequence and {𝑠𝑘 } be the estimate
of training sequence, then the equalizer coefficients at time 𝑘 + 1 is
updated using the training sequence received at time 𝑘.
• MSE between 𝑠𝑘 and 𝑠𝑘 is used to determine the equalizer coefficients
{𝑤−𝐿 𝑘 + 1 , … , 𝑤𝐿 𝑘 + 1 } at time 𝑘 + 1. MMSE algorithm has fast
convergence at the cost of high complexity.
• In case {𝑠𝑘 } is not known at receiver, properties of {𝑠𝑘 } are used to
estimate equalizer coefficients.
• Other less complex algorithms are least mean square (LMS).
Adaptive Equalizers (cont.)
• New weights = (previous weights)+constant × (previous error)
× (current input vector)
where
Previous error = Previous desired output –Previous actual output
• Constant is adjusted by algorithm in order to control variation in
equalizer coefficients in successive iterations
• The algorithm is repeated till the equalizer try to converge
• Once convergence is reached, the equalizer coefficients are frozen.
Blind Equalizers
• Do not use training, learn channel response through the detected data
only (tracking)
• Constant modulus algorithm
• Used for constant amplitude modulation
• Spectral Coherence Restoral Algorithm (SCORE)
• Exploits spectral redundancy or cyclostationarity property of the transmitted signal
Equalizer Structure
• Transversal
• Filter with 𝑁 − 1 delay elements and 𝑁 taps with tunable complex weights
• Types: Finite Impuse Response (FIR) and Infinite Impulse Response (IIR)

• Lattice
• Has complex recursive structure
• Has better numerical stability
• Has better convergence properties
• More flexibility in changing length
Comparison of Adaptive Equalization
Algorithms
References
• T. S. Rappaport, Wireless Communications: Principles and Practice
(2nd edition), Pearson Education, 2010.
• A. Goldsmith, Wireless communications, Cambridge University Press,
2005.