UNIT-III

EQUALIZATION TECHNIQUES
In telecommunication, equalization is the reversal of distortion incurred by a signal transmitted
through a channel. Equalizers are used to render the frequency response—for instance of a
telephone line—flat from end-to-end. When a channel has been equalized the frequency
domain attributes of the signal at the input are faithfully reproduced at the output.
Telephones, DSL lines and television cables use equalizers to prepare data signals for
transmission.

Equalizers are critical to the successful operation of electronic systems such as analog broadcast
television. In this application the actual waveform of the transmitted signal must be preserved,
not just its frequency content. Equalizing filters must cancel out any group delay and phase
delay between different frequency components. Modern digital telephone systems have less
trouble in the voice frequency range as only the local line to the subscriber now remains in
analog format, but DSL circuits operating in the MHz range on those same wires may suffer
severe attenuation distortion, which is dealt with by automatic equalization or by abandoning the
worst frequencies. Picture phone circuits also had equalizers.

In digital communications, the equalizer's purpose is to reduce inter symbol interference to allow
recovery of the transmit symbols. It may be a simple linear filter or a complex algorithm.

Digital equalizer types:

Linear equalizer: processes the incoming signal with a linear filter.

MMSE equalizer: designs the filter to minimize E[|e|2], where e is the error signal, which is the
filter output minus the transmitted signal.
Zero forcing equalizer: approximates the inverse of the channel with a linear filter.

Decision feedback equalizer: augments a linear equalizer by adding a filtered version of previous
symbol estimates to the original filter output.

Blind equalizer: estimates the transmitted signal without knowledge of the channel statistics,
using only knowledge of the transmitted signal's statistics.

Adaptive equalizer: is typically a linear equalizer or a DFE. It updates the equalizer parameters
(such as the filter coefficients) as it processes the data. Typically, it uses the MSE cost function;
it assumes that it makes the correct symbol decisions, and uses its estimate of the symbols to
compute e, which is defined above.

Viterbi equalizer: Finds the maximum likelihood (ML) optimal solution to the equalization
problem. Its goal is to minimize the probability of making an error over the entire sequence.
BCJR equalizer: uses the BCJR algorithm (also called the Forward-backward algorithm) to find
the maximum a posteriori (MAP) solution. Its goal is to minimize the probability that a given bit
was incorrectly estimated.

Turbo equalizer: applies turbo decoding while treating the channel as a convolutional code.

ISI has been identified as one of the major obstacles to high speed data transmission over mobile
radio channels. If the modulation bandwidth exceeds the coherence bandwidth of the radio
channel (i.e., frequency selective fading), modulation pulses are spread in time, causing ISI. An
equalizer at the front end of a receiver compensates for the average range of expected channel
amplitude and delay characteristics. As the mobile fading channels are random and time varying,
equalizers must track the time-varying characteristics of the mobile channel and therefore should
be time varying or adaptive. An adaptive equalizer has two phases of operation: training and
tracking.

These are as follows.

Training Mode: Initially a known, fixed length training sequence is sent by the transmitter so
that the receiver equalizer may average to a proper setting. Training sequence is typically a
pseudo-random binary signal or axed, of prescribed bit pattern. Training sequence is designed to
permit an equalizer at the receiver to acquire the proper filter coefficient in the worst possible
channel condition. An adaptive filter at the receiver thus uses a recursive algorithm to evaluate
130 the channel and estimate filter coefficients to compensate for the channel.

Tracking Mode: When the training sequence is finished the filter coefficients are near optimal.
Immediately following the training sequence, user data is sent. When the data of the users are
received, the adaptive algorithms of the equalizer track the changing channel. As a result, the
adaptive equalizer continuously changes the filter characteristics over time.

A Mathematical Framework:
The signal received by the equalizer is given by

x(t) = d(t) * h (t) + nb (t) …………….(3.1)

Where d(t) is the transmitted signal, h(t) is the combined impulse response of the transmitter,
channel and the RF/IF section of the receiver and nb (t) denotes the baseband noise.

If the impulse response of the equalizer is heq (t), the output of the equalizer is

^y (t) = d (t) * h (t) * heq (t) + nb (t) * heq (t) = d (t) * g (t) + nb (t) *heq (t) ………….(3.2)

However, the desired output of the equalizer is d(t) which is the original source data.
Assuming nb (t)=0, we can write y(t) = d(t), which in turn stems the following equation:

g (t) = h (t) * heq (t) = δ (t)…………….(3.3)

The main goal of any equalization process is to satisfy this equation optimally. In frequency
domain it can be written as

Heq (f)H (f) = 1 ………………(3.4)

Which 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 at,
composite received frequency response and linear phase response. For a time varying channel,
the equalizer is designed to track the channel variations so that the above equation is
approximately satisfied.

Zero Forcing Equalization

In a zero forcing equalizer, the equalizer coefficients cnare chosen to force the samples of the
combined channel and equalizer impulse response to zero. When each of the delay elements
provides a time delay equal to the symbol duration T, the frequency response Heq(f) of the
equalizer is periodic with a period equal to the symbol rate 1/T. The combined response of the
channel with the equalizer must satisfy Nyquist's criterion

Hch(f)Heq(f) = 1ǀfjǀ< 1/2T …………….(3.5)

Where Hch (f) is the folded frequency response of the channel. Thus, an infinite length zero-
forcing ISI equalizer is simply an inverse filter which inverts the folded frequency response of
the channel.

A Generic Adaptive Equalizer

The basic structure of an adaptive filter is shown in This filter is called the transversal filter, and
in this case has N delay elements, N+1 taps and N+1 tunable complex multipliers, called
weights. These weights are updated continuously by an adaptive algorithm. In the figure the
subscript k represents discrete time index. The adaptive algorithm is controlled by the error
signal ek. The error signal is derived by comparing the output of the equalizer, with some signal
dk which is replica of transmitted signal. The adaptive algorithm uses ek to minimize the cost
function and uses the equalizer weights in such a manner that it minimizes the cost function
iteratively.

Let us denote the received sequence vector at the receiver and the input to the equalizer as

xk = [xk; xk-1 ; :::::; xk-N]T; …………(3.6)

and the tap coefficient vector as

wk= [w0k;w1k; :::::;Wnk ]T :………….(3.7)

Now, the output sequence of the equalizer yk is the inner product of xk and wk, i.e.,

yk = xk;wk = xTkwk = wTkxk ……………(3.8)

The error signal is defined as

ek = dk-yk = dk-xTkwk:………..(3.9)

Assuming dk and xk to be jointly stationary, the Mean Square Error (MSE) is given as

MSE = E[e2k]

= E[(dk-yk)2]

= E[(dk-xTkwk)2]

= E[d2k] + wTk E[xkxTk]wk– 2E[dkxTk]wk ………………(3.10)

A generic adaptive equalizer

Where wk is assumed to be an array of optimum values and therefore it has been taken out of the
E() operator. The MSE then can be expressed as

MSE = ξ = σ2k + WTk Rwk- 2pTwk

Where the signal variance σ 2d = E[d2k] and the cross correlation vector p between the desired
response and the input signal is defined as

p = E [dkxk] = E [dkxk dkxk-1dkxk-2…………..dkxk-N]

The input correlation matrix R is defined as an (N + 1) x (N + 1) square matrix,Where

Clearly, MSE is a function of wk. On equating ϑε / ϑωkto 0, we get the condition for minimum
MSE (MMSE) which is known as Wiener solution:
wk= R-1p
Hence, MMSE is given by the equation

MMSE = ξ min = σ2d - pTwk

Choice of Algorithms for Adaptive Equalization

Since an adaptive equalizer compensates for an unknown and time varying channel, it requires a
specific algorithm to update the equalizer coefficients and track the channel variations. Factors
which determine algorithm's performance are:

• Rate of convergence:

Number of iterations required for an algorithm, in response to a stationary inputs, to converge

close enough to optimal solution. A fast rate of convergence allows the algorithm to adapt
rapidly to a stationary environment of unknown statistics.

• Mis-adjustment:

Provides a quantitative measure of the amount by which the final value of mean square error,
averaged over an ensemble of adaptive filters, deviates from an optimal mean square error
 • Computational complexity:
Number of operations required to make one completeiteration of the algorithm.

• Numerical properties
Inaccuracies like round-off noise and representation errors in the computer, which influence the
stability of the algorithm.

Three classic equalizer algorithms are primitive for most of today's wireless standards. These
include the Zero Forcing Algorithm (ZF), the Least Mean Square Algorithm (LMS), and the
Recursive Least Square Algorithm (RLS). Below, we discuss a few of the adaptive algorithms.

Least Mean Square (LMS) Algorithm

LMS algorithm is the simplest algorithm based on minimization of the MSE between the desired
equalizer output and the actual equalizer output, as discussed earlier.Here the system error, the
MSE and the optimal Wiener solution remain the same as given the adaptive equalization
framework. In practice, the minimization of the MSE is carried out recursively, and may be
performed by use of the stochastic gradient algorithm. It is the simplest equalization algorithm
and requires only 2N+1 operation per iteration. The filter weights are updated by the update
equation. Letting the variable n denote the sequence of iteration, LMS is computed iteratively by

wk(n + 1) = wk(n) + µek(n) x(n - k)

Where the subscript k denotes the kth delay stage in the equalizer and µ is the step size which
controls the convergence rate and stability of the algorithm. The LMS equalizer maximizes the
signal to distortion ratio at its output within the constraints of the equalizer filter length. If an
input signal has a time dispersion characteristics that is greater than the propagation delay
through the equalizer, then the equalizer will be unable to reduce distortion. The convergence
rate of the LMS algorithm is slow due to the fact that there is only one parameter, the step size,
that controls the adaptation rate. To prevent the adaptation from becoming unstable, the value off
is chosen from

0 < µ< 2/ ( i = 1ƩNλi )

,
Where λi is the ith eigen value of the covariance matrix R.

Linear Equalizers

As mentioned in section 6.5, a linear equalizer can be implemented as an FIR filter, otherwise
known as the transversal filter. This type of equalizer is the simplest type available. 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, as shown in. If the delays and the tap gains are
analog, the continuous output of the equalizer is sampled at the symbol rate and the
samples are applied to the decision device. The implementation is, however, usually carried out
in the digital domain where the samples of the received signal are stored in a shift register. The
output of this transversal filter before decision making (threshold detection) is

Where Cn* represents the complex filter coefficients or tap weights, ilk is the output at time index
k, y1 is the input received signal at time to + iT, t 0 is the equalizer starting time, and N = N1
+N2 + I is the number of taps. The values N1 and N2 denote the number of taps used in the
forward and reverse portions of the equalizer, respectively. The minimum mean squared error E [
ǀe(n)2ǀ ] that a linear transversal equalizer can achieve is

where f ( e jωT)is the frequency response of the channel, and N0 is the noise spectral density.
The linear equalizer can also be implemented as a lattice filter, whose structure is shown in
Figure 6.7. The input signal Yk is transformed into a set of N intermediate forward and backward
error signals, fjk) and b4(k) respectively, which are used as inputs to the tap multipliers and are
used to calculate the updated coefficients. Each stage of the lattice is then characterized by the
following recursive equations.
Where (k) is the reflection coefficient for the it th stage of the lattice. The backward error
signals, are then used as inputs to the tap weights, and the output of the equalizer is given by

Decision Feedback Equalization (OFE)

The basic idea behind decision feedback equalization is that once an information symbol has
been detected and decided upon, the 1ST that it induces on future symbols can be estimated and
subtracted out before detection of subsequent symbols [Pro891. The DFE can be realized in
either the direct transversal form or as a lattice filter. The direct form is shown in Figure 6.8. It
consists of a feed forward 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, and its output can be expressed as:

where c, and y, are tap gains and the inputs, respectively, to the forward filter, F1 are tap gains
for the feedback filter, and d1(i ck) is the previous decision made on the detected signal. That is,
once dk is obtained using above equation, dk is decided from it. Then, dk along with previous
decisions dk- 4k – are fed back into the equalizer, and + is obtained using equation (6.26). The
minimum mean squared error a DFE can achieve is

If there are nulls in F (c) a DFE has significantly smaller minimum MSE than an LTE.
Therefore, an LTE is well behaved when the channel spectrum is comparatively flat, but if the
channel is severely distorted or exhibits nulls in the spectrum, the performance of an LTE
deteriorates and the mean squared error of a DFE is much better than a LTE. Also, an LTE has
difficulty equalizing a non-minimum phase channel, where the strongest energy arrives after the
first arriving signal component. Thus, a DFE is more appropriate for severely distorted wireless
channels
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 Figure 6.9. 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 called a noise predictor because it
predicts the noise and the residual 151 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

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 [For78] in
which he set up a basic MLSE estimator structure and implemented it with the Viterbi algorithm.
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 tk 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 k a rank ordering of the M' 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 6.10.
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.

DIVERSITY TECHNIQUES
INTRODUCTION:

Another popular technique other than equalization to overcome the effects of multipath channel
is diversity. Diversity is primarily used to counter act fading effects and we have seen fading
causes a big degradation in the performance of any modulation techniques. What is diversity? It
is a technique used to compensate for the fading channel impairments. The different kinds of
diversity techniques. What are these? The popular ones are antenna diversity, frequency
diversity, time, polarization, angle and code diversity. Of this antenna diversity is one of the
more popular ones where in we use multiple antennas. Now this multiple antennas can be either
at the transmitter or at the receiver or both. Frequency diversity requires you to use frequency
signals which are separated by the coherence bandwidth of the channel.

Time diversity we must separate the transmission more than the coherence time of the channel.
Polarization diversity, the horizontal and the vertical polarized signals fade differently and
independently, angle diversity and used different codes so we can have code diversity.

So the popular Antenna diversity is implemented by using two or more receiving antennas and
as I mentioned before you can also have them at the transmitter. Now while equalization is used
to counter the effects of ISI or inter symbol interference, diversity is usually employed to reduce
the depth and duration of it’s that is the fundamental difference. So the depth of fade and the
duration of fades experienced by a receiver in a flat fading scenario can be overcome by using
diversity. These techniques can be employed both at base station and mobile receivers, any of the
diversity techniques can be deployed either at the base station or at the mobile receivers.

Spatial diversity is the most widely used diversity technique; you can have multiple antennas at
the base station because having multiple antennas in your handset is a little inconvenient or you
can even have antennas on different base stations. So you really get a good spatial diversity. The
key point in spatial diversity is the separation of the antenna elements depending upon how
clattered is the multipath environment, your antenna separations may have to be increased or
decreased. Let us talk about spatial diversity.

In this technique multiple antennas are strategically spaced and connected to common receiving
system; while one antenna sees a signal null, one of the other antennas may see a signal peak
provided the signals are not correlated. In this case the receiver is able to select the antenna with
the best signal at any time or it can do some kind of an intelligent combining. This CDMA or
code division multiple access systems use rake receivers which provide improvement through
time diversity.

Unlike equalization, diversity requires no training overhead as a transmitter doesn’t require one.
Today we’ll see that equalization has two modes, the training mode and the tracking mode. The
training mode will require sending a known signal and it is used to set the weights of the
equalizer whereas diversity doesn’t have that issue. What else these diversity techniques do? It
provides significant link improvement with little added cost, it also exploits random nature of
wave propagation by finding independent and hence uncorrelated signal paths for
communication that’s how diversity works. It is a very simple concept wherein one path
undergoes a deep fade and another independent path may have a strong signal component that’s
it.

• Random nature of radio propagation leads to

1. Multipath propagation
2. Independent fading of each Multipath component
3. If one radio path undergoes a deep fade, another independent path may have a strong signal.
Diversity exploits the random nature of radio propagation by finding independent signal paths
for communication, so as to boost the instantaneous SNR at the receiver.

Diversity is a powerful communication receiver technique that provides wireless link

improvement at relatively low cost.
l. Requires no training
2. In virtually all applications, diversity decisions are made by the receiver, and are unknown to
the transmitter.

• Two types of diversity

1. Microscopic diversity small scale fading

2. Macroscopic diversity large scale fading

Microscopic diversity
l. Small-scale fades: deep and rapid amplitude fluctuations over distances of just a few
wavelengths.
• caused by multiple reflections from the surroundings in the vicinity of the mobile.
• results in a Rayleigh fading distribution of signal strength over small distances.
2. Microscopic diversity techniques can exploit the rapidly changing signal.

3. By selecting the best signal at all times, a receiver can mitigate small-scale fading effects
Called antenna diversity or space diversity.
Samples: Rake receiver, MIMO transmission

Macroscopic diversity
l. Large-scale fading: caused by shadowing due to variations in both the terrain profile and the
nature of the surroundings.
2. In deeply shadowed conditions, the received signal strength at a mobile can drop well below
that of free space.
3. log-normally distributed with a standard deviation of about 10 dB in urban environments.
4. By selecting a base station which is not shadowed when others are, the mobile can improve
substantially the average ratio on the forward link.
It is the mobile that takes advantage of large separations between the serving base stations.
5. Macroscopic diversity is also useful at the base station receiver. By using base station
antennas that are sufficiently separated in space, the base station is able to improve the reverse
link by selecting the antenna with the strongest signal from the mobile.
6. Used to combat slow fading (shadowing).
7. Samples: Base-station handoff in cellular networks

II. Strategies used in diversity techniques

1. Selection diversity
2. Maximal ratio combining diversity
3. Equal-gain combining diversity
4. Hybrid schemes
 • Practical considerations: effectiveness, complexity, cost, and etc.

1. Selection diversity

Selection diversity offers an average improvement in the link margin without requiring
additional transmitter power or sophisticated receiver circuitry. The diversity improvement can
be directly related to the average bit error rate for various modulations. Selection diversity is
easy to implement because all that is needed is a side monitoring station and an antenna switch at
the receiver. However, it is not an optimal diversity technique because it does not use all of the
possible branches simultaneously.

The SNR out of the diversity combiner:

Further assume that each branch has the same average SNR given by

_
Where  2 =1 is assumed.

If each branch has an instantaneous SNR = γi, then the pdf of is γi

_
where  is the mean SNR of each branch.

The probability that a single branch has SNR less than some threshold γi is
Now, the probability that all M independent diversity branches receive signals which are
simultaneously less than some specific SNR threshold is

This is the probability of all branches failing to achieve SNR= γi. If a single branch achieves
SNR > γi, then the probability that SNR > ϒ for one or more branches is given by

This is the probability of exceeding a threshold when selection diversity is used.

To determine the average signal-to-noise ratio of the received signal when diversity is used,
find the pdf of γ(the instantaneous SNR when M branches are used). Thus we compute the
derivation of CDF PM(γ),

_
Then, we can compute the average SNR,  ,

Where, x = γ / Ƭ

The above equation can be evaluated to yield the average SNR improvement offered by selection
diversity.

2. Maximal ratio combining

Maximal ratio combining uses each of the M branches in a co-phased and weighted manner such
that the highest achievable SNR is available at the receiver at all times. The signals from all of
the M branches are weighted and then summed. The individual signals must be co-phased before
being summed requires an individual receiver and phasing circuit for each antenna element.
Output SNR equal to the sum of the individual SNRs.

Advantage:
1. produces an output with an acceptable SNR even when none of the individual signals are
themselves acceptable.
2. Gives the best statistical reduction of fading of any known linear diversity combiner.
The SNR out of the diversity combiner:
1. If each branch has gain, then the resulting signal envelope applied to the detector is

assuming that each branch has the same average noise power N, the total noise power NT applied
to the detector is simply the weighted sum of the noise in each branch. Thus

which results in an SNR applied to the detector, given by

Using Chebychev's inequality, is maximized when Gi = ri / N, which leads to

Conclusion:

The SNR out of the diversity combiner is simply the sum of the SNRs in each branch.
Gi = ri / N

2) The pdf of ‘A’ according to Chapter 3, is a Chi-square distribution of 2 Gaussian random

variables. Thus, the pdf is
The CDF of ϒM is
According to the abovementioned pdf, the probability that is less than some SNR threshold is

The average SNR out of the diversity combiner: can be calculated by using the pdf of (Eq.
(7.68)). But the direct way is to calculate it from Eq. (7-66).

That is to say, the average SNR, , is simply the sum of the individual from each branch.

The control algorithms for setting the gains and phases for maximal ratio combining receivers
are similar to those required in equalizers and RAKE receivers. Maximal ratio combining can be
applied to virtually any diversity application, although often at much greater cost and complexity
than other diversity techniques.

Space diversity (also known as antenna diversity), is one of the most popular forms of
diversity used in wireless systems. The signals received from spatially separated antennas on the
mobile would have essentially uncorrelated envelopes for antenna separations of one half
wavelength or more. Space diversity can be used at either the mobile or base station, or both.
Since the important scatterers are generally on the ground in the vicinity of the mobile,
when base station diversity is used, the antennas must be spaced considerably far apart to achieve
decorrelation (several tens of wavelengths).

In maximal ratio combining, the voltage signals from each of the M diversity branches are co
phased to provide coherent voltage addition and are individually weighted to provide optimal
SNR.

1) The SNR out of the diversity combiner:

• If each branch has gain , then the resulting signal envelope
applied to the detector is

• Assuming that each branch has the same average noise power N, the total noise power NT
applied to the detector is simply the weighted sum of the noise in each branch. Thus
 • which results in an SNR applied to the detector, γM given by

Using Chebychev's inequality, γM is maximized when Gi = ri / N, which leads to

Conclusion:

The SNR out of the diversity combiner is simply the sum of the SNRs in each branch.

2) The pdf of γM
According to Chapter 3, γM is a Chi-square distribution of 2M Gaussian random variables. Thus,
the pdf for γM is

3) The CDF of γM

According to the abovementioned pdf, The probability that is less than some SNR threshold is

4) The average SNR out of the diversity combiner, γM

γM can be calculated by using the pdf of (Eq. (7.68)). But the direct way is to calculate it from

That is to say, the average SNR, γM, is simply the sum of the individual from each branch.
The control algorithms for setting the gains and phases for maximal ratio combining receivers
are similar to those required in equalizers and RAKE receivers.
Maximal ratio combining can be applied to virtually any diversity application, although often at
much greater cost and complexity than other diversity techniques.
(2) Feedback or Scanning Diversity

1. Very similar to selection diversity

2. The M signals are scanned in a fixed sequence until one is found to be above a predetermined
threshold.
3. This signal is then received until it falls below threshold and the scanning process is again
initiated.
4. The resulting fading statistics are somewhat inferior to those obtained by the other methods.

Advantage: very simple to implement (only one receiver is required).

4) 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 Equal gain combining diversity sets all
weights to unity but the signals from each branch are co-phased. The possibility of producing an
acceptable signal from a number of unacceptable inputs is still retained; The performance is only
marginally inferior to maximal ratio combining and superior to selection diversity.

Alamouti Scheme:
In OFDM reference design, the diversity technique used is called Alamouti block code. It is a
complex space-time diversity technique that can be used in 2×1 MISO mode or in a 2×2 MIMO
mode. The Alamouti block code is the only complex block code that has a data rate of 1 while
achieving maximum diversity gain. Such performance is achieved using the following space-
time block code:
 Figure 2: Alamouti space-time diversity technique

Briefly, two antennas are used, to send two OFDM symbols and their conjugate, in two time
slots, which brings a diversity gain without having to compromise on the data rate. Over the air,
the transmitted symbols will suffer from channel fading and at the receiver, their sum will be
received. Here is the schematic diagram of an Alamouti wireless system in 2×2 MIMO mode:

Figure 3: A 2×2 MIMO wireless system using the Alamouti block code

Since the transmission is done over two periods of time, the decoding will also be done over two
periods of time. At the receiver, the received vector Y can be represented by the following
equation:
 This is for the first time period. For the second time period, the equation is as follows:

where represents the received OFDM symbol at the first time period, for antennas 1 and 2,

respectively, and where represents the received OFDM symbol at the second time period
for antennas 1 and 2, respectively. Both equations can easily be combined and arranged to
produce the following result:

The next step is to find a way to isolate the transmitted symbols, x1 and x2. One way to reduce the
number of unknowns is by using a channel estimator to estimate the channel coefficients. In
Nutaq’s OFDM reference design, channel estimation OFDM symbols are sent with each
transmitted packet to enable estimating those channel coefficients at the receiver. Given the
following matrix:

we can isolate x1 and x2by simply multiplying the matrix Y by the inverse of H. However, since
this matrix is not square, we need to use the Moore-Penrose pseudo-inverse H+ to solve our
equations:

Using this inverse matrix expression, the noisy estimated transmitted symbols can be found using
the following expression:
The last step would be to make a final decision on the transmitted symbols. In Nutaq’s OFDM
reference design, the decision is made based on the minimum squared Euclidian distance
criterion. In the next figure, we can see that the addition of diversity to the system brings a
significant performance gain in terms of BER in simulation.
The Alamouti space-time block coding is a simple MIMO technique that can be used to reduce
the BER of a system, at a specific SNR, without any loss on the data rate. The presented
decoding technique is called hard decision-based zero forcing and is probably the simplest to
implement in hardware. It is a linear decoding technique that has a low complexity. That being
said, further reductions in BER can be achieved by using non-linear decoding techniques, which
are usually better than linear techniques, and also by using soft decision techniques, with specific
QAM constellation decision boundaries.