Wong & Lok: Theory of Digital Communications

4. ISI & Equalization

Chapter 4 Intersymbol Interference and Equalization

The all-pass assumption made in the AWGN (or non-dispersive) channel model is rarely practical. Due to the scarcity of the frequency spectrum, we usually lter the transmitted signal to limit its bandwidth so that efcient sharing of the frequency resource can be achieved. Moreover, many practical channels are bandpass and, in fact, they often respond differently to inputs with different frequency components, i.e., they are dispersive. We have to rene the simple AWGN (or non-dispersive) model to accurately represent this type of practical channels. One such commonly employed renement is the dispersive channel model1:

where is the transmitted signal, with power spectral density by the linear lter

(4.1)

is the impulse response of the channel, and is AWGN

In essence, we model the dispersive characteristic of the channel

The simplest dispersive channel is the bandlimited channel for which the

channel impulse response

is that of an ideal lowpass lter.

This lowpass ltering smears the

transmitted signal in time causing the effect of a symbol to spread to adjacent symbols when a sequence of symbols are transmitted. The resulting interference, intersymbol interference (ISI), degrades the error performance of the communication system. There are two major ways to mitigate the detrimental effect of ISI. The rst method is to design bandlimited transmission pulses which minimize the the
1

For simplicity, all the signals considered in this chapter are real baseband signals. All the developments here can, of

course, be generalized to bandpass signals using either the real bandpass representation or the complex baseband representation.

4.1

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4. ISI & Equalization

effect of ISI. We will describe such a design for the simple case of bandlimited channels. The ISIfree pulses obtained are called the Nyquist pulses. The second method is to lter the received signal to cancel the ISI introduced by the channel impulse response. This approach is generally known as equalization.

4.1 Intersymbol Interference

To understand what ISI is, let us consider the transmission of a sequence of symbols with the basic waveform

. To send the th symbol , we send , where is the symbol interval.

Therefore, the transmitted signal is

(4.2)

Based on the dispersive channel model, the received signal is given by

where

(4.3)

is the received waveform for a symbol. If a single symbol, say the symbol , and then sample the matched lter output at

is transmitted, the optimal demodulator is the one that employs the matched lter, i.e., we can pass the received signal through the matched lter time

to obtain the decision statistic. When a sequence of symbols are transmitted, we can still
to obtain the decision statistic for the symbol
. At

employ this matched lter to perform demodulation. A reasonable strategy is to sample the matched lter output at time of the matched lter is

, the output

(4.4)

where is a zero-mean Gaussian random variable with variance

. The rst term in (4.4)

is the desired signal contribution due to the symbol and the second term contains contributions from the other symbols. These unwanted contributions from other symbols are called intersymbol interference (ISI).

4.2

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4. ISI & Equalization

1.5

0.5

0.5

1.5

10

15

20

25

30

Figure 4.1: Eye diagram of a BPSK signal with no ISI Suppose

were timelimited, i.e., except for except for . Therefore,

. Then it is easy to see that for all , and there is no ISI.

As a result, the demodulation strategy above can be interpreted as matched ltering for each symbol. Unfortunately, a timelimited waveform is never bandlimited. Therefore, for a bandlimited channel,

and, hence, are not timelimited and hence ISI is, in general, present. One common way
to observe and measure (qualitatively) the effect of ISI is to look at the eye diagram of the received signal. The effect of ISI and other noises can be observed on an oscilloscope displaying the output of the matched lter on the vertical input with horizontal sweep rate set at multiples of a display is called an eye diagram.

rectangular pulse and binary signaling is employed. The eye diagrams for the cases where the channel is all-pass (no ISI) and lowpass (ISI present) are shown in Figures 4.1 and 4.2, respectively. The effect of ISI is to cause a reduction in the eye opening by reducing the peak as well as causing ambiguity in the timing information.

. Such For illustration, let us consider the basic waveform is the

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4. ISI & Equalization

1.5

0.5

0.5

1.5

10

15

20

25

30

Figure 4.2: Eye diagram of a BPSK signal with ISI

4.2 Nyquist Pulses

A careful observation on (4.4) reviews that it is possible to have no ISI even if the is bandlimited, i.e., the basic pulse shape and/or the channel is bandlimited. More precisely, letting we can rewrite the decision statistic in (4.4) as:

(4.5)

There is no ISI if the Nyquist condition is satised:

for
for

(4.6)

where is some constant and, without loss of generality, we can set

. The Nyquist condition in

this form is not very helpful in the design of ISI-free pulses. It turns out that it is more illustrative to restate the Nyquist condition in frequency domain. To do so, rst let

4.4

(4.7)

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4. ISI & Equalization

Figure 4.3: Case

Taking Fourier transform,

: non-overlapping spectrum

where is the Fourier transform of . The Nyquist condition in (4.6) is equivalent to the condition or in frequency domain. Now, by employing (4.8), we get

(4.8)

(4.9)

This is the equivalent Nyquist condition in frequency domain. It says that the folded spectrum of has to be at for not having ISI. When the channel is bandlimited to Hz, i.e., the following implications:

for

, the Nyquist condition has

Suppose that the symbol rate is so high that

looks like the one in Figure 4.3.

looks, the Nyquist condition cannot be satised and ISI is inevitable.

. Then, the folded spectrum There are gaps between copies of . No matter how . Then, copies of can just touch is at if and only if

otherwise (4.10)

Suppose that the symbol rate is slower so that their neighbors. The folded spectrum

for for

The corresponding time domain function is the sinc pulse sinc

4.5

(4.11)

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4. ISI & Equalization

Figure 4.4: Case

: overlapping spectrum

We note that the sinc pulse is not timelimited and is not causal. Therefore, it is not physically

above which ISI is unavoidable is known as the Nyquist rate.

realizable. A truncated and delayed version is used as an approximation. The critical rate

Suppose that the symbol rate is even slower so that

their neighbors. The folded spectrum of

. An example is shown in Figure 4.4. Therefore, we can design an ISI-free pulse shape

. Then, copies of overlap with can be at with many different choices

which gives a at folded spectrum. When the symbol rate is below the Nyquist rate, a widely used ISI-free spectrum is the raised-cosine spectrum (Figure 4.5):

where

for

for for

(4.12)

It determines the excess bandwidth2 beyond . The corresponding time domain function (Figure 4.6) is:

is called the roll-off factor.

When

, it reduces to the sinc function. We note that for , decays as while for , (sinc pulse) decays as . Hence, the raised cosine spectrum gives a pulse that is much

(4.13)

Another way to interpret this is that to t

into a channel bandlimited to ,

is, the larger has to be and the slower is the symbol rate.

. Therefore, the larger

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4. ISI & Equalization

Raised Cosine Spectrum 1

0.9

0.8

0.7

0.6 X(f)

0.5

0.4

0.3

0.2 =0

0.1

0.5

0 1

0.8

0.6

0.4

0.2 0 0.2 frequency/symbol rate

0.4

0.6

0.8

Figure 4.5: Raised-cosine spectrum

0.8

0.6

0.4 x(t) 0.2 0 1 0.5 =0 0.2 0.4 3

0 t/T

Figure 4.6: Time domain function of the raised-cosine spectrum

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4. ISI & Equalization

HC ( f )

AWGN
Figure 4.7: Dispersive channel model less sensitive to timing errors than the sinc pulse. Just like all other bandlimited pulses, from the raised cosine spectrum is not timelimited. Therefore, truncation and delay is required for realization. Finally, recall that

is the overall response of the transmitted pulse passing through the ban-

dlimited channel and the receiving lter. Mathematically,

where ,

(4.14)

, and are the Fourier transform of , , and , respectively. Given

that an ISI-free spectrum choose

is chosen, we can employ (4.14) to obtain the simple case of a ban-

dlimited channel, i.e., the channel does not introduce any distortion within its passband, we can simply

also

to be . Then the Fourier transform the transfer function of the matched lter is . For example, if the raised-cosine spectrum is chosen, the resulting ISI-free pulse is

called the square-root raised-cosine pulse. Of course, suitable truncation and delay are required for physical realization.

4.3 Equalization
For many physical channels, such as telephone lines, not only are they bandlimited, but they also introduce distortions in their passbands. Such a channel can be modeled by an LTI lter followed by an AWGN source as shown in Figure 4.7. This is the dispersive channel model we describe before. In general, ISI is often introduced. For a communication system system employing a linear modulation, such as BPSK, through a dispersive channel, the whole system can be described the conceptual model
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4. ISI & Equalization

Ik

HT ( f )

HC ( f )

HR ( f )

Decision every T

Impulses

AWGN

Figure 4.8: Continuous-time communication model over a dispersive channel in Figure 4.8, in which the sequence of information symbols is denoted by and

are the transfer functions of the transmission (pulse-shaping) lter, the dispersive channel

and , ,

and the receiving lter3 , respectively. The Nyquist condition for no ISI developed in Section 4.2 can be easily generalized to the above communication system. Letting condition for no ISI is that the folded spectrum

, is constant for all frequencies, i.e.,

, the

(4.15)

One method to achieve the Nyquist condition is to x the receiving lter to be the matched lter, i.e., set

, and choose the transmission lter so that (4.15) is satised. This

is the Nyquist pulse design method described in Section 4.2. The major disadvantage of this pulse-

in practice. Moreover, we have to know the channel response in advance to construct

the transmission and receiving lters. An alternative method is to x the transmission lter4 and choose the receiving lter

shaping method is that it is in general difcult to construct the appropriate analog lters for

and

to

satisfy the condition in (4.15). As for the previous method, it is also difcult to build the appropriate

to eliminate ISI. However, notice that what we want eventually are the samples at intervals at the receiver. Therefore, we may choose to build a simpler (practical) lter , take samples at intervals , and put a digital lter, called equalizer, at the output to eliminate ISI
analog lter as shown below in Figure 4.9. This approach to remove ISI is usually known as equalization. The main advantage of this approach is that a digital lter is easy to build and is easy to alter for different equalization schemes, as well as to t different channel conditions.
3

If the matched lter demodulation technique is employed, the receiving lter

is chosen to be the complex

conjugate of . 4 Of course, we should choose one that can be easily built.

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4. ISI & Equalization

Ik

HT ( f )

HC ( f )

HR ( f )

Equalizer every T

Decision

Impulses

AWGN

Figure 4.9: Communication system with equalizer

Ik HT ( f ) HC ( f ) HR ( f ) Equalizer every T Decision

Impulses

colored Gaussian noise

Figure 4.10: Equivalent communication system with colored Gaussian noise

4.3.1 Equivalent discrete-time model

Our goal is to design the equalizer which can remove (or suppress) ISI. To do so, we translate the continuous-time communication system model in Figure 4.9 to an equivalent discrete-time model that is easier to work with. The following steps describe the translation process:

Instead of considering AWGN being added before the receiving lter , we can consider an equivalent colored noise is the output of Figure 4.10. equivalent colored Gaussian noise being added after

due to AWGN. The resulting model is shown in

when we analyze the system.

The

We input a bit or a symbol to the communication system every seconds, and get back a sample at the output of the sampler every seconds. Therefore, we can represent the communication system in Figure 4.10 from the information source to the sampler as a digital lter. Since ,

and are LTI lters, they can be combined and represented by an equivalent digital . The LTI lter. Denote its transfer function by and its impulse response by
result is the discrete time-linear lter model shown in Figure 4.11, in which the output sequence

is given by
4.10

(4.16)

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4. ISI & Equalization

Ik

H(z)

Ik

Equalizer

Decision

nk
Figure 4.11: Equivalent discrete-time communication system model with colored noise

HW ( z )

EQZ

Figure 4.12: Typical equalizer In general,

consists of samples of the colored Gaussian noise (AWGN ltered by ), and is not
white in general.

for some

Therefore, ISI is present. Notice that the noise sequence

Usually, the equalizer consists of two parts, namely, a noise-whitening digital lter and an equalizing circuit that equalizes the noise-whitened output as shown in Figure 4.12. The

is to whiten the noise sequence so that the noise samples are uncorrelated. Notice that depends only on , and can be determined a prior according to our choice of . At the output of , the noise sequence is white. Therefore, equivalently, we can consider the equivalent discrete-time model shown in Figure 4.13, in which is an
effect of AWGN sequence.

Let

. The communication system from the information source to the output

of the noise whitening lter can now be represented by the discrete-time white-noise linear lter model in Figure 4.14. The output sequence is given by

(4.17)

where

is the impulse response corresponding to the transfer function

4.11

, and

is an

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4. ISI & Equalization

Ik

H(z)

HW ( z )

~ Ik

EQZ

Decision

~ nk

Figure 4.13: Equivalent discrete-time communication system model with white noise

Ik

G(z)

~ Ik

EQZ

Decision

~ nk
Figure 4.14: Equivalent discrete-time white-noise linear lter model AWGN sequence. We will work with this discrete-time model in all the following sections.

Finally, the equalizing circuit (we simply call it the equalizer from now on) attempts to remove ISI from the output of

. The focus of our coming discussion is the design of this equalizer. Suppose that the equalizer is also an LTI lter with transfer function and corresponding
impulse response . Then the output of the equalizer is given by

(4.18)

Ideally, contains only contributions from the current symbol and the AWGN sequence with small variance.

4.3.2 Zero-forcing equalizer

First, let us consider the use of a linear equalizer, i.e., we employ an LTI lter with transfer function

as the equalizing circuit. The simplest way to remove the ISI is to choose so that the output of the equalizer gives back the information sequence, i.e., for all if noise is not present. This can be achieved by simply setting the transfer function . This method
is called zero-forcing equalization since the ISI component at the equalizer output is forced to zero.
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4. ISI & Equalization

In general, the corresponding impulse response truncation and delay is applied to get an approximation.

can be an innite length sequence. Suitable

We note that the effect of the equalizing lter on the noise is neglected in the development of the zero-forcing equalizer above. In reality, noise is always present. Although the ISI component is forced to zero, there may be a chance that the equalizing lter will greatly enhancing the noise power and hence the error performance of the resulting receiver will still be poor. To see this, let us evaluate the signal-to-noise ratio at the output of the zero-forcing equalizer when the transmission lter xed and the matched lter is used as the receiving lter, i.e.,

is

(4.19)

In this case, it is easy to see that the digital lter is given by

(4.20)

and the PSD of the colored Gaussian noise samples in Figure 4.11 is given by

(4.21)

Hence, the noise-whitening lter can be chosen as

(4.22)

and then the PSD of the whitened-noise samples lter

in Figure 4.14 is

is simply . As a result, the overall digital

(4.23)

Now, we choose the zero-forcing lter

as

(4.24)

Since the zero-forcing lter simply inverts the effect of the channel on the original information symbols

, the signal component at its output should be exactly . If we model the as iid random variables
with zero mean and unit variance, then the PSD of the signal component is
4.13

and hence the signal

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4. ISI & Equalization

energy at the output of the equalizer is just component at the output of the equalizer is
output is

. Hence the noise energy at the equalizer

. On the other hand, the PSD of the noise

. Dening the SNR as the ratio of the signal energy to the noise

energy, we have SNR

(4.25)

Notice that the SNR depends on the folded spectrum of the signal component at the input of the receiver. If there is a certain region in the folded spectrum with very small magnitude, then the SNR can be very poor.

4.3.3 MMSE equalizer

The zero-forcing equalizer, although removes ISI, may not give the best error performance for the communication system because it does not take into account noises in the system. A different equalizer that takes noises into account is the minimum mean square error (MMSE) equalizer. It is based on the mean square error (MSE) criterion.

beforehand, we model each symbol as a random variable. Assume that the information sequence is WSS. We choose a linear equalizer to minimize the MSE between the original information symbols and the output of the equalizer : MSE E E (4.26)
Without knowing the values of the information symbols Let us employ the FIR lter of order

shown in Figure 4.15 as the equalizer.

We note that a

delay of symbols is incurred at the output of the FIR lter. Then MSE E E where

(4.27)

4.14

(4.28) (4.29)

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4. ISI & Equalization

~ I

-1

-1

...

-1

h E,-L

hE,L

^ I k-L

Figure 4.15: FIR lter as an MMSE equalizer We want to minimize MSE by suitable choices of and setting the result to zero, we get E Rearranging, we get

. Differentiating with respect to each

(4.30)

where

(4.31)

E E

(4.32) (4.33)

If and

are available, then the MMSE equalizer can be found by solving the linear matrix equation

(4.31). It can be shown that the signal-to-noise ratio at the output of the MMSE equalizer is better than that of the zero-forcing equalizer. The linear MMSE equalizer can also be found iteratively. First, notice that the MSE is a quadratic function of . The gradient of the MSE with respect to gives the direction to change for the largest increase of the MSE. In our notation, the gradient is
4.15

. To decrease the MSE, we

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4. ISI & Equalization

can update

in the direction opposite to the gradient. This is the steepest descent algorithm:

At the th step, the vector

is updated as

(4.34)

where

is a small positive constant that controls the rate of convergence to the optimal solution. in advance. However, the transmitter can transmit

In many applications, we do not know and

a training sequence that is known a priori by the receiver. With a training sequence, the receiver can estimate and . Alternatively, with a training sequence, we can replace and at each step in the steepest descent algorithm by the rough estimates and , respectively. The algorithm becomes:

(4.35)

This is a stochastic steepest descent algorithm called the least mean square (LMS) algorithm.

4.3.4 LS equalizer
In the training period for the MMSE equalizer, the data sequence, i.e., the training sequence is known to the equalizer. Instead of minimizing the MSE, which is a statistical average, we can actually minimize the sum of the square errors. This is called the least squares (LS) criterion. Suppose that the known sequence lasts for symbols. Then the sum of the square errors is given by

(4.36)

Differentiating with respect to

and setting the result to zero, we get

(4.37)

This time,

(4.38) (4.39)

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4. ISI & Equalization

Suppose that we are given one more training symbol. Apparently, we have to recalculate and

, and solve the matrix equation all over again. However, actually, there is a more efcient approach. Assuming is non-singular, . Notice that

(4.40) (4.41)

Using the matrix inversion lemma5, we get

(4.42) (4.43)

The procedure is called the recursive least squares (RLS) algorithm. In many cases, the RLS algorithm converges much faster than the steepest descent algorithm at the expense of more complex computation.

4.3.5 Decision feedback equalizer

Recall from the equivalent discrete-time model in Figure 4.14 that

(4.44)

The current symbol we want to determine is . If we had known the other symbols exactly, an obvious approach to eliminate ISI would be to subtract their effects off6 , i.e., the equalizer would give

(4.45)

In general, we do not know all the symbols that are affecting the reception of the current symbol. However, it is possible to use previously decided symbols (output from the decision device) provided that we have made correct decisions on them. This approach is called decision feedback equalization. With decision feedback, we can think of the equalizer to contain two parts a feedforward part and a 5
6

Assuming all the required invertibilities, Of course, we need to have knowledge of

to be able to do so.

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4. ISI & Equalization

feedback part. Suppose that the feedforward lter is of order and the feedback lter is of order

. Then

(4.46) can be found by minimizing the

where are the decided symbols. Again, the lter coefcients

MSE. In general, signicant improvement over linear equalizers can be obtained with the decision feedback equalizer. Consider a DFE with a feedforward lter of order and a feedback lter of order . Assume perfect decision feedback, i.e.,

. Then

(4.47)

where

(4.48) (4.49)

(4.50) (4.51)

Further assume that the data symbols are zero-mean unit-variance iid random variables. We seek the lters and that minimize the MSE given by E

and E E

, we get

(4.52)

Differentiating with respect to

(4.53) (4.54)

Notice that E and

and E
E

, i.e., the identity matrix. The equations for optimal

reduce to

E E
4.18

(4.55) (4.56)

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4. ISI & Equalization

Solving these equations, we have E

E E

(4.57) (4.58)

Similar to the case of the MMSE equalizer, we can also solve for the feedforward and feedback lters using the steepest descent approach. If we do not know the expectations of the matrices above a priori, we can send a training sequence to facilitate the estimation of them.

4.4 Maximum Likelihood Sequence Receiver

Whenever it is not practical to construct ISI-free transmission pulses, we can use an equalizer to eliminate (or reduce) ISI, and then make a decision on the current symbol based on the equalizer output. Although this approach is simple and practical, we have no idea whether it is optimal in terms of minimizing the average symbol error probability. In fact, it turns out that all the equalization methods discussed in the previous section are not optimal. Because of the fact that the effect of a symbol is spread to other symbols, it is intuitive that the optimal receiver should observe not only the segment of received signal concerning the desired symbol, but the whole received signal instead. As a matter of fact, this strategy is also employed in the equalization techniques described previously7. Using the whole received signal, we can employ the MAP principle to develop the optimal symbolby-symbol detector, which decides one transmitted symbol at a time, to minimize the average symbol error probability. The development is rather involved and the optimal symbol-by-symbol detector is usually too complex to implement. Here, we opt for another possibility. Instead of deciding a transmitted symbol at a time, we can consider to decide the whole transmitted symbol sequence simultaneously from the received signal. In this way, we aim at minimizing the probability of choosing the wrong sequence of symbols instead of the average symbol error probability. With this sequence detection approach, we can employ the ML principle to achieve our goal. The resulting optimal receiver is referred to as the maximum likelihood sequence (MLS) receiver.
7

We assume the use of FIR lters for the MMSE, LS, and decision-feedback equalizers. In these cases, we are not

strictly observing the whole received signal, but some long segments of it.

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4. ISI & Equalization

Now, let us develop the MLS receiver. Suppose we transmit a sequence of information bits with rate ceived signal up to time , i.e., for

symbols per second and the transmission and channel lters are both causal. The re-

, is

(4.59)

This is simply a rewrite of (4.3) with the above assumptions applied. As mentioned above, we try to decide the sequence of transmitted symbols

8. To do so, we treat the whole summation on the right hand side of (4.59) as our desired signal9.
With this point of view, we are equivalently considering an AWGN channel with

based on our observation up to time

-ary communication system over the

exhausting all the possible transmitted bit sequences. Each bit sequence

is treated as an -ary symbol. This is the problem we have solved. Assuming all the bit sequences are equally probable, we can employ the ML receiver to minimize the probability of making an error when determining the transmitted sequence. From the results in Section 2.5.3, we know that the ML estimator of the bit sequence

(4.60)

up to time is

(4.61)

Dening

and

, we can write the metric as

(4.62)

Then the MLS receiver, which is depicted in Figure 4.16, decides

(4.63)

In order to make the decision, the MLS receiver has to check all the sequences to nd the one
8 9

We cannot decide the future symbols until we have observed them. Ironically, although our goal is to combat ISI, there is no interference in this interpretation.

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4. ISI & Equalization

@t=nT

r(t)

~ v(t)

Calculate metrics

Select largest

^ bK

Figure 4.16: Maximum likelihood sequence receiver that gives the largest metric. Hence its complexity increases exponentially with time. This is clearly impractical and this is why the relatively simple equalization techniques described in the previous section are more popular in practice. When is of nite-support10, i.e., the metric in the following way:

for

, a signicant simplication can be applied

to the calculation of the maximum metric rendering the MLS receiver practical. First, we can update

(4.64)

We are going to make use of two important observations from (4.64) to simplify the maximization of the metric. The rst observation is that the updating part (the second term on the right hand side) depends only on

If we decompose into

(4.65)

, (4.64) can be rewritten as

(4.66)

where

(4.67)

Recall our goal is to maximize . From (4.66)

10

(4.68)

This is generally not true for a channel with a bandlimited response unless some form of transmission pulse design is

applied. However, even when is not strictly of nite-support, the simplication here can still be applied approximately provided decays fast enough.

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4. ISI & Equalization

where

(4.69)

is the sequence that maximizes , then the segment

The second equality above is due to our second important observation that if

as long as we know the values of for all the patterns of . The order of complexity of the maximization reduces dramatically from to , which does not grow with time. However, there is still a problem needed to be solved before we can benet from this large complexity reduction. We need to calculate let each possible pattern of system at the time instant arrives at time

tant simplication in the maximization of that we do not need to test all the patterns of

This contradicts with our original claim that maximizes . This observation reveals the impor-

11 . Then it is clear from (4.66) that

Indeed, suppose that there is another segment such that

, where

must be the one that maximizes

for each of the patterns of . Fortunately, this

task can be efciently accomplished by an iterative algorithm known as the Viterbi algorithm. First,

represents a state of the communication . There are altogether states. When the new input bit

, the system changes state from to . When the received sample is available at the output of the matched lter in Figure 4.16, the receiver calculates making use of (4.68). Hence, all the receiver needs to know is the value of

for each state . But

(4.70)

and the

Therefore, during the transition from state to state , we can use (4.70) to update and store
that maximizes it:

11

The important observation here is that does not depends on .

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4. ISI & Equalization

SK-3
[+1,+1] [+1,-1] [-1,+1] [-1,-1]

SK-2

SK-1

SK

bn = -1 bn = +1
Figure 4.17: Trellis diagram for the case of

(4.71)

with as the sequence that maximizes . The other previous information can be discarded. This update is usually performed in a backward manner. For each state

namely

, we know that the transition can only come from two previous states,

and

. Therefore,

(4.72) If is the one chosen above, then needed values of for the determination of

. Otherwise, . We perform this update for all the states at state transition to make the

(4.73)

The update relations of the states are depicted by the trellis diagram. The trellis diagram for the case of

is given in Figure 4.17.

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Wong & Lok: Theory of Digital Communications

4. ISI & Equalization

Let us end our discussion on the MLS receiver by making two remarks. First, the Viterbi algorithm described above can be easily generalized to the cases of non-binary symbols. Second, although the MLS receiver minimizes only the sequence error probability, its symbol error performance is usually better than that of the equalization techniques.

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