5CS5-11: Wireless Communication Department of ECE

UNIT IV

MULTIPATH MITIGATION TECHNIQUES

Equalisation – Adaptive equalization, Linear and Non-Linear equalization, Zero forcing and
LMS Algorithms. Diversity – Micro and Macrodiversity, Diversity combining techniques, Error
probability in fading channels with diversity reception, Rake receiver,

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INTRODUCTION

Equalization, diversity, and channel coding are three techniques which can be used independently
or in tandem to improve received signal quality.

Equalization compensates for intersymbol interference (ISI) created by multipath within time
dispersive channels. If the modulation bandwidth exceeds the coherence bandwidth of the radio
channel, ISI occurs and modulation pulses are spread in time. An equalizer within a receiver
compensates for the average range of expected channel amplitude and delay characteristics.
Equalizers must be adaptive since the channel is generally unknown and time varying.

Diversity is another technique used to compensate for fading channel impairments, and is usually
implemented by using two or more receiving antennas. As with an equalizer, the quality of a
mobile communications link is improved without increasing the transmitted power or bandwidth.
However, while equalization is used to counter the effects of time dispersion (ISI), diversity is
usually employed to reduce the depth and duration of the fades experienced by a receiver in a flat
fading (narrowband) channel.

Diversity techniques can be employed at both base station and mobile receivers. The most common
diversity technique is called spatial diversity, whereby multiple antennas are strategically spaced
and connected to a common receiving system. While one antenna sees a signal null, one of the
other antennas may see a signal peak, and the receiver is able to select the antenna with the best
signal at any time. Other diversity techniques include antenna polarization diversity, frequency
diversity, and time diversity. CDMA systems often use a RAKE receiver, which provides link
improvement through time diversity.

Channel coding improves mobile communication link performance by adding redundant data bits
in the transmitted message. At the baseband portion of the transmitter, a channel coder maps a
digital message sequence into another specific code sequence containing a greater number of bits
than originally contained in the message. The coded message is then modulated for transmission
in the wireless channel.

Channel coding is used by the receiver to detect or correct some (or all) of the errors introduced
by the channel in a particular sequence of message bits. Because decoding is performed after the
demodulation portion of the receiver, coding can be considered to be a post detection technique.
The added coding bits lowers the raw data transmission rate through the channel (expands the
occupied bandwidth for a particular message data rate). There are two general types of channel
codes: block codes and convolutional codes. Channel coding is generally treated independently
from the type of modulation used, although this has changed recently with the use of trellis
coded modulation schemes that combine coding and modulation to achieve large coding gains
without any bandwidth expansion.

The three techniques of equalization, diversity, and channel coding are used to improve radio link
performance (i.e. to minimize the instantaneous bit error rate), but the approach, cost, complexity,
and effectiveness of each technique varies widely in practical wireless communication systems.

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PRINCIPLE OF DIVERSITY

Diversity is a technique used to compensate for fading channel impairments, and is usually
implemented by using two or more receiving antennas.

Diversity is usually employed to reduce the depth and duration of the fades experienced by a
receiver in a flat fading (narrowband) channel. Diversity techniques can be employed at both base
station and mobile receivers.

The principle of diversity is to ensure that the same information reaches the receiver (RX) on
statistically independent channels.

The basic principle of diversity is that the RX has multiple copies of the transmit signal, where
each of the copies goes through a statistically independent channel.

In Rayleigh fading, the BER decreases only linearly with the SNR. The reason for this different
performance is the fading of the channel. A way to improve the BER is thus to change the effective
channel statistics. Diversity is a way to achieve this.

Consider the simple case of an RX with two antennas. The antennas are assumed to be far
enough from each other that small-scale fading is independent at the two antennas. As the signals
are statistically independent, the probability that both antennas are in a fading dip simultaneously
is low. The diversity thus changes the SNR statistics at the detector input.

Diversity is a powerful communication receiver technique that provides wireless link improvement
at relatively low cost. Diversity exploits the random nature of radio propagation by finding
independent (or at least highly uncorrelated) signal paths for communication.

Diversity

Microdiversity Macrodiversity

To combat small-scale fading To combat large-scale fading

There are two types of fading
1. Small-Scale fading
2. Large-Scale fading.

Small-scale fades are characterized by deep and rapid amplitude fluctuations. In order to prevent
deep fades from occurring, microscopic diversity techniques are used.

Large-scale fading is caused by shadowing due to variations in both the terrain profile and the
nature of the surroundings. In order to prevent large-scale fades from occurring, microscopic
diversity techniques are used.

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MACRODIVERSITY

Macrodiversity is used to combat large-scale fading due to shadowing. Macrodiversity can be

achieved by
1. Simulcast
2. On-frequency repeaters
Macrodiversity

On-frequency
Simulcast Repeaters

Same signal is transmitted Receives signal and

simultaneously from different retransmit an amplified
SIMULCAST

In simulcast, the same signal is transmitted simultaneously from different BSs. In cellular
applications the two BSs should be synchronized, and transmit the signals intended for a specific
user in such a way that the two waves arrive at the RX almost simultaneously.

Simulcast is also widely used for broadcast applications, especially digital TV. In this case, the
exact synchronization of all possible RXs is not possible. Each RX would require a different timing
advance from the TXs.
Disadvantage:
A large amount of signaling information has to be carried on landlines. Synchronization
information as well as transmit data has to be transported on landlines (or microwave links) to the
BSs.

ON-FREQUENCY REPEATERS

The simplest method for macrodiversity is the use of on-frequency repeaters that receive the
signal and retransmit an amplified version of it.
Advantage:
The use of on-frequency repeaters is simpler than that of simulcast, as no synchronization is
required.
Disadvantage:
Delay dispersion is larger, because
1. The runtime from BS to repeater, and repeater to MS is larger and
2. The repeater introduces additional delays due to the group delays of electronic
components, filters, etc.

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MICROSCOPIC DIVERSITY

Small-scale fades are characterized by deep and rapid amplitude fluctuations which occur as the
mobile moves over small distances. In order to prevent deep fades from occurring, microscopic
diversity techniques are used.

Microscopic
Diversity

Spatial Temporal Frequency Angular Polarization

Diversity Diversity Diversity Diversity Diversity

Microscopic Diversity Techniques

Microscopic diversity is used to combat small-scale fading. The five most common methods are
as follows:
Spatial diversity: several antenna elements separated in space.
Temporal diversity: transmission of the transmit signal at different times.
Frequency diversity: transmission of the signal on different frequencies.
Angular diversity: multiple antennas (with or without spatial separation) with different antenna
patterns.
Polarization diversity: multiple antennas with different polarizations (e.g. vertical and
horizontal).

The Space/ Spatial Diversity Techniques

Space diversity also known as antenna diversity, is one of the most popular forms of diversity used
in wireless systems. Conventional wireless systems consist of an elevated base station antenna and
a mobile antenna close to the ground. The existence of a direct path between the transmitter and
the receiver is not guaranteed and the possibility of a number of scatterers in the vicinity of the
mobile suggests a Rayleigh fading signal.

From this model, Jakes deduced that the signals received from spatially separated antennas on
the mobile would have essentially uncorrelated envelops for antenna separations of one-half
wavelength or more.

The concept of antenna space diversity is also used in base station design. At each cell site, multiple
base station receiving antennas are used to provide diversity reception. However, since the
important scatterers are generally on the ground in the vicinity of the mobile, the base station
antennas must be spaced considerably far apart to achieve decorrelation. Separations on the order
of several tens of wavelengths are required at the base station. Space diversity can thus be used at
either the mobile or base station, or both.

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Space diversity reception method can be classified into four categories.

 Selection diversity
 Feedback Diversity
 Maximal Ratio Combining
 Equal Gain Diversity

Selection diversity:

Selection diversity is the simplest diversity techniques similar to that shown in figure. Where m
demodulators are used to provide m diversity branches whose gains are adjusted to provide the
same average SNR for each branch. The receiver branch having the highest instantaneous SNR is
connected to the demodulator.

The antenna signals themselves could be sampled and the best one sends to a signal demodulator.
In practice, the branch with the largest (S+N)/N is used, since it is difficult to measure SNR alone.
A practical selection diversity system cannot function on a truly instantaneous basis. But must be
designed so that the internal time constants of the selection circuitry are shorter than the reciprocal
of the signal fading rate.

Feedback Diversity or scanning diversity

Scanning diversity is very similar to selection diversity except that instead of always using the best
of M signals, the M signals are scanned in a fixed sequence until one is found to be a above
predetermined threshold. This signal is then received until it falls below threshold and the scanning
process is again initiated. The resulting fading statistics are somewhat inferior to those obtained
by the other methods, but the advantage with this method is that it is very simple to implement –
only one receiver is required. A block diagram of this method is shown in figure

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Maximal Ratio Combining

In this method first proposed by Kahn, the signals from all of the M branches are weighted
according to their individual signal voltage to noise power ratios and summed. Figure shows a
block diagram of the technique.

Here, the individual signals must be co phased before being summed which generally requires an
receiver and phasing circuit for each antenna element. Maximal Ratio Combining produces an
output SNR equal to the sum of the individual SNRs. Thus, it has the advantage of producing an
output with an acceptable SNR even when none of the individual signals are themselves
acceptable. This technique gives the best statistical reduction of fading of any known linear
diversity combiner. Modern DSP techniques and digital receiver are now making this optimal form
of diversity practical.

Equal Gain combining:

In certain cases it is not convenient to provide for the variable weighting capability required for
true maximal ratio combining.

In such cases, the branch weights are all set to unity, but the signals from each branch are co phased
to provide the equal gain combining diversity. This allows the receiver to exploit signals that are
simultaneously received on each branch. The possibility of producing an acceptable signal from a
number of unacceptable inputs is still retained, and performance is only marginally inferior to
maximal ratio combining and superior to selection diversity.

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The Polarization, Time and Frequency Diversity Techniques

POLARIZATION DIVERSITY

At the base station, space diversity is considerably less practical than at the mobile because the
narrow angle of incident fields requires large antenna spacing. The comparatively high cost of
using space diversity at the base station prompts the consideration of the using orthogonal
polarization to exploit polarization diversity. While this only provides two diversity branches, it
does allow the antenna elements to the co-located.

In the early days of cellular radio, all subscriber units were mounted in vehicles and used vertical
whip antennas. Today, however, over half the subscriber units are portable. This means that most
of subscribers are no longer using vertical polarization due to hand-tilting when the portable
cellular phone is used. This recent phenomenon has sparked interest in polarization diversity at the
base station.

Measured horizontal and vertical polarization paths between a mobile and a base station are
reported to be uncorrelated by Lee and Yeh. The decorrelation for the signals in each polarization
is caused by multiple reflections in the channel between the base station and antennas. The
reflection coefficient for each polarization is different, which results in different amplitudes and
phases for each, or at least some, of the reflections. After sufficient random reflections, the
polarization state of the signal will be independent of the transmitted polarization. In practice,
however, there is some dependence of the received polarization on the transmitted polarization.

Circular and linear polarization antennas have been used to characterize multipath inside buildings.
When the path was obstructed, polarization diversity was found to dramatically reduce the
multipath delay spread without significantly decreasing the received power. While polarization
diversity has been studied in the past, it has primarily been used for fixed radio linkswhich vary
slowly in time. Line-of-sight microwave links, for example, typically use polarization diversity to
support two simultaneous users on the same radio channel. Since the channel does not change
much in such a link, there is little likelihood of cross polarization interference. As portable users
proliferate, polarization diversity is likely to become more important for improving link margin
and capacity. An outline of theoretical model for the base station polarization diversity reception
as suggested by Kozono is given below.

Theoretical model for Polarization Diversity

It is assumed that the signal is transmitted from a mobile with vertical (or horizontal) polarization.
It is received at the base station by a polarization diversity antenna with two branches.

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V2 V1

X
X
Multipath
α
Main beam
Mobile

Figure shows the theoretical model and the system coordinates. As seen in the figure, polarization
diversity antenna is composed of two antenna elements V 1 and V2, which make + angle
(polarization angle) with the Y axis. A mobile station is located in the direction of offset angle
from the main beam direction of the diversity antenna as seen in figure above.

Some of the vertically polarized signals transmitted are converted to the horizontal polarizatied
signal because of multipath propagation. The signal arriving at the base station can be expressed
as
x = r1 cos (t + 1)
y = r2 cos (t + 2)

where x and y are signal levels which are received when =0. it is assumed that r1 and r2 have
independent Rayleigh distributions, and 1 and 2 have independent uniform distributions.

The received signal values at elements V1 and V2 can be written as:

V1 = (a r1 cos 1 + r2 b cos 2) cos t - (a r1 sin 1 + r2 b sin 2) sin t

V2 = (- a r1 cos 1 + r2 b cos 2) cos t - (- a r1 sin 1 + r2 b sin 2) sin t

a=sin  cos  and b=cos 


The correlation coefficient  can be written as

 = ( (tan 2 () cos 2 () -  ) / (tan 2 () cos 2 () +  ) ) 2

where, 

X=  R 2 2  /  R 1 2  And

R1 = ( r 12 2
a + r 22 b + 2 r 1 r2 ab cos(1 + 2) ) (1/2)
2

Here, X is the cross polarization discrimination of the propagation path between a mobile and a
base station.
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The correlation coefficient is determined by three factors: Polarization angle, Offset angle from
the main beam direction of the diversity antenna, and the cross polarization discrimination. The
correlation coefficient generally becomes higher as offset angle offset angle  becomes larger.
Also,  generally becomes lower as polarization angle  increases. This is because the horizontal
polarization component becomes larger as  increases.

Because antenna elements V1 and V2 are polarized at + to the vertical, the received signal is lower
than that received by a vertically polarized antenna. The average value of signal loss L, relative to
that received signal using vertical polarization is given by

L= a2 / X + b 2

The results of practical experiments carried out using polarization diversity show that polarization
diversity is viable diversity reception technique, and is exploited within wireless handsets as well
as at base stations.

Frequency Diversity
Frequency diversity is implemented by transmitting information on more than one carrier
frequency. The rationale behind this technique is that frequencies separated by more than the
coherence bandwidth of the channel will be uncorrelated and will thus not experience the same
fades. Theoretically, if the channels are uncorrelated, the probability of simultaneous fading will
be the product of the individual fading probabilities.

Frequency diversity is often employed in microwave line-of-sight links which carry several
channels in a frequency division multiplex mode (FDM).

Due to tropospheric propagation and resulting refraction, deep fading sometimes occurs. In
practice, 1:N protection switching is provided by a radio licensee, wherein one frequency is
nominally idle but is available on a stand-by basis to provide frequency diversity switching for any
one of the N other carriers (frequencies) being used on the same link, each carrying independent
traffic. When diversity is needed, the appropriate traffic is simply switched to the backup
frequency.

This technique has the disadvantages that it is not only requires spare bandwidth but also requires
that there be as many receivers as there are channels used for the frequency diversity. However,
for critical traffic, the expense may be justified.

Repeating the same information at two different frequencies will decrease spectral efficiency. So
information is spread over a large bandwidth, so that small parts of the information are conveyed
by different frequency components. The RX can then sum over the different frequencies to recover
the original information.

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This spreading can be done by different methods:

1. Compressing the information in time
2. Code Division Multiple Access (CDMA).
3. Multicarrier CDMA and coded orthogonal frequency division multiplexing
4. Frequency hopping in conjunction with coding

New OFDM modulation and access techniques exploit frequency diversity by providing
simultaneous modulation signals with error control coding across a large bandwidth, so that if a
particular frequency undergoes a fade, the composite signal will still be demodulated.

TIME DIVERSITY
Time diversity repeatedly transmits information at time spacing that exceed the coherence time
of the channel, so that multiple repetitions of the signal will be received with independent fading
conditions, thereby providing for diversity. One modern implementation of time diversity involves
the use of the RAKE receiver for spread spectrum CDMA, where the multipath channels provide
a redundancy in the transmitted message. By demodulating several replicas of the transmitted
CDMA signal, where each replica experience a particular multipath delay, the RAKE receiver is
able to align the replicas so that a better estimate of the original signal may be formed at the
receiver.

The wireless propagation channel is time variant. So the signals that are received at different times
are uncorrelated. For sufficient decorrelation, the temporal distance must be at least 1/(2𝑣max),
where 𝑣𝑚𝑎𝑥 is the maximum Doppler frequency.

Temporal diversity can be realized in different ways:

Temporal
diversity

Automatic Repeat Combination of

Repetitive Coding
Request interleaving &

Signal is repeated Rx sends a message to Different symbols of a

several times TX to indicate whether it codeword are transmitted
received the data with at different times
sufficient quality
Repetition coding

This is the simplest form. The signal is repeated several times, where the repetition intervals are
long enough to achieve decorrelation.
Disadvantages:
1. It is highly bandwidth inefficient.
2. Spectral efficiency decreases by a factor that is equal to the number of repetitions.

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Automatic Repeat reQuest (ARQ)

The RX sends a message to the TX to indicate whether it received the data with sufficient
quality. If this is not the case, then the transmission is repeated after a wait period.

Advantage: The spectral efficiency of ARQ is better than that of repetition coding, since it requires
multiple transmissions only when the first transmission occurs in a bad fading state.

Disadvantage: ARQ requires a feedback channel.

Combination of interleaving and coding

Advanced version of repetition coding is forward error correction coding with interleaving. The
different symbols of a code word are transmitted at different times. This increases the probability
that at least some of them arrive with a good SNR. The transmitted code word can then be
reconstructed.

ANGLE DIVERSITY

A fading dip is created when MPCs interfere destructively. If some of these waves are attenuated
or eliminated, then the location of fading dips changes In other words, two co-located antennas
with different patterns see differently weighted MPCs, so that the MPCs interfere differently for
the two antennas. This is the principle of angle diversity (also known as pattern diversity).

Angular diversity is usually used in conjunction with spatial diversity. Different antenna patterns
can be achieved very easily. Different types of antennas have different patterns. But evenidentical
antennas can have different patterns when mounted close to each other. This effect is due to
mutual coupling.

Antenna B acts as a reflector for antenna A, whose pattern is therefore skewed to the left. Similarly,
the pattern of antenna B is skewed to the right due to reflections from antenna A. Thus, the two
patterns are different.

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COMBINATION OF SIGNALS

Two ways of exploiting signals from the multiple diversity branches are
1. Selection diversity
2. Combining diversity

Combination of Signals

Selection Diversity Combining Diversity

RSSI-Driven Maximum Ratio Combining

BER-Driven Equal Gain Combining
Optimum
Hybrid-Selection

Combination of Signals
Selection diversity: In selection diversity the best signal copy is selected and processed
(demodulated and decoded), while all other copies are discarded.
Advantage: Requires only one RF (down conversion) chain.
Disadvantage: Selection diversity wastes signal energy by discarding (𝑁𝑟 − 1) copies of the
received signal

Combining diversity: In combining diversity all copies of the signal are combined (before or
after the demodulator), and the combined signal is decoded.
Advantage: Combining diversity leads to better performance, as all available information is
exploited.
Disadvantage: It requires a more complex RX than selection diversity. A RX with combining
diversity needs to downconvert all available signals, and combine them appropriately in the
baseband. Thus, it requires 𝑁𝑟 antenna elements as well as 𝑁𝑟 complete Radio Frequency (RF)
(downconversion) chains.

SELECTION DIVERSITY
Received-Signal-Strength-Indication-Driven Diversity
In this method, the RX selects the signal with the largest instantaneous power (or Received Signal
Strength Indication – RSSI), and processes it further. This method requires 𝑁𝑟 antenna elements,
Nr RSSI sensors, and a 𝑁𝑟-to-1 multiplexer (switch), but only one RF chain.

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The method allows simple tracking of the selection criterion even in fast-fading channels. Thus,
we can switch to a better antenna as soon as the RSSI becomes higher there.

1. If the BER is determined by noise, then RSSI-driven diversity is the best of all the selection
diversity methods, as maximization of the RSSI also maximizes the SNR.
2. If the BER is determined by co-channel interference, then RSSI is not a good selection
criterion.
3. Similarly, RSSI-driven diversity is suboptimum if the errors are caused by the frequency
selectivity of the channel.

Assume that the instantaneous signal amplitude is Rayleigh distributed. SNR distribution of nth
diversity branch is given by

where 𝛾̅ is the mean branch SNR. The cumulative distribution function (cdf) is then

The cdf of the selected signal is the product of the cdfs of each branch

Bit-Error-Rate-Driven Diversity

For BER-driven diversity, we first transmit a training sequence – i.e., a bit sequence that is known
at the RX. The RX then demodulates the signal from each receive antenna element and compares
it with the transmit signal. The antenna whose signal results in the smallest BER is judged to be
the best, and used for the reception of data signals.

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If the channel is time variant, the training sequence has to be repeated at regular intervals and
selection of the best antenna has to be done anew. The necessary repetition rate depends on the
coherence time of the channel.

Drawbacks:

BER-driven diversity has several drawbacks:

1. The RX needs either Nr RF chains and demodulators (which makes the RX more complex), or
the training sequence has to be repeated Nr times (which decreases spectral efficiency), so that
the signal at all antenna elements can be evaluated.
2. If the RX has only one demodulator, then it is not possible to continuously monitor the
selection criterion (i.e. the BER) of all diversity branches. This is critical if the channel changes
quickly.
3. Since the duration of the training sequence is finite, the selection criterion – i.e., bit error
probability – cannot be determined exactly.

COMBINING DIVERSITY
Basic Principle
Selection diversity wastes signal energy by discarding (𝑁𝑟 − 1) copies of the received signal. This
drawback is avoided by combining diversity, which exploits all available signal copies. Each
signal copy is multiplied by a (complex) weight and then added up. Each complex weight
𝑤𝑛* can be thought of as consisting of a phase correction, plus a (real) weight for the amplitude:

Phase correction causes the signal amplitudes to add up, and noise is added incoherently, so that
noise powers add up.

For amplitude weighting, two methods are widely used:

1. Maximum ratio combining: Maximum Ratio combining weighs all signal copies by
their amplitude. This is an optimum combination strategy.
2. Equal gain combining: In Equal Gain Combining, all amplitude weights are the same
(in other words, there is no weighting, but just a phase correction).

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Maximum ratio combining

MRC compensates for the phases, and weights the signals from the different antenna branches
according to their SNR.

Let us assume a propagation channel that is slow fading and flat fading. The only disturbance is
AWGN. Under these assumptions, each channel realization can be written as a time-invariant filter
with impulse response:
hn ( )  n ( )
where 𝛼𝑛 is the gain of diversity branch n.

The signals at the different branches are multiplied with weights 𝑤𝑛* and added up. The SNR is
maximized by choosing the weights as
wMRC  n
The output SNR of the diversity combiner is the sum of the branch SNRs:
Nr
 MRC    n
n1

If the SNR distribution in each branch is exponential (corresponding to Rayleigh fading), and all
branches have the same mean SNR

and the mean SNR of the combiner output is

Equal Gain Combining

In Equal Gain Combining, all amplitude weights are the same (in other words, there is no
weighting, but just a phase correction

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For EGC, we find that the SNR of the combiner output is

The mean SNR of the combiner output can be found to be

Optimum Combining
One of the assumptions in the derivation of MRC was that only AWGN disturbs the signal. If it
is interference that determines signal quality, then MRC is no longer the best solution. In order to
maximize the Signal-to-Interference-and-Noise Ratio (SINR), the weights should then be
determined according to a strategy called optimum combining.

The correlation matrix of noise and interference at the different antenna elements:
K

R   n I  E[r krk ]
2 *

k 1
where rk is the receive signal vector of the kth interferer. The complex transfer function of the Nr
diversity branches are written into the vector h𝑑.

The vector containing the optimum receive weights is then

wopt  R1hd

Hybrid Selection – Maximum Ratio Combining

In hybrid selection scheme, the best L out of Nr antenna signals are chosen, downconverted, and
processed. This reduces the number of required RF chains from 𝑁𝑟 to L, and thus leads to
significant savings. The approach is called Hybrid Selection/Maximum Ratio Combining (H-
S/MRC), or Generalized Selection Combining (GSC).

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The instantaneous output SNR of H-S/MRC is

Nr
 H S / MRC   (n)
n1

where 𝛾̅(𝑛) are ordered SNRs - 𝛾̅(1) > 𝛾̅(2) > ⋯ > 𝛾̅(𝑁r)

H-S/MRC schemes provide good diversity gain, as they select the best antenna branches for
combining. However, they do not provide full beamforming gain. the SNR gain of H-S/MRC is
only L, compared with 𝑁𝑟 for an MRC scheme.

The mean and variance of the output SNR for H-S/MRC is thus
 Nr
1 
 H S / MRC  L 1   


and
 nL1 n 

 Nr
1 
 2H S / MRC  L 1   2  2

 nL1 n 
 

RAKE RECEIVER CONCEPTS

In CDMA spread spectrum systems, the chip rate is typically much greater than the flat-fading
bandwidth of the channel. Whereas conventional modulation techniques require an equalizer to
undo the intersymbol interference between adjacent symbols, CDMA spreading codes are
designed to provide very low correlation between successive chips. Thus, propagation delay spread
in the radio channel merely provides multiple versions of the transmitted signal at the receiver.

If these multipath components are delayed in time by more than chip duration, they appear like
uncorrelated noise at a CDMA receiver, and equalization is not required. The spread spectrum
processing gain makes uncorrelated noise negligible after dispreading.

However, since there is useful information in the multipath components, CDMA receivers may
combine the time delayed versions of the original signal transmission in order to improve the SNR
at the receiver. A RAKE receiver does just this-it attempts to collect the time-shifted versions of
the original signal by providing a separate correlation receiver for each of the multipath signals.

Each correlation receiver may be adjusted in time delay, so that a microprocessor controller can
cause different correlation can cause different correlation receivers to search in different time
windows for significant multipath. The range of time delays that a particular correlator can search
is called a search window.

The RAKE receiver is essentially a diversity receiver designed specifically for CDMA, where the
diversity is provided by the fact that the multipath components are practically uncorrelated from
one another when their relative propagation delays exceed a chip period.

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Z1

CORRELATOR 1 α1
IF or
baseband
CDMA T
Z2
signal
CORRELATOR 2 α2
Σ ∫(.)dt < m’(t)
multipath 0 >

ZM αM
CORRELATOR M

A RAKE receiver utilizes multiple correlators to separately detect the M strongest multipath
components. The outputs of each correlators are then weighted to provide a better estimate of the
transmitted signal than is provided by a single component. Demodulation and bit decisions are
then based on the weighted outputs of the M correlators.

The basic idea of a RAKE receiver was first proposed by Price and Green. In outdoor
environments, the delay between multipath components is usually large and, if the chip rate is
properly selected, the low autocorrelation properties of a CDMA spreading sequence can assure
that multipath components will appear nearly uncorrelated with each other. However, the RAKE
receiver in IS-95 CDMA has been found to perform poorly in indoor environments, which is to be
expected since the multipath delay spreads in indoor channels (approximately 100 ns) are much
smaller than an IS-95 chip duration (approximately 800 ns). In such cases, a RAKE willnot
work since multipath is un resolvable, the Rayleigh flat-fading typically occurs within a single
chip period.

To explore the performance of the RAKE receiver, assume M correlators are used in a CDMA
receiver to capture the M strongest multipath components. A weighting network is used to provide
a linear combination of the correlator output for bit detection. Correlator 1 is synchronized to the
strongest multipath m1. Multipath component m2 arrives 1 later than component m1 . Where 2 -
1 is assumed to be greater than chip duration. The second Correlator is synchronized to m2. It
correlates strongly with m2, but has low correlation with m1. Note that ifonly a single correlator is
used in the receiver, once the output of the single correlator is corrupted by fading, the receiver
cannot correct the value. Bit decision based on only a single correlation may produce large bit
error . in a RAKE receiver, if the output from one from is corrupted by fading, the others may not
be, and the corrupted signal may be discounted through the weighting process. Decisions based on
the combination of the M, separate decision statistics offered by the RAKE provide a form of the
diversity which can overcome fading and thereby improves CDMA reception.

The M decision statistics are weighted to form an overall decision statistics as shown in figure.
The outputs of the M correlators are denoted as Z1, Z2, ................ ZM. They are weighted by 1,

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2, 3 ……….M respectively. The weighting coefficients are based on the power or the SNR
from each correlator output. If the power or SNR is small out of a particular correlator, it will be
assigned a small weighting factor just as in the case of a maximal ratio combining diversity
scheme, the overall signal Z’ is given by

The weighting coefficients, M are normalized to the output signal power of the correlator in such
a way the coefficients sum to unity,

As in the case of adaptive equalizer and diversity combining there are many ways to generate the
weighting coefficients. However, due to multiple access interference, RAKE fingers with strong
multipath amplitudes will not necessarily provide strong output after correlation. Choosing
weighting coefficients based on the actual outputs of the correlators yields better RAKE
performance.

EQUALIZERS
INTRODUCTION

Wireless channels can exhibit delay dispersion. Delay dispersion leads to Inter-Symbol
Interference (ISI). Equalization is a technique used to combat Inter-Symbol Interference.

Equalizers are RX structures that reduce or eliminate ISI, and at the same time exploit the delay
diversity inherent in the channel.

The operational principle of an equalizer can be visualized either in the time domain or the
frequency domain.

In radio channels, a variety of adaptive equalizers can be used to cancel interference while
providing diversity. Since the mobile fading channel is random and time varying, equalizers must
track the time varying characteristics of the mobile channel, and thus are called adaptive
equalizers.

The general operating modes of an adaptive equalizer include training and tracking. First,
a known, fixed-length training sequence is sent by the transmitter so that the receiver's
equalizer may average to a proper setting. The training sequence is typically a
pseudorandom binary signal or a fixed, prescribed bit pattern. Immediately following this
training sequence, the user data (which may or may not include coding bits) is sent, and
the adaptive equalizer at the receiver utilizes a recursive algorithm to evaluate the
channel and estimate filter coefficients to compensate for the channel. The training

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sequence is designed to permit an equalizer at the receiver to acquire the proper filter
coefficients in the worst possible channel conditions so that when the training sequence
is finished, the filter coefficients are near the optimal values for reception of user data.
As user data are received, the adaptive algorithm of the equalizer tracks the changing
channel. As a consequence, the adaptive equalizer is continually changing its filter
characteristics over time.

Time division multiple access (TDMA) wireless systems are particularly well suited for
equalizers. TDMA systems send data in fixed-length time blocks, and the training sequence
is usually sent at the beginning of a block. Each time a new data block is received, the
equalizer is retrained using the same training sequence.

An equalizer is usually implemented at baseband or at IF in a receiver. Since the

baseband complex envelope expression can be used to represent bandpass waveforms, the
channel response, demodulated signal, and adaptive equalizer algorithms are usually
simulated and implemented at baseband.

Figure shows a block diagram of a communication system with an adaptive equalizer in the
receiver.

If x (t) is the original information signal, and f(t) is the combined complex baseband
impulse response of the transmitter, channel, and the RF/IF sections of the receiver, the
signal received by the equalizer may be expressed as
𝑦(𝑡) = 𝗑(𝑡) 𝑓* (𝑡) + 𝑛𝑏(𝑡)

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where f* (t) is the complex conjugate of f(t). 𝑛𝑏(𝑡) is the baseband noise at the input of
the equalizer, and denotes the convolution operation. If the impulse response of the
equalizer is heq(t), then the output of the equalizer is

𝛿(𝑡) = 𝑥(𝑡) ƒ* (𝑡) ℎ𝑒𝑞(𝑡) + 𝑛𝑏(𝑡) ℎ𝑒𝑞(𝑡)

= 𝑥(𝑡) 𝑔(𝑡) + 𝑛𝑏(𝑡) ℎ𝑒𝑞(𝑡)

Where g (t) is the combined impulse response of the transmitter, channel, RF/IF sections of
the receiver, and the equalizer. The complex baseband impulse response of a
transversal filter equalizer is given by

ℎ𝑒𝑞(𝑡) = ∑ 𝑐𝑛𝛿(𝑡 − 𝑛𝑇)

𝑛

where 𝑐𝑛 are the complex filter coefficients of the equalizer. The desired output of the
equalizer is x(t), the original source data. Assume that 𝑛𝑏(𝑡) = 0 Then, in order to force V(𝑡)
=
𝑑 𝑥(𝑡) in equation, g (t) must be equal to

𝑔(𝑡) = 𝑓* (𝑡) ℎ𝑒q(𝑡) = ð( 𝑡)

𝐻𝑒𝑞(ƒ)𝐹*(−ƒ) = 1

Where 𝐻𝑒q(𝑓) and F (f) are Fourier transforms of heq (t) and f (t), respectively.

Equation indicates that an equalizer is actually an inverse filter of the channel. If the channel is
frequency selective, the equalizer enhances the frequency components with small amplitudes and
attenuates the strong frequencies in the received frequency spectrum in order to provide a flat,
composite, received frequency response and linear phase response. For a time-varying channel, an
adaptive equalizer is designed to track the channel variations so that equation is approximately
satisfied.

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A Generic Adaptive Equalizer

An adaptive equalizer is a time-varying filter which must constantly be retuned. The basic structure
of an adaptive equalizer is shown in Figure, where the subscript k is used to denote a discrete time
index

Notice in that there is a single input Yk at any time instant. The value of Yk depends upon the
instantaneous state of the radio channel and the specific value of the noise. As such, Y k is a random
process. The adaptive equalizer structure shown above is called a transversal filter, and in this
case has N delay elements, N + 1 taps, and N + 1 tunable complex multipliers, called weights.

The weights of the filter are described by their physical location in the delay line structure, and
have a second subscript, k, to explicitly show they vary with time. These weights are updated
continuously by the adaptive algorithm, either on a sample by sample basis (i.e., whenever k is
incremented by 1) or on a block by block basis (i.e., whenever a specified number of samples have
been clocked into the equalizer).

The adaptive algorithm is controlled by the error signal e k. This error signal is derived by
comparing the output of the equalizer, dk. with some signal which is either an exact scaled replica
of the transmitted signal xk or which represents a known property of the transmitted signal. The
adaptive algorithm uses Ck to minimize a cost function and updates the equalizer weights in a
manner that iteratively reduces the cost function. For example, the least mean squares (LMS)
algorithm searches for the optimum or near-optimum filter weights by performing the following
iterative operation:

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New weights = Previous weights + (constant) x (Previous error) x (Current input vector)

where
Previous error = Previous desired output — Previous actual output

and the constant may be adjusted by the algorithm to control the variation between filter weights
on successive iterations. This process is repeated rapidly in a programming loop while the
equalizer attempts to converge, and many techniques (such as gradient or steepest decent
algorithms) may be used to minimize the error. Upon reaching convergence, the adaptive
algorithm freezes the filter weights until the error signal exceeds an acceptable level or until a new
training sequence is sent.

Based on classical equalization theory, the most common cost function is the mean square error
(MSE) between the desired signal and the output of the equalizer. The MSE is denoted by E[e(k)
e*(k)]. and when a replica of the transmitted signal is required at the output of the equalizer (i.e.,
when dk is set equal to Xk and is known a priori), a known training sequence must be
periodically transmitted. By detecting the training sequence, the adaptive algorithm in the
receiver is able to compute and minimize the cost function by driving the tap weights untilthe
next training sequence is sent.

A more recent class of adaptive algorithms are able to exploit characteristics of the transmitted
signal and do not require training sequences. These modern algorithms are able to acquire
equalization through property restoral techniques of the transmitted signal, and are called blind
algorithms because they provide equalizer convergence without burdening the transmitter with
training overhead. These techniques include algorithms such as the constant modulus algorithm
(CMA) and the spectral coherence restoral algorithm (SCORE). is used forconstant envelope
modulation, and forces the equalizer weights to maintain a constant envelope on the received
signal, whereas SCORE exploits spectral redundancy or cyclostationarity in the transmitted signal.

To study the adaptive equalizer, it is helpful to use vector and matrix algebra.

Define the input signal to the equalizer as a vector Yk where

𝑌𝑘 = [𝑌𝑘 𝑌𝑘−1 𝑌𝑘−2 … … . . 𝑌𝑘−𝑁]𝑇

It should be clear that the output of the adaptive equalizer is a scalar given by
𝑁

𝑑𝑘̇ = ∑ W𝑛𝑘 𝑦𝑘−𝑛

𝑛=0

A weight vector can be written as

𝑤𝑘 = [𝑤0𝑘 𝑤1𝑘 𝑤2𝑘 … … . . 𝑤𝑁𝑘]𝑇

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May be written in vector notation as

𝑦𝑘 𝑇 𝑤𝑘 = 𝑤𝑘 𝑇 𝑦𝑘

It follows that when the desired equalizer output is known (i.e., dk = xk), the error signal ek is
given by
𝑒𝑘 = 𝑑𝑘 − 𝑑𝑘= 𝑥𝑘 − 𝑑 𝑘

and from

𝑒𝑘 = 𝑥𝑘 − 𝑦𝑘𝑇𝑤𝑘 = 𝑥𝑘 − W𝑘𝑇𝑦𝑘

To compute the mean square error |ek|2 at time instant k ,then equation becomes
|e𝑘|2 = 𝑥𝑘2 + W𝑘𝑇𝑦𝑘 𝑦𝑘𝑇𝑤𝑘 − 2𝑥𝑘 𝑦𝑘𝑇𝑤𝑘

Taking the expected value of |ek|2 over k (which in practice amounts to computing a time
average) yields
𝐸 [|e𝑘|2] = 𝐸[𝑥𝑘2] + W𝑘𝑇 𝐸 [𝑦𝑘 𝑦𝑘𝑇] 𝑤𝑘 − 2𝐸 [𝑥𝑘 𝑦𝑘𝑇] 𝑤𝑘

Notice that the filter weights Wk are not included in the time average since, for convenience, it is
assumed that they have converged to the optimum value and are not varying with time.

Equation would be trivial to simplify if Xk and Yk were independent. However, this is not true in
general since the input vector should be correlated with the desired output of the equalizer
(otherwise, the equalizer would have a very difficult time extracting the desired signal.). Instead,
the cross correlation vector p between the desired response and the input signal is defined as
𝑃 = 𝐸 [X𝑘 𝑌𝑘] = 𝐸[𝑥𝑘𝑦𝑘 𝑥𝑘𝑦𝑘−1 𝑥𝑘𝑦𝑘−2 … … . 𝑥 𝑘𝑦𝑘−𝑁]𝑇

and the input correlation matrix is defined as the (N + 1) x (N + 1) square matrix R where
𝑌𝑘2 𝑦𝑘𝑦𝑘−1 ⋯ 𝑦𝑘𝑦𝑘−𝑁
*
𝑅 = 𝐸 [𝑌𝑘 𝑌𝑘 ] = 𝑦𝑘−1𝑦𝑘 𝑦𝑘−1𝑦𝑘−1 ⋯ 𝑦𝑘−1𝑦𝑘−𝑁
⁝ 𝑦𝑘−𝑁 𝑦𝑘 ⁝ 𝑦𝑘−𝑁 𝑦𝑘−1 ⋯ ⁝ 𝑦𝑘−𝑁𝑦𝑘−𝑁

The matrix R is sometimes called the input covariance matrix. The major diagonal of R contains
the mean square values of each input sample, and the cross terms specify the autocorrelation terms
resulting from delayed samples of the input signal.

If Xk and Yk are stationary, then the elements in R and p are second order statistics which do not
vary with time. Using equations

𝑀𝑒𝑎𝑛 𝑆𝑞𝑢𝑎𝑟𝑒 𝐸𝑟𝑟𝑜𝑟 Ξ s = 𝐸 [𝑥𝑘2] + W𝑇𝑅𝑤 − 2𝑃𝑇W

By minimizing equation in terms of the weight vector Wk, it becomes possible to adaptively tune
the equalizer to provide a flat spectral response (minimal ISI) in the received signal. This is due to
the fact that when the input signal Yk and the desired response Xk are the mean square error (MSE)
is quadratic on Wk, and minimizing the MSE leads to optimal solutions for Wk.

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TYPES OF EQUALIZERS

Equalizer can be classified as

1. Linear equalizer
2. Non linear equalizer

Equalizer

Linear Equalizer Non Linear Equalizer

Decision ML ML
Feedback Symbol- Sequence -
equalizer equalizer equalizer

Classification of Equalizers
Linear Equalizer: Linear equalizer is usually a tapped-delay-line filter with coefficients that are
adapted to the channel state. A linear equalizer can be implemented as an FIR filter. In such an
equalizer, the current and past values of the received signal are linearly weighted by the filter
coefficient and summed to produce the output.

Advantage: This type of equalizer is the simplest type available.

Disadvantage: The linear equalizer places too much gain near the spectral null, thereby
enhancing the noise present in those frequencies.

Nonlinear Equalization: A Nonlinear equalizers are used in applications where the channel
distortion is too severe for a linear equalizer to handle.

Three very effective nonlinear methods offer improvements over linear equalization techniques.
These are
1. Decision Feedback Equalization (DFE)
2. Maximum Likelihood Symbol Detection
3. Maximum Likelihood Sequence Estimation (MLSE)

Decision feedback equalizer: The ISI created by past symbols can be computed (and subtracted)
from the received signal.
Maximum Likelihood Sequence Estimation: The optimum way of detection in a delay-dispersive
channel is the Maximum Likelihood Sequence Estimation.

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LINEAR EQUALIZERS

Linear equalizers are simple linear filter structures. Linear equalizers try to invert the channel in
the sense that the product of the transfer functions of channel and equalizer fulfills a certain
criterion. This criterion can either be
1. flat transfer function of the channel–filter concatenation, or
2. minimizing the mean-squared error at the filter output.

The basic structure of a linear equalizer is sketched in Figure.

𝑛(𝑡) 𝑐𝑖
𝑠(𝑡)
𝜓𝑖 Noise 𝑢𝑖
𝑐𝑖
𝑔(𝑡) ℎ(𝑡) Whitening
Filter
𝜀𝑖
𝑐𝑖̂
𝑐𝑖
𝑢𝑖
𝑐𝑖
𝐹(𝑧) 𝐸(𝑧) 𝜀𝑖
𝑐𝑖̂
𝑛𝑖
A transmit sequence {𝑐i} is sent over a dispersive, noisy channel, so that the sequence {𝑢i} is
available at the equalizer input. We now need to find the coefficients of a Finite Impulse Response
(FIR) filter (transversal filter, with 2K + 1 taps.

This filter should convert sequence {𝑢i} into sequence {𝑐î } that should be as close as possible to
the sequence {𝑐i}. Defining the deviation si as
 i  ci  ĉi
we aim to find a filter so that
i  0 for N0  0

 
which gives the ZF equalizer, or that
E  i 2  min for N having a finite value
0

which gives the Minimum Mean Square Error (MMSE) equalizer.

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Transversal Linear Equalizer (Explain the theory of adaptive equalizer)

Lattice Equalizer
Two main advantages of the lattice equalizer is its numerical stability and faster
convergence. Also, the unique structure of the lattice filter allows the dynamic
assignment of the most effective length of the lattice equalizer. Hence, if the channel is
not very time dispersive, only a fraction of the stages are used. When the channel
becomes more time dispersive, the length of the equalizer can be increased by the
algorithm without stopping the operation of the equalizer. The structure of a lattice
equalizer, however, is more complicated than a linear transversal equalizer.

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Zero-forcing equalizer
The ZF equalizer can be interpreted in the frequency domain as enforcing a completely flat
(constant) transfer function of the combination of channel and equalizer by choosing the equalizer
transfer function as
1
E(z) 
F (Z )
In the time domain, this can be interpreted as minimizing the maximum ISI (peak distortion
criterion). Zero-forcing equalizer is based on the peak distortion criterion, where the equalizer
forces the ISI to zero.
The ZF equalizer is optimum for elimination of ISI. However, channels also add noise, which is
amplified by the equalizer. At frequencies where the transfer function of the channel attains small
values, the equalizer has a strong amplification, and thus also amplifies the noise.

The Minimum Mean Square Error Equalizer

The ultimate goal of an equalizer is minimization of the bit error probability and not ISI. Noise
enhancement makes the ZF equalizer not suitable for this purpose. A better criterion is
minimization of the Mean Square Error (MSE) between the transmit signal and the output of the
equalizer.

We are thus searching for a filter that minimizes:

   
MSE  E  i 2  E  i i
This can be achieved with a filter whose coefficients eopt are given by
eopt  R1p
where
R = 𝐸{u*u𝑇} is the correlation matrix of the received signal
i i
p = 𝐸{u*i𝑐i} is the crosscorrelation between the received signal and the transmit signal.

Considering the frequency domain, cascade of the noise-whitening filter with the equalizer 𝐸(𝑧)
has the transfer function:
1
E(z) 
N0
(Z ) 
S2

The MSE is then

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The noise power of an MMSE equalizer is smaller than that of a ZF equalizer.

Adaptation Algorithms for Mean Square Error Equalizers

Least Mean Square Algorithm
The LMS algorithm; also known as the stochastic gradient method, consists of the following
steps:
1. Initialize the weights with values e0.
2. With this value, compute an approximation for the gradient of the MSE. We are using an
estimate for R and p:

R̂ n  unuTn
p̂ n  un cn

where subscript n indexes the iterations. The gradient is estimated as

̂n  2p̂n  2R̂ n e n

3. We next compute an updated estimate of the weight vector e by adjusting weights in the
direction of the negative gradient:

e n1  e n  ̂n

4. If the stop criterion is fulfilled – e.g., the relative change in weight vector falls below a
predefined threshold – the algorithm has converged. Otherwise, we return to step 2.

It can be shown that the LMS algorithm converges if

2
0   
max
Here 𝜆𝑚𝑎𝑥 is the largest eigenvalue of the correlation matrix R

The Recursive Least Squares Algorithm

In most cases, the LMS algorithm converges very slowly. Furthermore, the use of this algorithm
is justified only when the statistical properties of the received signal fulfill certain conditions.

The general Least Squares (LS) criterion does not require such assumptions. It just analyzes the
N subsequent errors si , and chooses weights such that the sum of the squared errors is minimized.
This general LS problem can be solved by a recursive algorithm as well – known as RLS.

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Comparison of Algorithms

LMS Algorithm RLS Algorithm

The LMS algorithm usually converges The RLS algorithm converges faster, but
too slowly. has a larger residual error.
The LMS algorithm requires fewer The RLS algorithm requires more
(complex) operations than RLS (complex) operations than LMS
algorithm. algorithm.

DECISION FEEDBACK EQUALIZERS

In decision feedback equalization, once a bit is correctly detected, the effect this bit on subsequent
bit is determined. The ISI caused by each bit is then subtracted from these later samples.

The block diagram of a DFE is shown in Figure below.

The DFE consists of a forward filter with transfer function 𝐸(𝑧), which is a conventional linear
equalizer, as well as a feedback filter with transfer function 𝐷(𝑧). As soon as the RX has decided
on a received symbol, its impact on all future samples (postcursor ISI) can be computed, and
(via the feedback) subtracted from the received signal.

The ISI is computed based on the signal after the hard decision. This eliminates additive noise
from the feedback signal. Therefore, a DFE results in a smaller error probability than a linear
equalizer.

One possible source of problems is error propagation. If the RX decides incorrectly for one bit,
then the computed postcursor ISI is also erroneous, so that later signal samples arriving at the
decision device are even more afflicted by ISI than the unequalized samples.

The basic idea behind decision feedback equalization is that once an information symbol has been
detected and decided upon, the ISI that it induces on future symbols can be estimated and
subtracted out before detection of subsequent symbols. The DFE can be realized in either the direct
transversal form or as a lattice filter.

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It consists of a feedforward filter (FFF) and a feedback filter (FBF). The FBF is driven by decisions
on the output of the detector, and its coefficients can be adjusted to cancel the ISI on the current
symbol from past detected symbols. The equalizer has N1 + N2 + I taps in the feed forward filter
and N3 taps in the feedback filter. The lattice implementation of the DFE is equivalent to a
transversal DFE having a feed forward filter of length N1 and a feedback filter of length N2, where
N1 > N2.

Another form of DFE proposed by Belfiore and Park is called a predictive DFE, and is shown in
below Figure. It also consists of a feed forward filter (FFF) as in the conventional DFE. However,
the feedback filter (FBF) is driven by an input sequence formed by the difference of the output of
the detector and the output of the feed forward filter. Hence, the FBF here is calleda noise predictor
because it predicts the noise and the residual ISI contained in the signal at the FFF output and
subtracts from it the detector output after some feedback delay. The predictive DFE performs as
well as the conventional DFE as the limit in the number of taps in the FFF and the FBF approach
infinity. The FEF in the predictive DFE can also be realized as a lattice structure. The RLS lattice
algorithm can be used in this case to yield fast convergence.

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MMSE Decision Feedback Equalizer
The goal of the MMSE DFE is again minimization of the MSE. We now aim to minimize the
sum of noise and (average) precursor ISI.

The coefficients of the feedforward filter can be computed from the following equation:




where 𝐾ƒƒ is the number of taps in the feedforward filter.

The coefficients of the feedback filter are then

0

dn    em fnm
mK fb
for n = 1 toK fb

where 𝐾ƒ𝑏 is the number of taps in the feedback filter.

The MSE at the equalizer output is   

 
  /Ts  0  

Zero-Forcing Decision Feedback Equalizer

In ZF DFE, the noise-whitening filter eliminates all precursor ISI, such that the resulting effective
channel is purely causal. Postcursor ISI is subtracted by the feedback branch. The effective noise
power at the decision device is

This equation demonstrates that noise power is larger than it is in the unequalized case, but smaller
than that for the linear ZF equalizer.

Comparison of Equalizer Structures

When selecting an equalizer for a practical system, we have to consider the following criteria:
1. Minimization of the BER: here MLSE is superior to all other structures. DFEs, though worse than
MLSE estimators, are better than linear equalizers.
2. Computational effort: The effort for linear equalizers and DFEs is not significantly different.
Depending on the adaptation algorithm, the number of operations increases linearly, quadratically,
or cubically with equalizer length (number of weights). For MLSE, the computation effort
increases exponentially with length of the impulse response of the channel.
3. Sensitivity to channel misestimation: Due to the error propagation effect, DFE equalizers are more
sensitive to channel estimation errors than linear equalizers. Also, ZF equalizers are more sensitive
than MMSE equalizers.
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Maximum Likelihood Sequence Estimation (MLSE) Equalizer

The MSE-based linear equalizers described previously are optimum with respect to the criterion
of minimum probability of symbol error when the channel does not introduce any amplitude
distortion. Yet this is precisely the condition in which an equalizer is needed for a mobile
communications link. This limitation on MSE-based equalizers led researchers to investigate
optimum or nearly optimum nonlinear structures. These equalizers use various forms of the
classical maximum likelihood receiver structure. Using a channel impulse response simulator
within the algorithm, the MLSE tests all possible data sequences (rather than decoding each
received symbol by itself), and chooses the data sequence with the maximum probability as the
output. An MLSE usually has a large computational requirement, especially when the delay spread
of the channel is large.

Using the MLSE as an equalizer was first proposed by Forney in which he set up a basic MLSE
estimator structure and implemented it with the Viterbi algorithm. This algorithm, was recognized
to be a maximum likelihood sequence estimator (MLSE) of the state sequences of a finite state
Markov process observed in memoryless noise. It has recently been implemented successfully for
equalizers in mobile radio channels.

The MLSE can be viewed as a problem in estimating the state of a discrete time finite state
machine, which in this case happens to be the radio channel with coefficients fk and with a channel
state which at any instant of time is estimated by the receiver based on the L most recent input
samples. Thus the channel has states, where M is the size of the symbol alphabet of the modulation.
That is, an trellis is used by the receiver to model the channel over time. The Viterbi algorithm
then tracks the state of the channel by the paths through the trellis and gives at stage ka rank
ordering of the ML most probable sequences terminating in the most recent L symbols.

The block diagram of a MLSE receiver based on the DFE is shown in Figure. The MLSE is optimal
in the sense that it minimizes the probability of a sequence error. The MLSE requires knowledge
of the channel characteristics in order to compute the metrics for making decisions. The MLSE
also requires knowledge of the statistical distribution of the noise corrupting the signal. Thus, the
probability distribution of the noise determines the form of the metric for optimum demodulation
of the received signal. Notice that the matched filter operates on the continuous time signal,
whereas the MLSE and channel estimator rely on discretized (nonlinear) samples.

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5CS5-11: Wireless Communication Department of ECE

Algorithms for Adaptive Equalization

The performance of an algorithm is determined by various factors which include:

Rate of convergence — This is defined as the number of iterations required for the algorithm, in
response to stationary inputs, to converge close enough to the optimum solution. A fast rate of
convergence allows the algorithm to adapt rapidly to a stationary environment of unknown
statistics. Furthermore, it enables the algorithm to track statistical variations when
operating in a nonstationary environment.

Misadjustment For an algorithm of interest, this parameter provides a quantitative measure of the
amount by which the final value of the mean square error, averaged over an ensemble of adaptive
filters, deviates from the optimal minimum mean square error.

Computational complexity — This is the number of operations required to make one complete
iteration of the algorithm.

Numerical properties — When an algorithm is implemented numerically, inaccuracies are

produced due to round-oft' noise and representation errors in the computer. These kinds of errors
influence the stability of the algorithm.

Figure shows a taxonomy of equalizer structures.

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 5CS5-11: Wireless Communication Department of ECE

SPEECH CODING

Speech coding is the process for reducing the bit rate of digital speech representation for
transmission or storage, while maintaining a speech quality that is acceptable for the application.

Speech coding is used to save bandwidth and improve bandwidth efficiency whereas channel
coding is employed to improve signal quality and reduce bit-error-rate (BER).

Speech coders are classified as

1. Waveform coders
2. Source coders
3. Hybrid coders

Speech coders

Waveform coders Source coders Hybrid coders

Error Probability in Fading Channels with Diversity Reception

Error Probability in Flat-Fading Channels

Classical Computation Method

The error probability of diversity systems by averaging the conditional error probability
(conditioned on a certain SNR) over the distribution of the SNR:

Symbol Error Rate:

As an example, let us compute the performance of BPSK with Nr diversity branches with MRC.
The SER of BPSK in AWGN is

Let us apply this principle to the case of MRC.

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5CS5-11: Wireless Communication Department of ECE

where b is defined as

For large values of γ , this can be approximated as

From this, we can see that (with Nr diversity antennas) the BER decreases with the Nr-th power
of the SNR.

Computation via the Moment-Generating Function

In the previous section, we averaged the BER over the distribution of SNRs, using the ―classical‖
representation of the Q-function. There is an alternative definition of the Q-function, which can
easily be combined with the moment-generating function Mγ (s) of the SNR. Let us start by writing
the SER conditioned on a given SNR in the form

Averaging over the SNRs in different branches then becomes

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5CS5-11: Wireless Communication Department of ECE

we can write the error probability for BPSK in Rayleigh fading as

Symbol Error Rate in Frequency-Selective Fading Channels

We now determine the SER in channels that suffer from time dispersion and frequency
dispersion.
We assume here FSK with differential phase detection. The analysis uses the correlation
coefficient ρXY between signals at two sampling times that was discussed in Chapter 12.
For binary FSK with selection diversity:

This formulation shows that MRC improves the SER by a factor Nr! compared with selection
diversity.

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