Digital Transmission Through Bandlimited AWGN Channel

Chapter 8
Digital Transmission through Bandlimited

AWGN Channel
Chapter 8
Digital Transmission through Bandlimited AWGN Channel
In this chapter, we treat digital communication over a channel
that is modeled as a linear filter with a bandwidth limitation
The bandwidth constrain generally precludes the use of
rectangular pulses at the output of the modulator. Instead, the
transmitted signals must be shaped to restrict their bandwidth
to that available on the channel
The channel distortion results in intersymbol interference (ISI)
at the output of the demodulator and leads to an increase in the
probability of error at the detector.
Devices or methods for correcting or undoing the channel
distortion, called channel equalizers.
8.1 DIGITAL TRANSMISSION THROUGH BANDLIMITED
CHANNELS
A bandlimited channel is characterized as a linear filter with
impulse response c(t) and frequency response c(f),

2
( ) ( )
j ft
C f c t e dt
t
=
}
DIGITAL TRANSMISSION THROUGH BANDLIMITED CHANNELS
If the channel is a baseband that is bandlimited to B
c
,then
C(f)=0 for |f|> B
c
Suppose that the input to a bandlimited channel is a signal
waveform g
T
(t). Then the response of the channel is the
convolution of g
T
(t) with c(t) ;i.e.,

Expressed in the frequency domain, we have

H(f)=C(f)G
T
(f)
( ) ( ) ( ) ( ) ( )
T T
h t c g t d c t g t t t t
= =
}
The signal at the input to the demodulator is of the form h(t)+n(t),
where n(t) denotes the AWGN. Let us pass the received signal
h(t)+n(t) through the matched filter that has a frequency response

where t
0
is some nominal time delay at which we sample the filter
output
The signal component at the output of the matched filter at the
sampling instant t=t
0
is

The noise component at the output of the matched filter has a zero
mean and a power-spectral density

0
2
( ) ( ) 8.1.4
j ft
R
G f H f e
t -
=
2
0
( ) ( ) 8.1.5
s h
y t H f df c
= =
}
2
0
( ) ( ) 8.1.6
2
n
N
S f H f =
The noise power at the output of the matched filter has a
variance

The SNR at the output of the matched filter is

Compared to the previous result, the major difference in this
development is that the filter impulse response is matched to the
received signal h(t) instead of the transmitted signal

2
2
0 0
( ) ( ) 8.1.7
2 2
h
n n
N N
S f df H f df
c
o

= = =
} }
2
0 0
2
8.1.8
/ 2
h h
o h
S
N N N
c c
c
| |
= =
|
\ .
Example 8.1.1
The signal pulse g
T
(t), defined as

The channel output is corrupted by AWGN with power spectral
density . determine the matched filter to the received
signal and the output SNR
1 2
( ) 1 cos ( ) , 0
2 2
T
T
g t t t T
T
t
(
= + s s
(

0
/ 2 N
DIGITAL TRANSMISSION THROUGH BANDLIMITED
CHANNELS
The spectrum of the signal is

2 2
2 2
sin
( )
2 2 (1 )
sin
=
2 (1 )
j fT
T
j fT
T fT
G f e
fT f T
T c fT
e
f T
t
t
t
t
t
( ) ( ) ( )
( )

0 ,
T
T
H f C f G f
G f f W
otherwise
=
s

The signal component at the output of the filter matched to H(f)
is

The variance of the noise component is

The output SNR is

2
2
2 2 2 2
2
2 2 2 2
( )
1 (sin )
=
(2 ) (1 )
(sin )
=
(2 ) (1 )
W
h T
W
W
W
WT
WT
G f df
fT
df
f f T
T
d
T
c
t
t
to
o
t o o
}
}
}
2
2
0 0
( )
2 2
W
h
n T
W
N N
G f df
c
o

= =
}
0 0
2
h
S
N N
c
| |
=
|
\ .
The amount of signal energy at the output of the matched filter
depends on the value of the channel bandwidth W.
The maximum value of
h
, obtained as W, is

???

2 2
0
max ( ) ( )
T
h T T
G f df g t df c

= =
} }
8.1.1Digital PAM Transmission through Bandlimited Baseband Channel
Let us consider the baseband PAM communication system
illustrated by the functional block in Figure 8.3.

Digital PAM Transmission through Bandlimited Baseband Channel
First we consider the M-ary PAM, the input binary data sequence
is subdivides into k-bit symbols and each symbol is mapped into a
corresponding amplitude level. The amplitude modulates the
output of the transmitting filter. The baseband signal at the output
of the transmitting filter may be expressed as

where T=k / R
b
is the symbol interval, R
b
is the bit rate and {a
n
} is a
sequence of amplitude corresponding to the sequence of k-bit
blocks of information bits
The received signal at the demodulator, may be expressed as

is the impulse response of the channel, and n(t)
represents the AWGN

( ) ( ) 8.1.9
n T
n
t a g t nT u

=
=
( ) ( ) ( ) 8.1.10
n
n
r t a h t nT n t
=
= +
( ) ( ) ( )
T
h t c t g t =
The output of the receiving filter may be expressed as

To recover the information symbols {a
n
}, the output of the
receiving filter is sampled periodically, every T seconds. The
sampler produces
( ) ( ) ( ) 8.1.11
n
n
y t a x t nT v t
=
= +
( ) ( ) ( ) ( ) ( ) ( ) and ( ) ( ) ( ) denotes
the additive noise at the output of the receiving filter
R T R R
x t h t g t g t c t g t v t n t g t = = =
( ) ( ) ( ) 8.1.12
n
n
y mT a x mT nT v mT
=
= +

or equivalently

The desired symbol a
m
, is scaled by the gain parameter x
0
.
When receiving filter is matched to the received signal h(t), the
scale factor is
0
= 8.1.13
where ( ), ( ), 0, 1, 2,.....
m n m n m
n
m n m n m
n m
m m
y a x v
x a a x v
x x mT v v mT and m
=
= +
+ +
= = =
2
2
0
W
2 2
T T
-W
( ) ( )
= G ( ) C ( ) 8.1.14
h
x h t dt H f df
f f df c

= =
=
} }
}
The second term on the RHS of Equation (8.1.13) represents the
effect of the other symbols at the sampling instant t = mT, called
the intersymbol interference (ISI).
v
m
is a zero-mean Gaussian random variable with variance

By appropriate design of the transmitting and receiving filter, it
is possible to satisfy the condition x
n
=0 for , so that the ISI
term vanishes
2
0
/ 2
n h
N o c =
0 n =
8.1.2 Digital transmission through Bandlimited Bandpass Channels
The baseband PAM given by in Equation (8.1.9) modulates the
carrier, so that the transmitted signal u(t) is simply

A QAM signal in its simplest form, may be viewed as two amplitude-
modulated carrier signals in phase quadrature. That is, the QAM
signal may be expressed as

and {a
nc
} and {a
ns
} are the two sequence of amplitudes carried on
the two quadrate carriers.
( ) t u
( ) ( ) cos 2 8.1.15
c
u t t f t u t =
( ) ( ) cos 2 ( ) sin 2 8.1.16
where
( ) ( )
8.1.17
( ) ( )
c c s c
c nc T
n
s ns T
n
u t t f t t f t
t a g t nT
t a g t nT
u t u t
u
u
=
= +
=
=
Digital transmission through Bandlimited Bandpass Channels

The equivalent complex-valued baseband signal

The corresponding bandpass QAM signal u(t) may also be
represented as

-
( ) ( ) ( )
( ) ( )
( ) 8.1.18

c s
nc ns T
n
n T
n
n nc ns
t t j t
a ja g t nT
a g t nT
a a ja
u u u
=
=
=
=
=
2
( ) Re ( ) 8.1.19
c
j f t
u t t e
t
u
(
=

When transmitted through the bandpass channel, the received bandpass
signal may be represented as

where r(t) is the equivalent lowpass (baseband) signal, which may be
expressed as

In the case of baseband transmission, h(t) is the impulse response of
the cascade of the transmitting filter and the channel

2
( ) Re ( ) 8.1.21
c
j f t
w t r t e
t
(
=

( ) ( ) ( ) 8.1.22
n
n
r t a h t nT n t
=
= +
( ) ( ) ( )
T
h t c t g t =
The received bandpass signal can be converted to a baseband signal by
multiplying w(t) with the carrier signals and and
eliminating the double frequency terms by passing the two quadrature
components through separate lowpass filters, as shown in Figure8.4
The two quadrature components at the output of these lowpass filters
can be expressed as an equivalent complex-valued signal of the form

cos 2
c
f t t sin 2
c
f t t
( ) ( ) ( ) 8.1.23
n
n
y t a x t nT v t
=
= +

8.2 THE POWER SPECTRUM OF DIGITALLY MODULATED
SIGNALS
8.2.1 The Power Spectrum of the bsaeband signal
The equivalent baseband tansmitted signal for a digital PAM, PSK, or
QAM signal is represent in the general form as

The information sequence {a
n
} is random, and g
T
(t) is the impulse
response of the transmitting filter, v(t) is a sample function of a random
process V(t)
The mean value of V(t) is

( ) ( ) 8.2.1
n T
n
v t a g t nT
=
=
[ ( )] ( ) ( )
( ) 8.2.2
n T
n
a T
n
E V t E a g t nT
m g t nT
=
=
=
8.2.1 The Power Spectrum of the baseband signal

The autocorrelation function of V(t) is

In general, we assume that the information sequence {a
n
} is wide-
sense stationary with autocorrelation sequence

Hence ,Equation(8.2.3) may be expressed as

-
( , ) [ ( ) ( )]
= ( ) ( ) ( ) 8.2.3
V
n m T T
n m
R t t E V t V t
E a a g t nT g t mT
t t
t
-

-
= =
+ = +
+

( ) ( ) 8.2.4
a m n m
R n E a a
-
+
=
-
( , ) [ ( ) ( )]
= ( ) ( ) ( )
= ( ) ( ) ( ) 8.2.5
V
a T T
n m
a T T
m n
R t t E V t V t
R m n g t nT g t mT
R m g t nT g t nT mT
t t
t
t
-

= =

= =
+ = +
+
+

The second summation in Equation(8.2.5) is periodic with
period T, namely the autocorrelation function is periodic
in the variable t

Therefore, the random process V(t) has a periodic mean and periodic
autocorrelation. Such a random process is cyclostationary.
The average autocorrelation function of V(t)
( , )
V
R t t t +
( , ) ( , ) 8.2.7
V V
R t T t T R t t t t = + + + = +
/ 2
/ 2
/ 2
/ 2
/ 2
/ 2
1
( ) ( , )
1
= ( ) ( ) ( )
1
= ( ) ( ) ( )
1
= ( ) ( ) ( )
T
T
V V
T
T
a T T
T
m n
nT T
a T T
nT T
m n
a T T
m
R R t t dt
T
R m g t nT g t nT mT dt
T
R m g t g t mT dt
T
R m g t g t mT dt
t t
t
t
t
= =

+

= =
=
= +
+
+
+
}

}

}
}
8.2.8
The time-autocorrelation function of g
T
(t) is defined as

The average autocorrelation function of V(t) becomes

The Fourier transform of Equation(8.2.10) becomes

( ) ( ) ( ) 8.2.9
g T T
R g t g t dt t t
= +
}
1
( ) ( ) ( ) 8.2.10
V a g
m
R R m R mT
T
t t
=
=
2
2
2
( ) ( )
1
= ( ) ( )
T
1
= ( ) ( ) 8.2.11
T
j f
V V
j f
a g
m
a T
S f R e d
R m R mT e d
S f G f
t t
t t
t t
t t
=
=
}
2 2
| ( ) |
j fmT
T
G f e
t
S
a
(f) is the power-spectrum of the information sequence {a
n
}

and G
T
(f) is the spectrum of the transmitting filter. is the Fourier
transform of
The power-spectral density S
a
(f) is periodic in frequency with period 1/T,
we note that S
a
(f) has the form of an exponential Fourier series with
{R
a
(m)} as the Fourier coefficients.

We consider the case in which the information symbols in the sequence
{a
n
} are mutually uncorrelated

2
( ) ( ) 8.2.12
j fmT
a a
m
S f R m e
t
=
=

2
( )
T
G f
( )
g
R t
1/ 2
2
1/ 2
( ) ( ) 8.2.13
T
j fmT
a a
T
R m T S f e df
t
=
}
2 2
2
, 0
( ) 8.2.14
, 0
a a
a
a
m m
R m
m m
o + =
=

=

We obtain the power-spectral density

The term involving the summation on the RHS of equation (8.2.15)
is periodic with period 1/T . It may be viewed as the exponential
Fourier series of a periodic train of impulse where each impulse
has an area 1/T

2 2 2
( ) 8.2.15
j fmT
a a a
m
S f m e
t
o

=
= +

2
2
( ) ( ) 8.2.16
a
a a
m
m m
S f f
T T
o o
=
= +

By substituting for S
a
(f) into (8.2.11)

The first term is the continuous spectrum and its shape
depends of G
T
(f). the second term consists of discrete frequence
components spaced 1/T apart frequency
We can eliminate the discrete frequency component by selecting {a
n
}
to have zero mean. Under the condition that m
a
=0, we have

2
2 2
2
2
( ) ( ) ( ) ( ) 8.2.17
a a
V T T
m
m m m
S f G f G f
T T T T
o
o
=
= +
2
2
( )
a
T
G f
T
o
2
2
( ) ( ) 8.2.18
a
V T
S f G f
T
o
=
Example 8.2.1
Determine the power-spectral density in Equation (8.2.17), when g
T
(t) is
the rectangular pulse

The Fourier transform of g
T
(t) is

2
2 2 2 2
sin
( )
Hence
sin
( ) ( ) ( ) =( ) sin ( )
j fT
T
T
fT
G f AT e
fT
fT
G f AT AT c fT
fT
t
t
t
t
t
=
=
By substitution for into Equation (8.2.17)

2
( )
T
G f
2 2 2 2 2
2 2 2 2 2
sin
( ) ( ) ( )
= sin ( ) ( )
V a a
a a
fT
S f A T A m f
fT
A T c fT A m f
t
o o
t
o o
= +
+
Example 8.2.2
Consider a binary sequence {b
n
}, from which we form the symbols

The {b
n
} are assumed to be uncorrelated binary valued ( ) random
variable, each having a zero mean and a unit variance. Determine
the power-spectral density of the transmitted signal
The autocorrelation function of the sequence {a
n
} is

1 n n n
a b b

= +
1
1 1
( ) [ ]
[( )( )]
2 =0
= 1 = 1
0 otherwise
a n n m
n n n m n m
R m E a a
E b b b b
m
m
+
+
=
= + +

The power-spectral density of the input sequence is

2
( ) 2(1 cos 2 )
=4cos
a
S f fT
fT
t
t
= +
The corresponding power spectrum for the modulated signal is,
from Equation (8.2.17)

2
2
4
( ) ( ) cos
V T
S f G f fT
T
t =
8.2.2 The Power spectrum of a Carrier-Modulated Signal
Let us consider the bandpass PAM signal as an Example. The
autocorrelation function of the bandpass signal

is

( ) ( ) cos 2
c
u t t f T u t =
( , ) [ ( ) ( )]
= [ ( ) ( )]cos 2 cos 2 ( )
= ( , ) cos 2 cos 2 ( )
U
c c
V c c
R t t E U t U t
E V t V t f t f t
R t t f t f t
t t
t t t t
t t t t
+ = +
+ +
+ +
8.2.2 The Power spectrum of a Carrier-Modulated Signal
By expressing the product of the two cosine function

Then , the average of over a signal period T yields

The Fourier transform of yields the power spectrum of the
bandpass signal as

Although the derivation that resulted in Equation (8.2.23) was
carried out for a bandpass PAM signal, the same expression applies
to QAM and PSK. The three bandpass signals differ only in the
autocorrelation R
a
(m) and power spectrum S
a
(f) of the sequence
{a
n
}.
1
( , ) ( , )[cos 2 cos 2 (2 )]
2
U V c c
R t t R t t f f t t t t t t t + = + + +
( , )
U
R t t t +
1
( ) ( ) cos 2 8.2.22
2
U V c
R R f t t t t =
( )
V
R t
1
( ) [ ( ) ( )] 8.2.23
4
U V c V c
S f S f f S f f = + +
8.3 SIGNAL DESIGN FOR BANDLIMITED CHANNELS
Recall form section 8.1.1 the transmitted signal in a digital PAM
or PSK or QAM may be express as

and received signal may be express as

h(t) is c(t) g
T
(t), c(t) is the impulse response of the channel, g
T
(t)
is the impulse response of the transmission filter, and n(t) is a
sample function of an AWGN process.
( ) ( ) 8.3.1
n T
n
t a g t nT u

=
=
( ) ( ) ( ) 8.3.2
m
n
r t a h t nT n t
=
= +

Since H(f) = C(f)G
T
(f), the condition for distortion-free
transmission is that the frequency response characteristic C(f) of
the channel have a constant magnitude and a linear phase over
the bandwidth of the transmitted signal ;i.e.,

where W is the available channel bandwidth , t
0
represents an
arbitrary finite delay, which we set to zero for convenience, and
C
0
is a constant gain factor which we set to unity for
convenience. Under the condition that the channel is distortion-
free, H(f)=G
T
(f) for |f| W and zero for |f| > W, the matched
filter has a frequency response and its sampling
output is

where x(t)=g
T
(t) g
R
(t)
0
2
0
,
( ) 8.3.3
0 ,
j f t
C e f W
C f
f W
t
=

>
( ) (0) ( ) ( ) 8.3.5
m n
n
y mT x a a x mT nT v mT
=
= + +
* *
( ) ( )
T
H f G f =
or more simply,

The middle term on the RHS of Equation(8.3.5) represent the ISI.
The resulting oscilloscope display is called an eye pattern
0
8.3.4
m m n m n m
n m
y x a a x v
=
= + +
( ) ( ) ( )
m
n
r t a x t nT n t
=
= +

8.3.1 Design of Bandlimited Signals for Zero ISIThe Nyquist Criterion
We have seen that the output of the receiving filter, sampled at t
= mT is given by

The Fourier transform of x(t) is X(f) = G
T
(f)C(f)G
R
(f) where G
T
(f)
G
R
(f) denote the transmitter and receiver filters frequency
response and C(f) denotes the frequency response of the channel
To remove the effect of ISI, it is necessary and sufficient that
x(mT-nT) = 0 for n m and x(0) 0, we can assume x(0) = 1

( ) (0) ( ) ( )
m n
n
y mT x a a x mT nT v mT
=
= + +
1, 0
( ) 8.3.7
0, 0
n
x nT
n
=
=

=

Theorem 8.3.1 [Nyquist]. A necessary and sufficient condition for x(t) to
satisfy

X(f) satisfy

Proof .

At the sampling instant t = nT, this relation becomes

1, 0
( ) 8.3.8
0, 0
n
x nT
n
=
=

=
( ) 8.3.9
m
m
X f T
T
=
+ =
2
( ) ( ) 8.3.10
j ft
x t X f e df
t
=
}
2
( ) ( ) 8.3.11
j fnT
x nT X f e df
t
=
}
Let us break up the integral in Equation (8.3.11) into integral
covering the finite range of 1/T. thus we obtain

Where we have defined Z(f) by

(2 1) / 2
2
(2 1) / 2
1/ 2
2
1/ 2
1/ 2
2
1/ 2
1/ 2
2
1/ 2
( ) ( )
( )
[ ( )]
( ) 8.3.12
m T
j fnT
m T
m
T
j fnT
T
m
T
j fnT
T
m
T
j fnT
T
x nT X f e df
m
X f e df
T
m
X f e df
T
Z f e dt
t
t
t
t
=
= +
= +
=
}
}
( ) ( ) 8.3.13
m
m
Z f X f
T
=
= +

Z(f) is a periodic function with period 1/T, and it can be expanded
in term of its Fourier series coefficients {z
n
} as

where

Comparing Equation (8.3.15) and (8.3.12) we obtain

The necessary and sufficient conditions for Equation(8.3.8) to be
satisfied is that

2
( ) 8.3.14
j nfT
n
n
Z f z e
t
=
=

1
2
2
1
2
( ) 8.3.15
j nfT
T
n
T
z T Z f e df
t
=
}
( ) 8.3.16
n
z Tx nT =
, 0
8.3.17
0, 0
n
T n
z
n
=
=

=

When substituted into Equation (8.3.14), yields

or equivalently,

This concludes the proof of the theorem.

( ) 8.3.18 Z f T =
( ) 8.3.19
m
m
X f T
T
=
+ =

Now we distinguish three cases:
1.T < 1/2W, or equivalently, 1/T > 2W
2.T = 1/2W, or equivalently, 1/T = 2W (the Nyquist rate)

3.T > 1/2W, or equivalently, 1/T < 2W

A particular pulse spectrum, for the T> 1/2W case, that is the
raised cosine spectrum and the frequency characteristic is given
as

is called the rolloff factor, which takes values in the range
01

(1 )
0
2
1 1 1
( ) [1 cos ( )] 8.3.22
2 2 2 2
1
0
2
rc
T f
T
T T
X f f f
T T T
f
T
o
t o o o
o
o
s s
+
= + s s

>
8.3.1 Design of Bandlimited Signals for Zero ISIThe

Nyquist Criterion
The bandwidth occupied by the signal beyond the Nyquist
frequency 1/2T is called excess bandwidth and is usually
expressed as a percentage of the Nyquist frequency.

For example, when =1/2, the excess bandwidth is 50%, and
when =1 the excess bandwidth is 100%. The pulse x(t) having
the raised cosine spectrum is

2 2 2
2 2 2
sin / cos( / )
( )
/ 1 4 /
cos( / )
=sinc(t/T) 8.3.23
1 4 /
t T t T
x t
t T t T
t T
t T
t to
t o
to
o
=

Illustrates the raised cosine spectral

In the special case where the channel is ideal with , we have

In this case, if the received filter is matched to the transmitter filter we
have X
rc
(f)=G
T
(f)G
R
(f)=|G
T
(f)|
2
. ideally,

and , where t0 is some nominal delay that is required to
assure physical realizability of the filter.

( ) ( )
2
f
C f
W
= H
( ) ( ) ( ) 8.3.24
rc T R
X f G f G f =
0
2
( ) ( ) 8.3.25
j ft
T rc
G f X f e
t
=
*
( ) ( )
R T
G f G f =
8.3.2 Design of Bandlimited Signals with Controlled ISI Partial
Response Signal
As we have observed from our discussion of signal design for
zero ISI, It is necessary to reduce the symbol rate 1/T below the
Nyquist rate of 2W symbol/sec in order to realize practical
transmitting and receiving filter.

We choose to relax the condition of zero ISI and achieve a
symbol transmission rate of 2W symbols/sec by allowing for a
control amount of ISI.

We allow one additional nonzero value in the samples {x(nT)}

Response Signal
One special case that leads to (approximately) physically
realizable transmitting and receiving filters is specified by the
samples

Now using Equation(8.3.16), we obtain

Which when substituted into Equation(8.3.14) yields

1, 0,1
( ) 8.3.26
0
n
x nT
otherwise
=
0, -1
8.3.27
0
n
T n
z
otherwise
=
2
( ) 8.3.28
j fT
Z f T Te
t
= +
Response Signal
For , we obtain

Therefore, x(t) is given by

x(t)=sinc(2Wt)+sinc(2Wt - 1) 8.3.30

This pulse id called a duobinary signal pulse.
1
2
T
W
=
2
1
[1 ],
( )
2
0 otherwise
1
cos ,
= 8.3.29
2
0 otherwise
f
j
W
f
j
W
e f W
X f
W
f
e f W
W W
t
t
t
+ <
=

| |
<
|
\ .

Response Signal
Duobinary signal pulse

Response Signal
Another special case that leads to physically realizable
transmitting and receiving filters is specified by the samples

The corresponding pulse x(t) is given as
x(t)=sinc(t+T)/T sinc(t-T)/T 8.3.32
and its spectrum is

1 , -1
( ) ( ) 1 , 1 8.3.31
2
0 ,
n
n
x x nT n
W
otherwise
=
= = =
1
sin ,
2 ( ) 8.3.33
0,
f f
j j
W W
j f
e e f W
W W W X f
f W
t t
t

| |
= s
|
=
\ .
>

Response Signal
This is called a modified duobinary signal pulse. It suitable for
transmission over a channel that does not pass DC.

Response Signal
In general, the class of bandlimited signals pulse that have the
form

and their corresponding spectra

are called partial response signals when controlled ISI is
purposely introduces by selecting two or more nonzero samples
from the set {x(n/2W)}. The resulting signal pulses allow us to
transmit information symbols at the Nyquist rate of 2W symbols
per second.
sin 2 ( / 2 )
( ) ( ) 8.3.34
2 2 ( / 2 )
n
n W t n W
x t x
W W t n W
t
t
/
1
( ) ,
2 2 ( ) 8.3.35
0 ,
jn f W
n
n
x e f W
W W X f
f W
t
>
8.4 PROBABILITY OF ERROR IN DETECTION OF DIGITAL PAM

8.4.1 Probability of Error for Detection of Digital PAM with Zero ISI
In the absence of ISI, the received signal sample at the output of
receiving matched filter has the from

where

In general, a
m
takes one of M possible equally spaced amplitude values
with equal probability.
Given a particular amplitude level, the problem is to determine the
probability of error.
0
8.4.1
m m m
y x a v = +
2
0
2
0
( ) 8.4.2
/ 2 8.4.3
w
T g
w
v g
x G f df
N
c
o c
= =
=
}
8.4 PROBABILITY OF ERROR IN DETECTION OF DIGITAL PAM
The probability of error for digital PAM in a bandlimited, additive
white Gaussian noise channel, in the absence of ISI, is identical to the
Section 7.6.2.

Hence

In the treatment of PAM given this chapter we imposed the additional
constraint that the transmitter signal is bandlimited to the bandwidth
allocated for the channel.
0
2
2( 1)
8.4.4
g
M
M
p Q
M N
c
(
=
(
(

2
3 /( 1),
g av av bav
M k c c c c = =
2
2
0
6(log ) 2( 1)
8.4.5
( 1)
bav
M
M M
p Q
M M N
c
(
=
(

8.4 .2 Symbol-by Symbol Detection of Data with Controlled ISI
In particular, we consider the detection of the duobinary and the
modified duobinary partial response signals.
In the both case, we assume that the desired spectral characteristic X(f)
for the partial response signal is split evenly between the transmitting
and receiving filters; i.e.

For the duobinary signal pulse, the samples at the output of the
receiving filter have the from

Let us ignore the noise for the moment and consider the binary case
where with equal probability. Then b
m
takes on one of three
possible values, b
m
= -2, 0, 2 with corresponding probabilities 1/4, 1/2 ,
1/4.
1/ 2
( ) ( ) ( )
T R
G f G f X f = =
1
8.4.6
m m m m m m
y b v a a v
= + = + +
1
m
a =
If is the detected symbol from the (m-1)st signal interval, its
effect on b
m
, the received signal in the m th signal interval, can be
eliminated by subtraction, thus allowing a
m
to be detected. This
process can be repeated sequentially for every received symbol.
The major problem with this procedure is that errors arising from the
additive noise tend to propagate.
Error propagation can be avoided by precoding the data at the
transmitter instead of eliminating the controlled ISI by subtraction at
the receiver.
The precoding is performed on the binary data sequence prior to
modulation. From the data sequence {d
n
} of 1s and 0s that is to be
transmitted, a new sequence {p
m
} is called the precoded sequence

1
, 1, 2,... 8.4.7
m m m
p d p m
= =
1 m
a

Then , we set a
m
= -1 if p
m
= 0 and a
m
= 1 if p
m
= 1; i.e., a
m=
2p
m
-1.
The noise free samples at the output of the receiving filter are given
as

Consequently,

Since d
m
=p
m
p
m-1
, it follows that the data sequence d
m
is obtained
from b
m
by using the relation

Consequently, if b
m
= 2, d
m
= 0 and b
m
=0, d
m
=1.

1
-1
1
(2 -1) (2 -1)
2( -1) 8.4.8
m m m
m m
m m
b a a
p p
p p
= +
= +
= +
1
1 8.4.9
2
m
m m
b
p p

+ = +
1 1
( ) ( ) ( mod 2)
1(mod 2) 8.4.10
2
m m m m m
m
d p p p p
b

= = +
= +

An example that illustrates the precoding and decoding operations is
given Table 8.1

In this case y
m
=b
m
+ v
m
is compared with the two thresholds set at +1
and -1.the data sequence {d
n
} is obtained according to the detection
rule

m
m
1, if -1<y 1
8.4.11
0, if y 1
n
d
<
=

>

The extension from binary PAM to multilevel PAM signaling using the
duobinary pulse is straightforward. In this case the M - level amplitude
sequence {a
m
} results in a (noise-free) sequence

Which has 2M-1 possible equally spaced levels. The amplitude levels
are determined from the relation

In the absence of noise, the samples at the output of the receiving filter
may be expressed as

1
, =1,2,... 8.4.12
m m m
b a a m
= +
1
2 ( 1)
(mod )
m m
m m m
a p M
p d p M
=
=
1
-1
2[ - ( -1)] 8.4.15
m m m
m m
b a a
p p M
= +
= +
0,1,..., ( 1)
m
p M =
Since d
m
=p
m
+ p
m-1
(mod M), if follows that

An example illustrating multilevel precoding and decoding is given in
table 8.2

( 1)(mod ) 8.4.17
2
m
m
b
d M M = +
In the case of the modified duobinary pulse, the controlled ISI is
specified by the values x(n/2W) = -1, for n=1, x(n/2W) = 1 for n=-1, and
zero otherwise, the noise-free sampled output from the receiving filter
is given as

According to the relation Equation (8.4.14) and

The detection rule for recovering the data sequence {d
m
} from {b
m
} in
the absence of noise is

2
8.4.18
m m m
b a a

=
2
(mod ) 8.4.19
m m m
p d p M
=
( 1)(mod ) 8.4.20
2
m
m
b
d M M = +
8.4 .3 Probability of Error for Detection of Partial Response Signals
We determine the probability of error for detection of digital M-ary
PAM signaling using duobinary and modified duobinary pulses. The
model for the communications system is shown in Figure 8.14

Symbol-by-Symbol Detector
The precoder output is mapped into one of M possible amplitude levels.
Then the transmitting filter with frequency response G
T
(f) has an
output

The partial-response function X(f) is divided equally between the
transmitting and receiving filters.

The matched filter output is sampled at t= nT = n/2W. For the
duobinary signal , the output of the matched filter at the sampling
instant may be expressed as
( ) ( ) 8.4.21
n T
n
t a g t nT u

=
=
( ) ( ) ( ) 8.4.22
T R
G f G f X f =
1

m m m m
m m
y a a v
b v
= + +
= +
, 3 ,..., ( 1)
0, 2 , 4 ,..., 2( 1)
m
m
a d d M d
b d d M d
=
=
And for the modified duobinary signal is

The input transmitted symbols {a
m
} are assumed to be equally probable.
Then, for duobinary and modified duobinary signals, the received
output levels have a (triangular) probability mass function of the form

where b denotes the noise-free received level and 2d is the distance
between any two adjacent received signal levels.
We assume that a symbol error is committed whenever the magnitude of
the additive noise exceeds the distance d. The noise component v
m
is zero-
mean, Gaussian with variance

2

m m m m
m m
y a a v
b v
= +
= +
2
( 2 ) , 0, 1, 2,..., ( -1) 8.4.25
M m
p b md m M
M
= = =
2
2
0
0
0
( )
2
= ( ) 2 / 8.4.26
2
W
v R
W
W
W
N
G f df
N
X f df N
o
t
=
=
}
}
, 3 ,..., ( 1)
0, 2 , 4 ,..., 2( 1)
m
m
a d d M d
b d d M d
=
=
For both the duobinary and the modified duobinary signals. An upper
bound on the symbol probability of error is

But

2
( 2)
-1
0
2
( 2 2 ) ( 2 )
2 ( 2( -1) -2( -1) ) ( -2( -1) )
( 0) 2 ( 2 ) ( 0) ( 2( 1) )
1
(1- ) ( 0)
M
M
m M
M
m
p p y md d b md p b md
p y M d d b M d p b m d
p y d b p b md p b p b M d
p y d b
M
=
=
< > = =
+ + > = =
(
= > = = = =
(

= > =
8.4.27
2 2
/ 2
2
0
2
( 0)
2
=2Q 8.4.28
2
v
x
d
v
p y d b e dx
d
N
o
to
t

> = =
| |
|
|
\ .
}
The average probability of a symbol error is upper-bounded as

For the M-ary PAM signal in which the transmitted levels are equally
probable, the average power at the output of the transmitting filter is

where is the mean square value of the M signal levels,

2
2
0
1
2(1 ) 8.4.29
2
M
d
p Q
M N
t
| |
< |
|
\ .
2
2
2
2
( )
( )
( ) 4
( ) ( ) 8.4.30
w
m
av T
w
w
m
m
w
E a
p G f df
T
E a
X f df E a
T T t
=
= =
}
}
2
( )
m
E a
2 2
2
( 1)
( ) 8.4.31
3
m
d M
E a

=
therefore,

By substituting the value of from Equation (8.4.32) into Equation
(8.4.29), we obtain the upper-bound for the symbol error probability as

Where
av
is the average energy/transmitted symbol, which can be also
expressed in terms of the average bit energy as
av
=k
bav
=(log
2
M)
bav
.

The expression in Equation (8.4.33) for the probability of error of M-ary
PAM holds for both a duobinary and a modified duobinary partial
response signal.
2
2
3
8.4.32
4( 1)
av
p T
d
M
t
=
2
d
2
2 2
0
1 6
2(1 )
4 1
av
M
p Q
M M N
c t
| |
| |
|
<
|
|
\ .
\ .
2
2
0
6(log ) 2( 1)
8.4.5
( 1)
bav
M
M M
p Q
M M N
c
(
=
(

If we compare this result with the error probability of M-ary PAM with
zero ISI, which can be obtained by using a signal pulse with a raised
spectrum, we note that the performance of partial response duobinary
or modified duobinary has a loss of (/4)
2
or 2.1db.
To observe the memory in the received sequence, let us look at the noise-
free received sequence for binary transmission given in Table 8.1.
The sequence {b
m
} is 0,-2,0,2,0,-2,0,2,2,... We note that it is not possible to
have a transition from -2 to +2 or from +2 to -2 in one symbol interval.
In other words, it is not possible to encounter a transition form -2 to +2
or vice versa between two successive received samples from the matched
filter.
8.5 DIGITALLY MODULATED SIGNALS WITH MEMORY
We observed that we can shape the spectrum of the transmitted signal
by introducing memory in the modulation,. The two examples cited in
that section are the duobinary and modified duobinary partial
response signal.
Signal dependence among signals transmitted in different signal
intervals is generally accomplished by encoding the data at the input to
the modulator by means of a modulation code.
Such a code generally places restrictions on the sequence of symbols
into the modulator and introduces memory in the transmitted signal.
Signal dependence among signals transmitted in different signal
intervals can also result from intersymbol interference introduced by
channel distortion.
8.5.1 Modulation Codes and Modulation Signals with Memory
Modulation codes are usually employed in magnetic recoding, in optical
recording, and in digital communications over cable systems to achieve
spectral shaping of the modulated signal that matches the passband
characteristics of the channel.
In magnetic recoding, we encounter two basic problems . One problem
is concerned with the packing density that is used to write the data on
the magnetic medium (disk or tape).

The binary data sequence to be stored is used to generate a write
current. This current may be viewed as the output of the modulator.
The two most commonly used methods to map the data sequence into
the write current waveform ate the so-called NRZ (non-return-to -zero)
and NRZI (non-return-to zero-inverse) methods.
We note that is identical to binary PAM in which the information bit 1
is represented by a rectangular pulse of amplitude A and the
information bit 0 is represented by a rectangular pulse of amplitude
A .
In contrast, the NRZI signal waveform is different from NRZ in that
transitions from one amplitude level to another (A to A or A to
A ),the amplitude level remains the same as the previous signal level
The positive amplitude pulse results in magnetizing the medium on one
(direction) polarity and the negative pulse magnetizes the medium in
the opposite (direction) polarity.
Since the input data sequence is basically random with equally probable
1s and 0s, whether we use NRZ of NRZI, we will encounter level
transitions for A to A or A to A with probability for every data bit
The readback signal for a positive transition (-A to A) is a pulse that is
well modeled mathematically as

Where T
50
is defined as the width of the pulse at its 50% amplitude level

2
50
1
( ) 8.5.1
1 (2 / )
p t
t T
=
+
The readback signal for a negative transition (A to - A) is the pulse p(t).
The value of T
50
is determined by the characteristics of the medium and
the read/write heads.
Now, suppose we write a positive transition followed by a negative
transition, and let us vary the time interval between the two transitions,
which we denote as T
b
(the bit time interval). Figure 8.18 illustrates the
readback signal pulses, which are obtained by a superposition of p(t)
with p(t - T
b
).
The parameter, =T
50
/T
b
, is defined as the normalized density.
We notice that as is increased, the peak amplitudes of the readback
signal are reduced and are also shifted in time from the desired time
instants. In the other words, the pulses interfere with one another, thus,
limiting the density with which we can write.
This problem serves as a motivation to design modulation codes that
take the original data sequence and transform (encode) it into another
sequence that results in a write waveform in which amplitude
transition are spaced further apart.
For example, if we use NRZI, the encoded sequence into the modulator
must contain one or more 0s between 1s.
The second problem encountered in magnetic recording is the need to
avoid (or minimize) having a dc content in the modulated signal (the
write current), due to the frequency-response characteristics of the
readback system and associated electronics
This problem can also be overcome by altering (encoding) the data
sequence into the modulator.

Runlength-Limited Codes
Codes that have a restriction on the number of consecutive 1s or 0s in
a sequence are generally called runlenght-limited code. These codes ate
generally described by two parameters, say d and , where d denotes
the minimum number of 0s between 1s in a sequence, and denotes
the maximum number of 0s between two 1s in a sequence.
When used with NRZI modulation, the effect of placing d zeros
between successive 1s is to spread the transition farther apart, thus,
reducing the overlap in the channel response due to successive
transition. By setting an upper limit on the runlength of 0s ensures
that transitions occur frequently enough so that symbol timing
information can be recovered from the received modulated signal.
Runlength-limited codes are usually called (d , ) codes.

The (d, k)-code sequence constraints may be represented by a finite-
state sequential machine with +1 states, denoted as
i
, 1 i +1, as
shown in Figure 8.19

We observe that an output data bit 0 takes the sequence from state
i
to
i+1
, i . The output data bit a takes the sequence to state
i
, the
output bit from the encoder may be a 1 only when the sequence is in
state
i
, d i +1. when the sequence is in state
i+1
, the output bit is
always 1.
The finite-state sequential machine may also be represented by a state
transition matrix, denoted as D, which is a square (+1) (+1) matrix
with elements d
ij
, where
d
ij
= 1, i d +1
d
ij
= 1, j = i +1 (8.5.2)
d
ij
= 0, otherwise

Example 8.5.1
Determine the state transition matrix for a (d , k) = (1 , 3) code
The (1,3) code has four states, we obtain its state transition matrix
which is

0 1 0 0
1 0 1 0
1 0 0 1
1 0 0 0
D
(
(
(
=
(
(

An important parameter of any (d,)code is the number of sequences
of a certain length, say n, which satisfy the (d, ) constraints. The
number of information bits that can be uniquely represented with N(n)
code sequences is
k = [log
2
N(n)]
where [x] denotes the largest integer contained in x. The maximum
code rate is then R
c
=k/n.
The capacity of a (d, ) code is defined as

C (d, ) is the maximum possible rate that can be achieved with the (d,
) constraints.

Where is the largest real eigenvalue of the state transition matrix
D.
2
1
( , ) lim log ( ) (8.5.4)
n
C d k N n
n
=
2 max
( , ) log (8.5.5) C d k =
max

Example 8.5.2
Determine the capacity of a (d, ) = (1,3) code.
Using the state transition matrix given in Example 8.5.1 for the (1,3)
code, we have

The maximum real root of this polynomial is found to be
max
= 1.4656.
Therefore, the capacity C(1,3) = log
2
max
= 0.5515.
capacities of (d, ) codes for 0 d 6 and 2 k 15, are given in Table 8.3.
We observe that C (d, ) < for d 3, and any value of . The most
commonly used codes for magnetic recording employ a d 2, hence, their
rate R
c
is at least 1/2
4 2
- 1 0 0
1 - 1 0
det(D I) det
1 0 - 1
1 0 0 -
= 1 0 8.5.6

(
(
(
=
(
(

=
Now, let us turn our attention to the construction of some runlength-
limited codes.
In general, (d, ) codes can be constructed either as fixed length codes
or variable-length codes. In a fixed-length code, each bit or block of k
bits, is encoded into a block of n > k bits.
For a given block length n, we may select the subset of the 2n
codewords that satisfy the specified runlength constraints. From this
subset, we eliminate codewords that do not satisfy the runlength
constraint when concatenated. Thus, we obtain a set of codewords that
satisfies the constraints and can be use in the mapping of the input
data bits of the encoder.
Example 8.5.3
Construct a d = 0, = 2 code of length n = 3, and determine its efficiency.
By listing all the codewords, we find that the following five codewords
satisfy the (0,2) constraint: (010), (011), (101), (110), (111).
We may select any four of these codewords and use then to encode the
pairs of data bits (00, 01, 10, 11). Thus, we have a rate k/n = 2/3 code that
satisfies the (0,2) constraint.
The fixed-length code in this example is not very efficient. The capacity
of a (0,2) code is C(0,2) = 0.8791, so that this code has an efficiency of

Surely, better (0,2) codes can be constructed by increasing the block
length n
2/ 3
0.76
( , ) 0.8791
c
R
efficiency
C d k
= = =
Example 8.5.4
construct a d = 1 , = code of length n = 5.
In this case, we are placing no constraint on the number of
consecutive zeros.
There are eight such codewords , which implies that we can encode
three information bits with each codeword. The code is given in Table
8.4

This code has a rate R
c
= 3/5. when compared with the capacity C (1,)
= 0.6942, obtained from Table 8.3, the code efficiency is 0.864, which is
quite acceptable.

The code construction method described in the two example above
produces fixed-length (d, ) codes that are state independent. By state
independent, we mean that fixed-length codewords can be
concatenated without violating the (d, ) constraints.

In general, fixed-length, state-independent (d, ) coeds require large
block lengths, except in cases such as those in the examples above,
where d is small.
Example 8.5.5
Construct a simple variable-length d = 0, = 2 code.
A very simple uniquely decodable (0,2) code is the following:
001
1010
1111
The code in above example has a fixed output block size but a variable
input block size. In general, both the input and output blocks may be
variable.

Example 8.5.6
Construct a (2,7) variable block size code.
The solution to this code construction is certainly not unique nor is it
trivial.
The code is listed in Table 8.5. we observe that the input data blocks of
2, 3, and 4 bits are mapped into output data blocks of 4, 6, and 8
bits ,respectively.
The code rate is R
c
= 1/2. this code rate for all codewords, the code is
called a fixed-rate code.
Another code that has been widely sued in magnetic recording is the
rate 1/2, (d, ) = (1,3) code given in Table 8.6

We observe that the first bit of the two-bit block is redundant and may
be discarded.
The second bit is the information bit . This code is usually called the
Miller code.

We observe that this is a state-dependent code which is described by
the state diagram shown in Figure 8.20

When the encoder is a state S
1
, an input bit 1 results in the encoder
staying in state S
1
and outputs 01. this denoted as 1/01. If the input bit
is a 0 the encoder enters state S
2
and output 01. This is denoted as 0/00.
Similarly, if the encoder is in state S
2
, an input bit 0 causes no
transition and the encoder output is 10. On the other hand , if the
input bit is a 1, the encoder enters state S
1
and output 01.
Trellis Representation of State- Dependent (d, ) Codes.
The state diagram provides a relatively compact representation of a
state-dependent code.
Another way to describe such codes that have memory is by means of a
graph called a trellis.
A trellis is a graph that illustrates the state transitions as a function of
time.
For example, Figure 8.21 shows the trellis for the d = 1, = 3 Miller
code whose state diagram is shown in Figure 8.20

8.5.2 The Maximum-Likelihood Sequence Detector
When the transmitted signal has memory; i.e., the signals transmitted
in successive symbol intervals are interdependent, the optimum
detector bases its decisions on observation of a sequence of received
signals over successive signal intervals.
Consider as an example the NRZI signal described in Section 8.5.1.
The signal transmitted in each signal interval is binary PAM. Hence,
there are two possible transmitted signals corresponding to the signal
points s
1
= -s
2
= , where
b
is the energy/bit. The output of the
matched-filter or correlation-type demodulator for binary PAM in the
kth signal interval may be expressed as

where n
k
is a zero-mean Gaussian random variable with variance

b
c
8.5.10
k b k
r n c = +
2
0
/ 2
n
N o =
The conditional PDFs for the two possible transmitted signals are

8.5.11

Now, suppose we observe the sequence of matched-filter outputs r
1
,r
2
,...,r
k
.
The channel noise is assumed to be white and Gaussian, and E(n
k
n
j
)=0,
kj, the noise sequence n
1
,n
2
,...,n
k
is also white.
2 2
2 2
( ) / 2
1
( ) / 2
2
1
( )
2
1
( )
2
k b n
k b n
r
k
n
r
k
n
f r s e
f r s e
c o
c o
to
to

=
=
Consequently, for any given transmitted sequence s
(m)
, the joint PDF
of r
1
,r
2
,...,r
k
. May be expressed as a product of k marginal PDFs; i,e.,

Where either s
k
= or s
k
=- . Then , given the received sequence
r
1
,r
2
,...,r
k
at the output of the matched filter or correlation type
demodulator, the detector determines the sequence
that maximizes the conditional PDF p( ). Such a detector
is called the maximum-likelihood (ML) sequence detector.
( ) 2 2
1 2
1
( ) / 2
1
( ) 2 2
1
( , ,..., ) ( )
1

2
1
( ) exp[ ( ) / 2 ] 8.5.12
2
m
m
k n k
K
m
k k k
k
K
r s
k
n
K
k m
k k n
k
n
p r r r s p r s
e
r s
o
to
o
to
=

=
=
=
=
=
[
[
b
c
b
c
( )
1 2
{ , ,... }
m m m m
K
s s s s =
1 2
, ,...,
m
k
r r r s
By taking the logarithm of Equation (8.5.12) and neglecting the terms
that are independent of ( r
1
,r
2
,...,r
k
) we find that an equivalent ML
sequence detector selects the sequence s
(m)
that minimizes the
Euclidean distance metric

In searching through the trellis for the sequence that minimizes the
Euclidean distance D( r, s
(m)
), it may appear that we must compute the
distance D( r, s
(m)
) for every possible path (sequence). For the NRZI
example given above, which employs binary modulation, the total
number of paths is 2
k
, where K is the number of outputs obtained from
the demodulator.
We may reduce the number of sequences in the trellis search by using
the Viterbi algorithm to eliminate sequences as new data is received
from the demodulator.
( ) ( ) 2
1
( , ) ( ) 8.5.13
K
m m
k k
k
D r s r s
=
=

The Viterbi algorithm is a sequential trellis search algorithm for
performing ML sequence detection. We described it below in the
context of the NRZI signal. We assume that the search process begins
initially at state S
1
. the corresponding trellis is shown in Figure 8.30. At
time t = T, we receive , from the demodulator and at t = 2T
we receive .

1 1
m
r s n = +
2 2
m
r s n = +
The signal memory is one bit, which we denote as L = 1, we observe
that the trellis reaches its regular ( steady-state ) form after the first
transition. Thus, upon receipt of r
2
at t = 2T (and thereafter), we
observe that there are two signal paths entering each of the two nodes
and two signal paths leaving each node. The two paths entering node
S
1
at t = 2T correspond to the information bits (0,0) and (1,1)
or ,equivalently, to the signal points ( ) and ( ),
respectively. The two paths entering node S
2
at t = 2T correspond to
the information bits (0,1) and (1,0) or, equivalently, to the signal points
( ) and ( ) respectively.
For the two paths entering node S
1
, we compute the two Euclidean
distance metrics

by using the output r
1
and r
2
,
b b
c c
,
b b
c c
,
b b
c c
2 2
2 1 2
2 2
2 1 2
(0, 0) ( ) ( )
(1,1) ( ) ( ) 8.5.14
b b
b b
u r r
u r r
c c
c c
= + + +
= + +
,
b b
c c
The Viterbi algorithm compares these two metrics and discards the
path having the larger (greater distance) metric. The other path with
the lower metric is saved and is called the survivor at t = 2T. The
elimination of one of the two paths may be done without compromising
the optimality of the trellis search, because any extension of the path
with the larger distance beyond t = 2T will always have a larger metric
than the survivor that is extended along the same path beyond t = 2T.

For the two paths entering node S
2
at t = 2T, we compute the two
Euclidean distance metrics

by using the output r
1
and r
2

2 2
2 1 2
2 2
2 1 2
(0,1) ( ) ( )
(1, 0) ( ) ( ) 8.5.15
b b
b b
u r r
u r r
c c
c c
= + +
= +
The two metrics are compared and the signal path with the larger
metric is eliminated. Thus , at t = 2T, we are left with two survivor
paths, one at node S
1
and the other at node S
2
, and their corresponding
metrics.
The signal paths at nodes S
1
and S
2
are then extended along the two
survivor paths.
Upon receipt of r3 at t = 3T, we compute the metrics of the two paths
entering state S
1
. Suppose the survivors at t = 2T are the paths (0,0) at
S
1
and (0,1) at S
2
. Then, the two metrics for the paths entering S1 at t =
3T are

2
3 2 3
2
3 2 3
(0, 0, 0) (0, 0) ( )
(0,1,1) (0,1) ( ) 8.5.16
b
b
u u r
u u r
c
c
= + +
= + +
These two metrics are compared and the path with the larger
( distance ) metric is eliminated. Similarly, the metrics for the two
paths entering S
2
at t = 3T are

These two metrics are compared and the path with the larger
( distance ) metric is eliminated.

This process is continued as each new signal sample is received from
the demodulator. Thus, the Viterbi algorithm computes two metrics
for the two signal paths entering a node at each stage of the trellis
search and eliminates one of the two paths at each node . The two
survivor paths are then extended forward to the next stage.

2
3 2 3
2
3 2 3
(0, 0,1) (0, 0) ( )
(0,1, 0) (0,1) ( ) 8.5.17
b
b
u u r
u u r
c
c
= +
= +
8.6 System Design In The Presence Of Channel
Distortion
Recall that a signal pulse will satisfy the condition of zero ISI
at the sampling instant
if

From this condition that we conclude that for ISI free transmission
over a channel, the transmitter-receiver filters and channel
transfer function must satisfy

There are infinite number of transmitter-receiver filter pairs satisfy
the above condition.
( ) x t
, 1, 2,... t nT n = =
( )
m
m
X f T
T
=
+ =
( ) ( ) ( ) ( )
T R rc
G f C f G f X f =
System Design In The Presence Of Channel
Distortion
Two types of distortion
amplitude distortion
phase distortion
Amplitude distortion result when is not constant for
Phase distortion result when is nonlinear

envelope delay:

when is linear, is constant.

( ) C f
f W s
( )
C
f u
( ) 1
( )
2
C
d f
f
df
u
t
t
=
( )
C
f u ( ) f t
Distortion
However, when is nonlinear and the various frequency
component in the input signal undergo different delays in passing
through the channel. In such a case we say that transmitted signal
has suffered from delay distortion.

Both amplitude and delay distortion cause ISI in received signal.

( )
C
f u
Distortion

For example, let us assume that we have designed a pulse with a
raised cosine spectrum that has zero ISI at the sampling instants.
An example of such a pulse is illustrated in Figure 8.33(a).When the
pulse is passed through a channel filter with for and
a quadratic phase characteristic (linear envelop delay).
The received pulse at output of the channel is shown in Figure
8.33(b).

( ) 1 C f = f W <
Note that the periodic zero crossing have been shifted by the delay distortion.

8.6.1 Design of Transmitting and Receiving Filters
for a known channel

We assume that is known and design
that maximize the SNR at the output of the receiving filter
and result in zero ISI.

( ) C f
( ), ( )
T R
G f G f
0
( ) ( ) ( ) ( ) ,
j ft
T R rc
G f C f G f X f e f W
t
= s
( )
rc
X f

where is the desired raised cosine spectrum that
yields zero ISI at sampling instants.
: time delay
The noise at the output of the receiving filter:

2
( ) ( ) ( )
( ) ( ) ( )
R
v n R
v t n t g d
S f S f G f
t t t
=
=
}
0
t
Design of Transmitting and Receiving Filters
for a known channel
For binary PAM transmission, the sampled output of the
matched filter is
where
is normalized to unity,

Probability of error

0 m m m m m
y x a v a v = + = +
0
x
2
2
: the noise term, zero-mean Gaussian with variance
( ) ( )
m
m
v n R
a d
v
S f G f df o

=
=
}
2
2
2
2
2
1
2
v
y
d
v
d
P e dy Q
o
o
t

| |
= = |
|
\ .
}
for a known channel
Now suppose that we select the filter at the transmitter to have the
frequency response

where is a suitable delay to ensure causality.

The filter at receiver is designed to be matched to the receiver
signal pulse.

where is an appropriate delay.

( )
( )
( )
0
2
rc
j ft
T
X f
G f e
C f
t
=
0
t
( ) ( ) ( )
0
2

j ft
T rc
G f C f X f e
t
==> =
( )
2
( )
r
j ft
R rc
G f X f e
t
==> =
r
t
( ) ( )
( )
( )
( )
( )
( )
( )
( )
( )
2
2
2
2
0 0 0
2
2
2
2
1
2
1
2
0
2 2 2
.
2 .
v
W
v R rc
W
W
m
rc
av T
W
W
rc
av
W
W
rc
av
W
d
SNR
N N N
G f df X f df
E a
X f
d
P g t dt df
T T
C f
X f
d P T df
C f
X f
P T
SNR df
N
C f
o
o

=
= = =
= =
(
(
=
(

(
(
==> =
(

} }
} }
}
}
( )
( )
10
2
10log
W
rc
W
X f
df
C f
}
( )
( )
1 , in
1 ,
C f for f W loss performance
C f for f W noperformanceloss
s s =>
= s =>
We note that the term,
for a known channel
example 8.6.1 : Determine and for a binary
communication system that transmits data at a rate of 4800 bits/sec .

( )
T
G f
( )
R
G f
( )
( ) ( )
2
15 0
1
, , 4800
1
: , , , 10 /
2
n
C f f f W Hz
f
W
N
n t zero mean white Gaussian S f W Hz
= s =
| |
+
|
\ .
= =
( )
( )
( )
( ) ( )
2
2
:
1
4800
1 cos( ) cos
2 9600
, 1 cos , 4800
9600
cos , 4800
9600
0 4800
rc
T
R
T R
Solution
W
T
f
T
X f T f T
f
f
Then G f T f Hz
W
f
G f T f Hz
and G f G f for f Hz
t
t
t
t
= =
| |
= + ( =
|

\ .
(
| |
| |
= + s
( | |
\ .
( \ .

| |
= s
|
\ .
= = >
8.6.2 Channel Equalization
In practice we often encounter channel whose frequency
response are either unknown or change with time. We may
design the transmitting filter to have a square root raised
cosine frequency response ; i.e.,

and the receiving filter, with frequency response , to be
matched to .

( )
( )
0
2
,
0 ,
j ft
rc
T
X f e f W
G f
f W
t
=

>
( )
R
G f
( )
T
G f
( ) ( ) ( )
T R rc
G f G f X f ==> =
Channel Equalization
The output of the receiving filter is
( ) ( )
( ) ( ) ( ) ( )
( )

n
n
T R
y t a x t nT v t
where x t g t c t g t
=
= +
= - -
( )
0

, 0, 1, 2,..
m n n m m
n
m n n m m
n
n m
ISI
n
y a x v
x a a x v
where x x nT n
=
=
= +
= + +
= =

In any practical system, it is reasonable to assume that the ISI affect
a finite number of symbols.
Hence, we may assume for and ,where
and are finite, positive integer.
Consequently, the ISI observed at the output of the receiving filter
may viewed as being generated by passing the data sequence
through an FIR filter with coefficients , as
shown in Figure 8.35.
This filter is called the equivalent discrete-time channel filter.
0
n
x =
1
n L <
2
n L >
1
L
2
L
{ }
m
a
{ }
1 2
,
n
x L n L s s
Maximum-Likelihood Sequence Detection
The optimum detector for the information sequence based
on the observation of the received sequence is a ML
detector.
The Viterbi algorithm provides a method for searching through the
trellis for the ML signal path.
Linear Equalizer
To compensate for the channel distortion, we may employ a linear
filter with adjustable parameters. These adjustable filter are called
channel equalizers or simply equalizers
First, we consider the design characteristic for a linear equalizer
from a frequency domain viewpoint.

{ }
m
a
{ }
m
y
( ) ( ) ( )
( )
( )
( )
1

C
T R rc
j f
E
G f G f X f
G f e f W
C f
u
=
= s

In this case, the equalizer is said to be the inverse channel filter to
the channel response. Since it forces the ISI to be zero at the
sampling times , the equalizer is called zero-forcing
equalizer.

t nT =
Hence, the input to the detector is of the form
where is the noise component, which is zero-mean Gaussian
with a variance

When noise is white
m
v
m m m
y a v = +
( ) ( ) ( )
( ) ( )
( )
2 2
2
2

v n R E
W
n rc
W
S f G f G f df
S f X f
df
C f
o

=
=
}
}
( )
( )
( )
2
0 0
2
,
2 2
W
rc
n v
W
X f
N N
S f df
C f
o

= =
}

example 8.6.2 : The channel given in Example 8.6.1 is equalized by
a zero-forcing equalizer.
Assume ,determine the value of the noise
variance at the sampling instants and probability of error.

( ) ( ) ( )
T R rc
G f G f X f =

( )
( )
( )
( )
( )
2 0
2
2
2 0
1
2 2
0
0
0
2
2 2
2
:

2
1 cos
2 2
1 cos
2
2 1

3
1

3
W
rc
v
W
W
W
av T
solution
X f
N
df
C f
f
TN f
df
W W
x
N x dx
N
M d
P G f
T
o
t
t
t
=
(
| |
| |
= +
( | |
\ .
( \ .

| |
= +
|
\ .
| |
=
|
\ .
=
}
}
}
( )
( )
( )
( )
( )
( )
2 2
2 2
2
2 0
1
3
1

3
2 1
3

2 1
1
3
W W
rc
W W
av
M
M d
df X f df
T
M d
T
M
P T
P Q
M
M N
t

=
| |
|
=
|

|
\ .
} }
Let us now consider the design of a linear equalizer from a time-
domain view-point.
In practice the channel equalizer is approximated by a finite duration
impulse response (FIR) filter, or a transversal filter, with adjustable
taps coefficients , as illustrated in Figure 8.37.
{ }
n
c

In practice , is often selected as . Notice that, in this case,
the sampling rate at the output of the filter is .

where are the equalizer coefficients, and .

and is the signal pulse corresponding
to . Then the equalized output signal pulse is

t
2
T
t =
( )
R
G f
2
T
( ) ( )
2
( )
N N
j fn
E n E n
n N n N
g t c t n G f c e
t t
o t

= =
= =

{ }
n
c
( )
2 1 N+ 2 1 N L + >
( ) ( ) ( ) ( )
T R
X f G f C f G f =
( )
x t
( )
X f
( ) ( )
N
n
n N
q t c x t nt
=
=
The zero-forcing condition can now be applied to the samples of
taken at times .

where is a matrix with element
is the coefficient vector and is the
column vector with one nonzero element.
Solution

( )
q t
t mT =
( ) ( )
1 , 0
0 , 1, 2,......
N
n
n N
m
q mT c x mT n
m N
t
=
=
= =

=
==>
Xc = q
X
(2 1) (2 1) N N + + ( ) { }
, x mT nt
c
(2 1) N + q
(2 1) N +
1
= c X q

Example 8.6.3: Consider a channel distorted pulse

where is the symbol rate. The pulse is sampled at
the rate and equalized by a zero-forcing equalizer.
Determine the coefficient of a five-tap zero-forcing equalizer.

( )
2
1
2
1
x t
t
T
=
| |
+
|
\ .
1
T
2
T
( )
( )
2
2
:
1 , 0

2
0 , 1, 2
1 1 1 1 1

5 10 17 26 37
1 1 1 1
1
2 5 10 17
1 1 1 1
X 1
5 2 2 5
1 1 1 1
1
17 10 5 2
1 1 1 1 1

37 26 17 10 5
n
n
solution
m
nT
g mT c x mT
m
=
=
= =

=
(
(
(
(
(
=

2
1
0
1
2
-1
opt
0
0
c q 1
0
0
2.2
4.9
c X q 3
4.9
2.2
c
c
c
c
c
(
(
(
(
(
( (
( (
( (
( ( = =
( (
( (
( (

(
(
(
( = =
(
(
(

Equalized
Output
One drawback to the zero-forcing equalizer is that it ignores the
presence of additive noise.
Let us consider the noise-corrupted output of the FIR equalizer which
is , where is the input to the equalizer,
given by

( ) ( )
N
n
n N
z t c y t nt
=
=
( )
y t
( ) ( ) ( )
.
N
n
n N
y t a x t nT v t
=
= +
( ) ( )
( )
( )
( ) ( ) ( ) ( )
2
2
2

, :

2 8.6.33
N
n
n N
m m
N
n m
n N
N N N
n k Y k AY m
n N k N k N
z mT c y mT n
J MSE E z mT a a transmitted symbol
E c y mT n a
c c R n k c R k E a
t
t
=
=
= = =
- =
= = (

(
=
(

= +


The MMSE solution is obtained by differentiating Equation(8.6.33)

( ) ( ) ( )
( ) ( )
Y
AY m
R n k E y mT n y mT k
R k E y mT k a
t t
t
= (

= (

( ) ( )
( ) ( ) ( )
( ) ( )
1
1
0
, 0, 1, 2,...,
1

1

k
N
n Y YA
n N
K
Y
k
K
AY k
k
J
c
c R n k R k k N
R n y kT n y kT
K
R n y kT n a
K
t
t
=
=
=
c
=
c
==> = =
=
=
Adaptive Equalizer

where is a matrix, is a column vector
representing the equalizer coefficients, and is a
dimensional column vector.

( ) ( )
( ) ( )
1 , 0
Zero Forcing:
0 , 1, 2,...,
MSE: , 0, 1, 2,....,

N
n
n N
N
n Y YA
n N
m
q mT c x mT n
m N
c R n k R k k N
t
=
=
=
= =

=
= =
==>
Bc = d
B
(2 1) (2 1) N N + + c
2 1 N+
d
(2 1) N +
-1
opt
c B d =
The equalizer coefficients can be updated step-by-step.
Choose arbitrarily the initial coefficient vector .
At the (m+1)th step the filter coefficients are updated
according to

The gradient vector

is the step-size parameter for the iterative procedure
As and
as illustrated in Figure 8.38

0
c
1 m m m +
= A c c g
, 0,1, 2,...
m m
i = = g Bc d
A
0
m
g
m
opt m
c c
Let be the nth coefficient of the m-th iteration,
for
From

the nth coefficient of the gradient vector can be expressed as

The term represents the filter output at
time m.
m,n
c
n=-N,-N+1,...,N
, 0,1, 2,...
m m
i = = g Bc d
,
,
[ ( ) ( )] [ ( ) ]
( ) ( )
N
m,n m k m
k N
N
m k m
n N
g E y mT n y mT k c E y mT n a
E y mT n y mT k c a
t t t
t t
=
=
=
(
| |
=
( |
\ .

,
( )
N
m m k
n N
z y mT k c t
=
=
Define the error

The error vector denotes the difference between the desired
output from the equalizer at the mth time instant and
actual output
Define the input vector at time m as

m m m
e a z =
( )
( )
( )
( 1)
m
y mT N
y mT N
y mT N
t
t
t
+ (
(
+
(
=
(
(
(

y
[ ]
m m m
E e = g y
m
a
m
z
The algorithm for adjusting the equalizer tap coefficients may
be express as

The gradient vector can be approximated by

stochastic gradient algorithm (LMS algorithm)

m+1 m m
A c = c - g
m m m
g e = y

m+1 m m
m+1 m m m
m m m
e
g e
A

==> A
`
=
)
c = c - g
c = c + y
y
At the demodulator, the equalizer employs a known pseudo random
sequence to adjust its coefficients.
Upon initial adjustment, the adaptive equalizer switches from a
training mode to a decision-directed mode, in which case the error
signal is .

where is the output of the detector.
the step-size parameter:

where denotes the received signal-plus-noise power, which can
be estimated from the received signal.

{ }
m
a
m
e
m m
a z
. .,
m m m
i e e a z =
m
a
1
5(2 1)
R
N P
A =
+
R
P
The convergence characteristics of the stochastic gradient algorithm
is illustrated in Figure 8.40.

The block diagram of an adaptive zero-forcing equalizer is shown in
Figure 8.41.

Decision-Feedback Equalizer
The severity of the ISI is directly related to the spectral
characteristics and not necessarily to the time span of the ISI.
For example, consider the ISI resulting from two channel which are
illustrated in Figure 8.42.
In spite of the shorter ISI span , Channel B results in more severe ISI.
This is evidenced in frequency response characteristics of these
channels, which are shown in Figure 8.43.

The noise in Channel B will be enhanced much more than in Channel
A.
This implies that the performance of the linear equalized for Channel
B will be significantly poorer than that for Channel A.
This fact is borne out by the computer simulation results for the
performance of the linear equalizer for two channels, as shown in
Figure 8.44.

Hence, the basic limitation of a linear equalizer is that it performs
poorly on channels having spectral nulls.
A decision-feedback equalizer (DFE) is a nonlinear equalizer.

where , are the previously detected symbols.
The tap coefficients of the feedforward and feedback filters are
selected to optimized some desired performance measure.
For mathematical simplicity, the MSE criterion is usually applied
and a stochastic gradient algorithm is commonly used to implement
an adaptive DFE.

( ) y t
{ },
n
c
1
N
m
z
m
a
{ },
n
b
2
N
( )
1 2
1 1
N N
m n
m n n
n n
z c y mT n b a t

= =
=

m n
a
2
1, 2,..., n N =
Figure 8.46 illustrated the block diagram of an adaptive DFE whose
tap coefficients are adjusted by means of LMS stochastic gradient
algorithm.

Figure 8.47 illustrated the probability of error performance of the
DFE, for binary PAM transmission over Channel B
Although the DFE outperforms a linear equalizer, it is not the optimum
equalizer from the viewpoint of minimize the probability
of error.

The performance of the ML sequence detector is about 4.5 dB
better than that of the DFE at an error probability of .
4
10
8.7 Multicarrier Modulation And OFDM

We considered digital transmission through nonideal channels
and observed that such channels cause intersymbol interference
when the reciprocal of the system rate is significant smaller
than the time dispersion (duration of the impulse response) of
the nonideal channel.
We also observed that ISI results in some performance
degration.
An alternative approach to the design of a bandwidth-efficient
communication system in the presence of distortion is to
subdivide the available channel bandwidth into a number of
equal-bandwidth subchannels.
The bandwidth of each subchannel is sufficient narrow so that
the frequency response characteristic of the subchannels are
nearly ideal.
Thus, we creat subchannels.
Different information symbols can be transmitted
simultaneously in the subchannels.
Consequently, the data is transmitted by frequency-division
multiplexing (FDM).
With each subchannel, we associated a carrier

where is the mid-frequency in the th subchannel.
W
K
f
=
A
K
( )
sin2 , 0,1,.... 1
k k
x t f t k K t = =
k
f
k

where independent of the values of the
phase and
In this case, we have orthogonal frequency-division
multiplexing (OFDM).
With an OFDM system having subchannels, the symbol rate
on each subcarrier is reduced by a factor of N relative the
symbol rate on a single carrier system.
( ) ( )
0
sin 2 sin 2 0
T
k k j j
f t f t dt t | t | + + =
}
, 1, 2,...,
k j
n
f f n
T
= =
k
|
.
j
|
K
By selecting to be sufficiently large, the symbol interval T
can be made significantly larger than the time duration of the
channel-time dispersion.
Thus, ISI can be made arbitrarily small by selection of .
In other words, each subchannel appears to have fixed
frequency response .
The modulator and demodulator in an OFDM system can be
implemented by use of a parallel bank of filters based on the
discrete Fourier transformation (DFT).
When the number of subchannels is large, say ,the
modulator and demodulator in OFDM system are efficiently
implement by use of the fast Fourier transform algorithm(FFT)
to compute DFT.

K
K
( )
, 0,1,..., 1
k
C f k K =
25 K >
A major problem with multicarrier modulation in general and
OFDM system in particular is high peak-to-average power
ratio (PAR) that is inherent in the transmitted signal.
Large signal peaks may saturate the power amplifer at the
transmitter and, this cause intermodulation in the transmitted
signal.
8.7.1 An OFDM System Implemented via
the FFT Algorithm
In this section, we describe an OFDM system in which QAM
is used for data transmission on each of the subcarriers.

the FFT Algorithm
A serial-to-parallel buffer subdivides the information
sequences into frames of bits.
The bits in each frame are parsed into groups, where the
th group is assigned bits.
Hence

We may view the multicarier modulation as generating
indepent QAM subchannels, where the symbol rate for each
symbol is and the signal in each subchannel has a distinct
QAM constellation.

f
B
f
B
K
i
i
b
1
K
i f
i
b B
=
=
K
1
T
the FFT Algorithm
Hence, the number of signal points for the th subchannel is
.
Let us denote the complex-valued signals points corresponding
the information on the subchannels by
.
represent the values of the DFT of a OFDM signal .
Since must be real-valued signal, its -point must satisfy
the symmetry property .
Therefore, we create symbols from information
symbols by defining

i
2
i
b
i
M =
, 0,1,..., 1
k
X k K =
{ }
k
X
( )
x t
( )
x t
N k k
X X
-
=
N
2 N K =
K
( )
( )
'
0 0
0
1, 2,...., 1
Re
Im
N k k
N
X X k K
X X
X X
-
= =
=
=
the FFT Algorithm
If we denote the new sequence of symbol as
, the -point IDFT yields the real-valued sequence.

where is simply a scale factor.
This sequence corresponds to samples of .

{ }
'
, 0,1,... 1
k
X k N =
N
1
2
'
0
1
, 0,1, , , , 1
N
k
j n
N
n k
k
x X e n N
N
t
=
= =
1
N
{ }
, 0 1
n
x n N s s
( )
x t
( )
( )
1
2
'
0
1
0
: signal duration
N
t
j k
T
k
k
n
x t X e t T
N
nT
T x x
N
t
=
= s s
=

the FFT Algorithm
The signal samples generated by computing the IDFT are
passed through a digital-to-analog(D/A) convert, where output,
ideally, is the OFDM signal waveform .
The channel output at the receiver may be express

Since the bandwidth of each subchannel is selected to be very
small relative to the overall channel ,the symbol
duration is large compared to the duration of the
channel impulse response.
{ }
n
x
( )
x t
( ) ( ) ( ) ( )
( )
: the impluse response of the channel
: convolution
r t x t c t n t
c t
= - +
-
W K f = A
1
T
f
=
A
the FFT Algorithm
Suppose that channel impulse response spans signal
samples where .
A simple way to completely avoid ISI is to insert a time guard
of duration between transmission of successive data
blocks.
This allows the response of the channel to die out before the
next block symbols are transmitted.
An alternative method to avoid ISI is to append a so-called
cyclic prefix to each block of signal samples
1 m+
m N
T
m
N
N
{ }
, 0 1
n
x n N s s
the FFT Algorithm
The cyclic prefix for the block of samples consists of samples
.
These samples are appended to beginning of the block, thus
creating a signal sequence of length samples, which
may be indexed from to , where the first
samples constitute the cyclic prefix.
Then, if the samples values of the channel response are
, the convolution of with
produce the received signal .

1 1
, ,....,
N m N m N
x x x
+
N m +
n m =
1 n N =
m
{ }
, 0
n
c n m s s
{ }
n
c
{ }
, 1
n
x m n N s s
{ }
n
r
the FFT Algorithm
Since the ISI in any pair of successive signal transmission
blocks affect the first m signal samples, we discard the first
samples of and demodulated the signal based the
received signal samples .
The channel frequency response at the subcarrier frequency
is

Since the ISI is eliminated by the use of either the cyclic prefix
or the time guard band, the demodulated sequence of symbols
may be expressed as

m
{ }
n
r
{ }
, 0 1
n
r n N s s
k
k
f
T
=
2
0
2
, 0,1, 2,...,
m
k
j n
N
k n
n
k
C C C e k N
N
t t
=
| |
= = =
|
\ .

{ }
'
, 0,1,.., 1
: the additive noise
k k k k
k
X C X k N q
q
= + =
the FFT Algorithm

If the channel characters vary slowly with time, the time
variations can be tracked by using the decisions at the output
of the detector in a decision-directed manner.
The transmission rate on each of the suncarriers can be
optimized by properly allocating the average transmitted
power and number of bits that are transmitted by each
subcarrier.
the FFT Algorithm
The per subchannel may be defined as

where is the symbol duration, is the average transmitted
power allocated to the subchannel, is the squared
magnitude of the frequency response of the th sunchannel,
and is the corresponding noise variance.

SNR
2
2
k k
k
nk
TP C
SNR
o
=
T
k
P
k
C
k
2
nk
o
the FFT Algorithm
In subchannels with high ,we transmit more bits/symbol
by using a larger QAM constellation compared to subchannels
with low
Thus, the bit rate on each subchannel can be optimized in such
a way that the error-rate performance among the subchannel is
equalized to satisfy the desired specifications.
SNR
. SNR

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Chapter 8

Digital Transmission through Bandlimited

DIGITAL TRANSMISSION THROUGH BANDLIMITED CHANNELS

Digital PAM Transmission through Bandlimited Baseband Channel

Digital transmission through Bandlimited Bandpass Channels

Digital transmission through Bandlimited Bandpass Channels

8.2.1 The Power Spectrum of the baseband signal

8.2.1 The Power Spectrum of the baseband signal

8.2.1 The Power Spectrum of the baseband signal

8.2.1 The Power Spectrum of the baseband signal

8.3 SIGNAL DESIGN FOR BANDLIMITED CHANNELS

8.3 SIGNAL DESIGN FOR BANDLIMITED CHANNELS

8.3.1 Design of Bandlimited Signals for Zero ISIThe Nyquist Criterion

8.3.1 Design of Bandlimited Signals for Zero ISIThe Nyquist Criterion

8.3.1 Design of Bandlimited Signals for Zero ISIThe Nyquist Criterion

8.3.1 Design of Bandlimited Signals for Zero ISIThe Nyquist Criterion

8.3.1 Design of Bandlimited Signals for Zero ISIThe

8.3.1 Design of Bandlimited Signals for Zero ISIThe Nyquist Criterion

8.3.2 Design of Bandlimited Signals with Controlled ISI Partial

8.3.2 Design of Bandlimited Signals with Controlled ISI Partial

8.4 PROBABILITY OF ERROR IN DETECTION OF DIGITAL PAM

8.4 .2 Symbol-by Symbol Detection of Data with Controlled ISI

8.4 .2 Symbol-by Symbol Detection of Data with Controlled ISI

8.5.1 Modulation Codes and Modulation Signals with Memory

8.5.2 The Maximum-Likelihood Sequence Detector

Define the error

8.7 Multicarrier Modulation And OFDM

8.7.1 An OFDM System Implemented via

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