DC 04 Signal&System

Wireless Information Transmission System Lab.
Chapter 4 Characterization of Communication Signals and Systems
Reference: Chapter 6, Communication Systems, 4th Edition. by Simon Haykin Institute of Communications Engineering National Sun Yat-sen University
Table of Contents
4.1 Representation of Band-Pass Signals and Systems
4.1.1 Representation of Band-Pass Signals 4.1.2 Representation of Linear Band-Pass Systems 4.1.3 Response of a Band-Pass System to a Band-Pass Signal 4.1.4 Representation of Band-Pass Stationary Stochastic Processes
4.2 Signal Space Representations

4.2.1 Vector Space Concepts 4.2.2 Signal Space Concepts 4.2.3 Orthogonal Expansions of Signals
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Table of Contents
4.3 Representation of Digitally Modulated Signals
4.3.1 Memoryless Modulation Methods 4.3.2 Linear Modulation with Memory 4.3.3 Non-linear Modulation Methods with Memory CPFSK and CPM
4.4 Spectral Characteristics of Digitally Modulated Signals

4.4.1 Power Spectra of Linearly Modulated Signals 4.4.2 Power Spectra of CPFSK and CPM Signals (*) 4.4.3 Power Spectra of Modulated Signals with Memory (*)
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4.1 Representation of Band-Pass Signals and Systems The channel over which the signal is transmitted is limited in bandwidth to an interval of frequencies centered about the carrier. Signals and channels (systems) that satisfy the condition that their bandwidth is much smaller than the carrier frequency are termed narrowband band-pass signals and channels (systems). With no loss of generality and for mathematical convenience, it is desirable to reduce all band-pass signals and channels to equivalent low-pass signals and channels.
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4.1.1 Representation of Band-Pass Signals

Suppose that a real-valued signal s(t) has a frequency content concentrated in a narrow band of frequencies in the vicinity of a frequency fc, as shown in the following figure:
Spectrum of a band-pass signal.
Our object is to develop a mathematical representation of such signals.

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A signal that contains only the positive frequencies in s(t) may be expressed as: S + ( f ) = 2u ( f ) S ( f )
s+ ( t ) = S + ( f ) e j 2 ft df
1 = F 1 2 u f F ( ) S ( f )
where S( f ) is the Fourier transform of s(t) and u( f ) is the unit step function, and the signal s+(t) is called the analytic signal or the pre-envelope of s(t). j 1 F 2u ( f ) = (t ) + t j 1 s+ ( t ) = ( t ) + s ( t ) = s ( t ) + j s ( t ) s ( t ) + js ( t ) t t
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Define:
1 1 s (t ) = s (t ) = t
s ( ) t
A filter, called a Hilbert transformer, is defined as: 1 <t < h (t ) = , t The signal s (t ) may be viewed as the output of the Hilbert transformer when excited by the input signal s(t). The frequency response of this filter is: j ( f > 0 ) 1 1 j 2 ft j 2 ft H ( f ) = h (t ) e dt = e dt = 0 ( f = 0) t ( f < 0) j
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We observe that |H( f )|=1 and the phase response ( f )=-/2 for f >0 and ( f )=/2 for f <0. Thus, this filter is basically a 90 degrees phase shifter for all frequencies in the input signal. The analytic signal s+(t) is a band-pass signal. To obtain an equivalent low-pass representation, we define: Sl ( f ) = S + ( f + f c )
sl ( t ) = s+ ( t ) e = s ( t ) + j s ( t ) e j 2 f c t In general, sl(t) is complex-valued: j 2 f t + = s t j s t s t e ( ) ( ) ( ) l sl ( t ) = x ( t ) + jy ( t ) j 2 f c t
c
s ( t ) = x ( t ) cos 2 f c t y ( t ) sin 2 f c t
s ( t ) = x ( t ) sin 2 f c t + y ( t ) cos 2 f c t
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s(t)=x(t)cos2fct-y(t)sin2fct is the desired form for the representation of a band-pass signal. The low-frequency signal components x(t) and y(t) may be viewed as amplitude modulations impressed on the carrier components cos2fct and sin2fct, respectively. x(t) and y(t) are called the quadrature components of the band j 2 f t pass signal s(t). s t j s t s t e + = ( ) ( ) ( ) l s(t) can also be written as: j 2 f c t j 2 f c t + = s ( t ) = Re x t jy t e Re s t e ( ) ( ) ( ) l
c
The low pass signal sl(t) is usually called the complex envelope of the real signal s(t) and is basically the equivalent low-pass signal.
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sl(t) can be also be written as: sl ( t ) = a ( t ) e j ( t )
where a ( t ) = x ( t ) + y ( t ) and ( t ) = tan
2 2 1
s(t) can be represented as: j j 2 f c t 2 f c t + ( t ) s ( t ) = Re s t e Re a t e = ( ) ( ) l = a ( t ) cos 2 f c t + ( t ) a(t) is called the envelope of s(t), and (t) is called the phase of s(t).
x (t )
y (t )
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Three equivalent representations of band-pass signals: s ( t ) = x ( t ) cos 2 f c t y ( t ) sin 2 f c t
j 2 f c t = Re s t e ( ) l
= a ( t ) cos 2 f c t + ( t )
The Fourier transform of s(t) is:

S ( f ) = s (t ) e
j 2 ft
dt =
1 Re ( ) = + 2
j 2 f c t j 2 ft Re s t e e dt ( ) l
1 j 2 f c t j 2 f c t j 2 ft S( f )= s t e s t e e dt + ( ) ( ) l l 2 1 S f f S = + ( ) l c l ( f fc ) 2
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The energy in the signal s(t) is defined as:
= s ( t ) dt =
2
Re sl ( t ) e
j 2 f c t
dt
R e (
)=
1 + 2
j ( t ) 2 j 4 f t 1 2 j 4 f c t * * s t = a t e ( ) ( ) c l = sl e dt + 2 sl sl + sl e 4 2 ( j 4 f t + 2 ( t ) ) 2 1 1 2 j 4 f c t + 2 ( t ) * c dt = sl ( t ) dt + a (t )e + a (t ) e 2 4 2 2 1 1 = sl ( t ) dt + sl ( t ) cos dt 4 f c t + 2 ( t ) 2 2
( )
where sl ( t ) = a (t ) = a ( t )
2 2 *
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Since the signal s(t) is narrow-band, the real envelope a(t)=|sl(t)| or, equivalently, a2(t) varies slowly relative to the rapid variations exhibited by the cosine function.
The net area contributed by the second integral is very small relative to the value of the first integral, hence, it can be 2 1 neglected. = sl ( t ) dt (4.1-24) 2
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4.1.2 Representation of Linear BandPass Systems

A linear filter or system may be described either by its impulse response h(t) or by its frequency response H( f ), which is the Fourier transform of h(t). Since h(t) is real, H*(-f )=H( f ), * 2 j f t ( ) because: H * ( f ) = h (t ) e dt
= h* ( t ) e j 2 ft dt

Note that h ( t ) is real.
= h ( t ) e j 2 ft dt = H ( f
H ( f ) Define H l ( f f c ) = 0
* l
(f (f
> 0) < 0)
0 then H ( f f c ) = * H ( f ) = H ( f )
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(f (f
> 0) < 0)
4.1.2 Representation of Linear BandPass Systems

As a result H ( f ) = H l ( f f c ) + H l* ( f f c ) thus h ( t ) = hl ( t ) e j 2 fc t + hl*e j 2 fc t
j 2 f c t = 2 Re h t e ( ) l
where
H l* ( f f c ) e j 2 ft df using x = f f c
= H l* ( x ) e j 2 xt dx e j 2 fc t
Hl ( x) e
j 2 xt
dx e j 2 fc t = hl* ( t ) e j 2 fc t
In general, the impulse response hl(t) of the equivalent low-pass system is complex-valued.
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4.1.3 Response of a Band-Pass System to a Band-Pass Signal

We have shown in Sections 4.1.1 and 4.1.2 that narrow band band-pass signals and systems can be represented by equivalent low-pass signals and systems. We demonstrate in this section that the output of a band-pass system to a band-pass input signal is simply obtained from the equivalent low-pass input signal and the equivalent low-pass impulse response of the system. The output of the band-pass system is also a band-pass signal, and, therefore, it can be expressed in the form: r ( t ) = s ( ) h ( t ) d where r(t) is related to the input signal s(t) and the impulse response h(t) by the convolution integral. j 2 f c t r ( t ) = Re r t e l( )
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4.1.3 Response of a Band-Pass System to a Band-Pass Signal

The output of the system in the frequency domain is:
R( f ) = S ( f )H ( f =
From 4.1-21 and 4.1-28.
For a narrow band signal, Sl( f-fc)0 for f <0 and H*l(-f-fc)=0 for f >0.
1 S f f + S f f H f f + H ( ) ( ) ( ) l c l c l c l ( f fc ) 2
Sl ( f f c ) H l ( f f c ) = 0
4.1-27
Sl ( f f c ) H l ( f f c ) = 0 4.1-26 1 R( f ) = S f f H f f S f f H + ( ) ( ) ( ) l c l c l c l ( f fc ) 2 Rl ( f ) Sl ( f ) H l ( f ) 1 Compared R f f R f f = + ( ) ( ) l c l c r t = with 4.1-21 l ( ) sl ( ) hl ( t ) d 2

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For a narrow band signal, S*l(-f-fc)0 for f >0 and Hl( f-fc)=0 for f <0.
4.1.4 Representation of Band-Pass Stationary Stochastic Processes

In this section, we extend the representation to sample functions of a band-pass stationary stochastic process. In particular, we derive the relations between the correlation functions and power spectra of the band-pass signal and the correlation function and power spectra of the equivalent lowpass signal. Suppose that n(t) is a sample function of a wide-sense stationary stochastic process with zero mean and power spectral density nn(f ). The power spectral density is assumed to be zero outside of an interval of frequencies centered around fc, where fc is termed the carrier frequency. The stochastic process n(t) is said to be a narrowband band-pass process if the width of the spectral density is much smaller than fc.
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Under this condition, a sample function of the process n(t) can be represented by the following equations from Section 4.1.1: n ( t ) = a ( t ) cos 2 f c t + ( t )
= x ( t ) cos 2 f c t y ( t ) sin 2 f c t
j 2 f c t z t e = Re ( )
a(t) is the envelope and (t) is the phase of the real-valued signal. x(t) and y(t) are the quadrature components of n(t). z(t) is called the complex envelope of n(t). If n(t) is zero mean, then x(t) and y(t) must also have zero mean values. The stationarity of n(t) implies that:
xx ( ) = yy ( )
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xy ( ) = yx ( )
Proved next.
4.1.4 Representation of Band-Pass Stationary Stochastic Processes Proof of xx ( ) = yy ( ) and xy ( ) = yx ( )

Autocorrelation function of n(t) is: nn ( ) = E n ( t ) n ( t + )
=E x ( t ) cos 2 f c t y ( t ) sin 2 f c t
x ( t + ) cos 2 f c ( t + ) y ( t + ) sin 2 f c ( t + )
= xx ( ) cos 2 f c t cos 2 f c ( t + ) + yy ( ) sin 2 f c t sin 2 f c ( t + ) xy ( ) sin 2 f c t cos 2 f c ( t + ) yx ( ) cos 2 f c t sin 2 f c ( t + )

1 cos ( A B ) + cos ( A + B ) by using: cos A cos B = 2 sin A sin B = 1 cos ( A B ) cos ( A + B ) 2 sin A cos B = 1 sin A B ) + sin ( A + B ) 2 (
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We can obtain: nn ( ) = E n ( t ) n ( t + )
1 1 f xx ( ) yy ( ) cos 2 cos 2 f c ( 2t + ) + + xx ( ) yy ( ) c 2 2 1 1 f sin 2 sin 2 f c ( 2t + ) yx ( ) + xy ( ) yx ( ) xy ( ) c 2 2 Since n(t) is stationary, the right-hand side must be independent of t. As a result, xx ( ) = yy ( ) and xy ( ) = yx ( ) Q.E.D. Therefore, nn ( ) = xx ( ) cos 2 f c yx ( ) sin 2 f c Note that this equation is identical in form to: =
n ( t ) = x ( t ) cos 2 f c t y ( t ) sin 2 f c t
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The autocorrelation function of the equivalent low-pass process z(t)=x(t)+jy(t) is defined as: 1 zz ( ) = E z ( t ) z ( t + ) 2 1 zz ( ) = xx ( ) + yy ( ) j xy ( ) + j yx ( ) 2 Since xx()=yy() and xy()= -yx() we obtain: zz()=xx()+ jyx() This equation relates the autocorrelation function of the complex envelope to the autocorrelation and cross-correlation functions of the quadrature components.
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By combining zz ( ) = xx ( ) + j yx ( ) and
nn ( ) = xx ( ) cos 2 f c yx ( ) sin 2 f c
we can obtain: nn()=Re[zz()e j2fc] Therefore, the autocorrelation function nn() of the bandpass stochastic process is uniquely determined from the autocorrelation function zz() of the equivalent low-pass process z(t) and the carrier frequency fc. The power density spectrum of the stochastic process n(t) is:
nn ( f ) = =

j 2 f c j 2 f Re e e d ( ) zz
1 zz ( f f c ) + zz ( f f c ) 2
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Properties of the quadrature components
yx ( ) = xy ( ) xy ( ) = yx ( )
(2.2-10) (4.1-41)
zz ( ) = xx ( ) + yy ( ) j xy ( ) + j yx ( )
1 2
xy ( ) = xy ( )
xy ( ) is an odd function of and xy ( 0 ) = 0. If xy ( ) = 0 for all , then zz ( ) is real (from 4.1-48) and
the power spectral density satisfies zz ( f ) = zz ( f (i.e. zz ( f
is symmetric about f = 0).
If n(t) is Gaussian, x(t) and y(t +) are jointly Gaussian. For =0, they are statistically independent, and the joint PDF is: 2 2 2 ( x + y ) 2 2 1 p ( x, y ) = e = xx ( 0 ) = yy ( 0 ) = nn ( 0 ) , 2 2
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Representation of white noise
White noise is a stochastic process that is defined to have a flat (constant) power spectral density over the entire frequency range. This type of noise cant be expressed in terms of quadrature components, as a result of its wideband character. In the demodulation of narrowband signals in noise, it is mathematically convenient to model the additive noise process as white and to represent the noise in terms of quadrature components. This can be accomplished by postulating that the signals and noise at the receiving terminal have passed through an ideal band-pass filter.
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Representation of white noise (cont.)
The noise resulting from passing the white noise process through a spectrally band-pass filter is termed band-pass white noise and has the power spectral density:
The band-pass white noise can be represented by:

n ( t ) = a ( t ) cos 2 f c t + ( t )
j 2 f c t = Re z t e ( )
= x ( t ) cos 2 f c t y ( t ) sin 2 f c t
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Representation of white noise (cont.)
The equivalent low-pass noise z(t) has a power spectral 1 density: N f B 0 2 zz ( f ) = 1 0 f > B 2 sin B and zz ( ) = N 0 ( ) zz ( ) = N 0
4.1-54
The power spectral density for white noise and band-pass white noise is symmetric about f =0, so yx()=0 for all . zz ( ) = xx ( ) = yy ( ) (from 4.1-48)
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4.2 Signal Space Representations

We will demonstrate that signals have characteristics that are similar to vectors and develop a vector representation for signal waveforms. 4.2.1 Vector Space Concepts A vector v in an n-dimensional space is characterized by its n components [v1 v2 vn] and may also be represented as a linear combination of unit vectors or basis vectors ei, 1in,
= vi ei
i =1 n
The inner product of two n-dimensional vectors is defined as:

1 2 = v1i v2 i
i =1 n
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4.2.1 Vector Space Concepts

A set of m vectors vk, 1km are orthogonal if: i j = 0 for all 1 i, j m, and i j. The norm of a vector v is denoted by ||v|| and is defined as:
= ( )
12
2 v i i =1
A set of m vectors is said to be orthonormal if the vectors are orthogonal and each vector has a unit norm. A set of m vectors is said to be linearly independent if no one vector can be represented as a linear combination of the remaining vectors. Two n-dimensional vectors v1 and v2 satisfy the triangle inequality: 1 + 2 1 + 2
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Cauchy-Schwarz inequality: 1 2 1 + 2 The norm square of the sum of two vectors may be expressed as: 2 2 2 1 + 2 = 1 + 2 + 2 1 2 Linear transformation in an n-dimensional vector space:
' = A In the special case where v=v, A = the vector v is called an eigenvector and is the corresponding eigenvalue.
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Gram-Schmidt procedure for constructing a set of orthonormal vectors.
Arbitrarily selecting a vector v1 and normalizing its length: 1 u1 = 1 Select v2 and subtract the projection of v2 onto u1. u '2 = 2 ( 2 u1 ) u1 ' u Normalize the vector u2 to unit length. u 2 = 2 u '2 ' u ' Selecting v3: u3 = 3 - ( v 3 u1 ) u1 - ( v 3 u 2 ) u 2 u3 = 3 ' u3 By continuing this procedure, we construct a set of orthonormal vectors.
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4.2.2 Signal Space Concepts

The inner product of two generally complex-valued signals x1(t) and x2(t) is denote by < x1(t),x2(t)> and defined as:
x1 ( t ) ,x2 ( t ) = x1 ( t ) x2 ( t ) dt a b
The signals are orthogonal if their inner product is zero. The norm of a signal is defined as:
x (t ) =
x ( t ) dt
2
12
A set of m signals are orthonormal if they are orthogonal and their norms are all unity. A set of m signals is linearly independent, if no signal can be represented as a linear combination of the remaining signals.
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4.2.2 Signal Space Concepts

The triangle inequality for two signals is:
x1 ( t ) + x2 ( t ) x1 ( t ) + x2 ( t )
The Cauchy-Schwarz inequality is:
x ( t ) x ( t ) dt
a 1 2
x1 ( t ) dt
2
12
x2 ( t ) dt
2
12
with equality when x2(t)=ax1(t), where a is any complex number.
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4.2.3 Orthogonal Expansions of Signals

Suppose that s(t) is a deterministic, real-valued signal with finite energy: 2 s = s ( t ) dt
Suppose that there exists a set of functions {fn(t), n=1,2,,K} that are orthonormal in the sense that: 0 ( m n ) f n ( t ) f m ( t ) dt = 1 ( m = n ) We may approximate the signal s(t) by a weighted linear combination of these functions, i.e.,
s ( t ) = sk f k ( t )
k =1 K
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The approximation error incurred is:
e (t ) = s (t ) s (t ) The energy of the approximation error:

k =1 To minimize the energy of the approximation error, the optimum coefficients in the series expansion of s(t) may be found by:

e = s t s t dt s t s f t = ( ) k k ( ) dt ( ) ( )
2 K
(4.2-24)
Differentiating Equation 4.2-24 with respect to each of the coefficients {sk} and setting the first derivatives to zero. Use a well-known result from estimation theory based on the meansquare-error criterion, which is that the minimum of e with respect to the {sk} is obtained when the error is orthogonal to each of the functions in the series expansion.
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Using the second approach, we have:
K s ( t ) sk f k ( t ) f n ( t ) dt = 0, n = 1, 2,..., K k 1 Since the functions {fn(t)} are orthonormal, we have:
sn = s ( t ) f n ( t ) dt , n = 1, 2,..., K
Thus, the coefficients are obtained by projecting the signals s(t) onto each of the functions. The minimum mean square approximation error is:
min = e ( t ) s ( t ) dt = s ( t ) dt
2 2 = s sk k =1 K
s f ( t ) s ( t )dt
k =1 k k
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When the minimum mean square approximation error min=0,
s = s = s ( t ) dt
k =1 2 k 2
Under such condition, we may express s(t) as:
s ( t ) = sk f k ( t )
k =1
When every finite energy signal can be represented by a series expansion of the form for which min=0, the set of orthonormal functions {fn(t)} is said to be complete.
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Example 4.2-1 Trigonometric Fourier Series: Consider a finite energy signal s(t) that is zero everywhere except in the range 0 t T and has a finite number of discontinuities in this interval. Its periodic extension can be represented in a Fourier series as: 2 kt 2 kt s ( t ) = ak cos + bk sin T T k =0 where the coefficients {ak, bk} that minimize the mean square error are given by: 1 T 2 kt 1 T 2 kt ak = s ( t ) cos dt , bk = s ( t ) sin dt 0 0 T T T T
The set of trigonometric functions
2 / T cos 2 kt / T , 2 / T sin 2 kt / T
is complete, and the series expansion results in zero mean square error.
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Gram-Schmidt procedure
Constructing a set of orthonormal waveforms from a set of finite energy signal waveforms {si(t), i=1,2,,M}. Begin with the first waveform s1(t) which has energy 1. The first orthonormal waveform is: s1 ( t ) f1 ( t ) =
The 2nd waveform is constructed from s2(t) by first computing the projection of f1(t) onto s2(t):
c12 = s2 ( t ) f1 ( t ) dt
Then c12 f1(t) is subtracted from s2(t): f 2' ( t ) = s2 ( t ) c12 f1 ( t )

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Gram-Schmidt procedure (cont.)
If 2 denotes the energy of f2(t), the normalized waveform that is orthogonal to f1(t) is: f 2' ( t ) f2 (t ) = In general, the orthogonalization of the kth function leads to k 1 f k' ( t ) ' fk (t ) = where f k ( t ) = sk (T ) cik f i ( t )
i =1
cik = sk ( t ) f i ( t ) dt ,
i = 1, 2,..., k 1
The orthogonalization process is continued until all the M signal waveforms have been exhausted and N M orthonormal waveforms have been constructed.
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Once we have constructed the set of orthonormal waveforms {fn(t)}, we can express the M signals {sn(t)} as linear combinations of the {fn(t)}:
sk ( t ) = skn f n ( t ),
n =1 2
k = 1, 2,..., M
N 2
2 k = s t dt s = ( ) kn = sk k n =1
sk = [ sk1 sk 2 ... skN ]

Each signal may be represented as a point in the Ndimensional signal space with coordinates {ski, i=1,2,,N}.
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The energy in the kth signal is simply the square of the length of the vector or, equivalently, the square of the Euclidean distance from the origin to the point in the Ndimensional space. Any signal can be represented geometrically as a point in the signal space spanned by the {fn(t)}. The functions {fn(t)} obtained from the Gram-Schmidt procedure are not unique. If we alter the order in which the orthogonalization of the signals {sn(t)} is performed, the orthonormal waveforms will be different. Nevertheless, the vectors {sn(t)} will retain their geometrical configuration and their lengths will be invariant to the choice of orthonormal functions {fn(t)}.
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Consider the case in which the signal waveforms are band-pass and represented as: j 2 f c t sm ( t ) = Re s t e ( ) lm , m = 1, 2,..., M
2 1 m = s ( t ) dt = slm ( t ) dt (from 4.1-24) 2 Similarity between any pair of signal waveforms is measured by the normalized cross correlation: 1 1 s t s t dt s t s t dt Re = ( ) k( ) ( ) lk ( ) (4.2-44) m lm m k 2 m k Complex-valued cross-correlation coefficient km is defined as: 1 km = s ( t ) slk ( t ) dt (4.2-45) lm 2 m k 2 m
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Re ( km ) = 1
m k sm sk sm sk Re ( km ) = = sm sk m k
sm ( t ) sk ( t ) dt
From 4.2-44 & 4.2-45.
The Euclidean distance between a pair of signals is defined as:
d km = sm sk =
(e)
sm ( t ) sk ( t ) dt
2
12
= m + k 2 m k Re ( km )
12
When m=k=for all m and k, this expression simplifies to:
d km = 2 1 Re ( km )
(e)
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12

Digitally modulated signals, which are classified as linear, are conveniently expanded in terms of two orthonormal basis functions of the form:
2 2 cos 2 f c t sin 2 f c t f1 ( t ) = f2 (t ) = T T If slm(t) is expressed as slm(t)=xl(t)+jyl(t), sm(t) may be expressed as: sm ( t ) = xl ( t ) f1 ( t ) + yl ( t ) f 2 ( t ) From 4.2-42.
In the transmission of digital information over a communication channel, the modulator is the interface device that maps the digital information into analog waveforms that match the characteristics of the channel.
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The mapping is generally performed by taking blocks of k=log2M binary digits at a time from the information sequence {an} and selecting one of M=2k deterministic, finite energy waveforms {sm(t), m=1,2,,M} for transmission over the channel. When the mapping is performed under the constraint that a waveform transmitted in any time interval depends on one or more previously transmitted waveforms, the modulator is said to have memory. Otherwise, the modulator is called memoryless. Functional model of passband data transmission system
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The digital data transmits over a band-pass channel that can be linear or nonlinear. This mode of data transmission relies on the use of a sinusoidal carrier wave modulated by the data stream. In digital passband transmission, the incoming data stream is modulated onto a carrier (usually sinusoidal) with fixed frequency limits imposed by a band-pass channel of interest. The modulation process making the transmission possible involves switching (keying) the amplitude, frequency, or phase of a sinusoidal carrier in some fashion in accordance with the incoming data. There are three basic signaling schemes: amplitude-shift keying (ASK), frequency-shift Keying (FSK), and phase-shift keying (PSK).
47

Illustrative waveforms for the three basic forms of signaling binary information. (a) ASK (b) PSK (c) FSK.
48

Unlike ASK signals, both PSK and FSK signals have a constant envelope. This property makes PSK and FSK signals impervious to amplitude nonlinearities. In practice, we find that PSK and FSK signals are preferred to ASK signals for passband data transmission over nonlinear channels. Digital modulation techniques may be classified into coherent and noncoherent techniques, depending on whether the receiver is equipped with a phase-recovery circuit or not. The phase-recovery circuit ensures that the oscillator supplying the locally generated carrier wave in the receiver is synchronized (in both frequency and phase) to the oscillator supplying the carrier wave used to originally modulated the incoming data stream in the transmitter.
49

M-ary signaling scheme:
For almost all applications, the number of possible signals M=2n. Symbol duration T=nTb, where Tb is the bit duration. We have M-ary ASK, M-ary PSK, and M-ary FSK. We can also combine different methods of modulation into a hybrid form. For example, M-ary amplitude-phase keying (APK) and M-ary quadrature-amplitude modulation (QAM). M-ary PSK and M-ary QAM are examples of linear modulation. An M-ary PSK signal has a constant envelope, whereas an Mary QAM signal involves changes in the carrier amplitude.
50

M-ary signaling scheme (cont.):
M-ary PSK can be used to transmit digital data over a nonlinear band-pass channel, whereas M-ary QAM requires the use of a linear channel. M-ary PSK, and M-ary QAM are commonly used in coherent systems. ASK and FSK lend themselves naturally to use in noncoherent systems whenever it is impractical to maintain carrier phase synchronization. We cant have noncoherent PSK.
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4.3.1 Memoryless Modulation Methods

Pulse-amplitude-modulated (PAM) signals
Double-sideband (DSB) signal waveform may be represented as: j 2 f c t sm ( t ) = Re A g t e ( ) m
Bandpass Signal
= Am g ( t ) cos 2 f c t , m = 1, 2,..., M , 0 t T
where Am denote the set of M possible amplitudes corresponding to M=2k possible k-bit blocks of symbols. The signal amplitudes Am take the discrete values: Am = ( 2 m 1 M ) d , m = 1, 2,..., M ( ( M 1) d ... ( M 1) d ) 2d is the distance between adjacent signal amplitudes. g(t) is a real-valued signal pulse whose shape influences the spectrum of the transmitted signal. The symbol rate is R/k, Tb=1/R is the bit interval, and T=k/R=kTb is the symbol interval.
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Pulse-amplitude-modulated (PAM) signals (cont.)
The M PAM signals have energies: T 1 2 T 2 1 2 2 m = sm ( t ) dt = Am g ( t ) dt = Am g 0 0 2 2 These signals are one-dimensional and are represented by: sm ( t ) = sm f ( t ) f(t) is defined as the unit-energy signal waveform given as: 2 f (t ) = g (t ) cos 2f c t
1 sm = Am g , m = 1,2,..., M 2 Digital PAM is also called amplitude-shift keying (ASK).

53

Signal space diagram for digital PAM signals:
54

Gray encoding: The mapping of k information bits to the M=2k possible signal amplitudes may be done in a number of ways. The preferred assignment is one in which the adjacent signal amplitudes differ by one binary digit. The Euclidean distance between any pair of signal points is:
d mn =
(e)
( sm sn )
1 g Am An = d 2 g m n 2
The minimum Euclidean distance between any pair of signals is: (e) d min = d 2 g
55

Single Sideband (SSB) PAM is represented by: 4.1-9 j 2 f c t sm ( t ) = Re Am g ( t ) j g ( t ) e , m = 1, 2,..., M where g (t ) is the Hilbert transform of g(t). The digital PAM signal is also appropriate for transmission over a channel that does not require carrier modulation and is called baseband signal: sm ( t ) = Am g ( t ) , m = 1, 2,..., M If M=2, the signals are called antipodal and have the special property that: s1 ( t ) = s2 ( t )
56

Four-amplitude level baseband and band-pass PAM signals
57

Phase-modulated signals (Binary Phase-Shift Keying)
In a coherent binary PSK system, the pair of signals s1(t) and s2(t) used to represent binary symbols 1 and 0, respectively is defined by
s1 ( t ) = s2 ( t ) = 2 Eb cos ( 2 f c t ) Tb 2 Eb 2 Eb cos ( 2 f c t + ) = cos ( 2 f c t ) Tb Tb
where 0tTb and Eb is the transmitted signal energy per bit. A pair of sinusoidal waves that differ only in a relative phaseshift of 180 degrees are referred to as antipodal signals.
58

To ensure that each transmitted bit contains an integral number of cycles of the carrier wave, the carrier frequency fc is chosen equal to nc/Tb for some fixed integer nc. In the case of binary PSK, there is only one basis function of unit energy: 2 1 (t ) = cos (2f c t ), 0 t < Tb Tb
s1 ( t ) = Eb 1 ( t ) , s2 ( t ) = Eb 1 ( t ) , 0 t < Tb
The coordinates of the message points are: s11 = s1 ( t )1 ( t ) dt = + Eb

0 Tb
s21 = s2 ( t ) 1 ( t ) dt = Eb
0
Tb
59

60

Phase-modulated signals (Quadriphase-Shift Keying)
Quadriphase-Shift Keying (QPSK)
2E + cos 2 f t 2 i 1 , ( ) c si ( t ) = T 4 0, 0t T elsewhere
where i = 1, 2, 3, 4; E is the transmitted signal energy per symbol, and T is the symbol duration. Equivalently
2E 2E si ( t ) = cos ( 2i 1) cos ( 2 f ct ) sin ( 2i 1) sin ( 2 f ct ) T 4 T 4
61

Defined a pair of quadrature carriers:
1 (t ) =
2 cos (2f c t ), T
0tT
2 2 (t ) = sin (2f c t ), 0tT T There are four message points, and the associated signal vectors are defined by ( ) E cos 2 i 1 4 , i = 1,2,3,4 si (t ) = E sin (2i 1) 4
62

Each possible value of the phase corresponds to a unique dibit. For example, we may choose the Gray coding.
63

Signal space diagram of coherent QPSK system
64

65

Phase-modulated signals (M-ary PSK)
The M signal waveforms are represented as:
sm ( t )
j 2 ( m 1) M j 2 f c t , m = 1, 2,..., M , 0 t T g t e e = Re ( )
2 1 = g ( t ) cos 2 f c t + m ( ) M 2 2 1 cos 2 sin m f t g t = g ( t ) cos ( ) () ( m 1) sin 2 f c t c M M
Digital phase modulation is usually called phase-shift keying (PSK).

66

Signal space diagram for octaphase shift keying (i.e., M=8)
67

The signal waveforms have equal energy: T 1 T 2 1 2 = sm ( t ) dt = g ( t ) dt = g 0 2 0 2 The signal waveforms may be represented as a linear combination of two orthonormal signal waveforms: sm ( t ) = sm1 f1 ( t ) + sm 2 f 2 ( t ) 2 2 f1 ( t ) = g ( t ) cos 2 f c t and f 2 ( t ) = g ( t ) sin 2 f c t
The two-dimensional vectors sm=[sm1 sm2] are given by: 2 2 2 2 cos sin sm = ( m 1) ( m 1) , m = 1, 2,..., M g M M g
68

The Euclidean distance between signal points is: (e) d mn = sm sn
2 = ( sm1 sn1 ) + ( sm 2 sn 2 ) = g 1 cos ( m n ) M The minimum Euclidean distance corresponds to the case in which |m-n|=1, i.e., adjacent signal phases. 2 (e) d min = g 1 cos M
2 2 12
Average probability of symbol error for coherent M-ary PSK: E sin Pe erfc N M 0
69

Phase-modulated signals (Offset QPSK)
The carrier phase changes by 180 degrees whenever both the in-phase and quadrature components of the QPSK signal changes sign. This can result in problems for power amplifiers. The problem may be reduced by using offset QPSK. In offset QPSk, the bit stream responsible for generating the quadrature component is delayed (i.e. offset) by half a symbol interval with respect to the bit stream responsible for generating the in-phase component. 2 1 (t ) = cos (2f c t ), 0tT T
70

The two basis functions of offset QPSK are defined by 2 1 (t ) = cos (2f c t ), 0tT T
2 (t ) =
2 sin (2f c t ), T
T 3T t 2 2
The phase transitions likely to occur in offset QPSK are confined to 90 degrees. However, 90 degrees phase transitions in offset QPSK occur twice as frequently.
71

Amplitude fluctuations in offset QPSK due to filtering have a smaller amplitude than in the case of QPSK. The offset QPSK has exactly the same probability of symbol error in an AWGN channel as QPSK. The reason for the equivalence is that the statistical independence of the in-phase and quadrature components applies to both QPSK and offset QPSK.
72

73

Phase-modulated signals (/4-Shifted QPSK)
The carrier phase used for the transmission of successive symbols is alternately picked from one of the two QPSK constellations in the following figure and then the other.
74

It follows that a /4-shifted QPSK signal may reside in any one of eight possible phase states:
75

Attractive features of the /4-shifted QPSK scheme The phase transitions from one symbol to the next are restricted to /4 and 3/4. Envelope variations due to filtering are significantly reduced. /4-shifted QPSK signals can be noncoherently detected, thereby considerably simplifying the receiver design. Like QPSK signals, /4-shifted QPSK can be differently encoded, in which case we should really speak of /4shifted DQPSK .
76

Dual-Carrier Modulation (DCM)
Adopted in Multi-band OFDM (Ultra Wideband) The coded and interleaved binary serial input data, b[i] where i = 0, 1, 2, , shall be divided into groups of 200 bits and converted into 100 complex numbers using a technique called dual-carrier modulation. The conversion shall be performed as follows: 1.The 200 coded bits are grouped into 50 groups of 4 bits. Each group is represented as (b[g(k)], b[g(k)+1], b[g(k) + 50)], b[g(k) + 51]), where k [0, 49] and k [ 0, 24] 2k g (k ) = 2k + 50 k [ 25, 49]
77

2. Each group of 4 bits (b[g(k)], b[g(k)+1], b[g(k) + 50)], b[g(k) + 51]) shall be mapped onto a four-dimensional constellation, and converted into two complex numbers (d[k], d[k + 50]). 3. The complex numbers shall be normalized using a normalization factor KMOD.
The normalization factor KMOD = 10-1/2 is used for the dual-carrier modulation. An approximate value of the normalization factor may be used, as long as the device conforms to the modulation accuracy requirements.
78

79

Dual-Carrier Modulation (DCM) Encoding Table
80

Quadrature amplitude modulation (QAM)
Quadrature PAM or QAM: The bandwidth efficiency of PAM/SSB can also be obtained by simultaneously impressing two separate k-bit symbols from the information sequence {an} on two quadrature carriers cos2fct and sin2fct. The signal waveforms may be expressed as: j 2 f c t sm ( t ) = Re A jA g t e + ( ) ( ) ms mc , m = 1, 2,..., M , 0 t T
= Amc g ( t ) cos 2 f c t Ams g ( t ) sin 2 f c t
j m j 2 f c t sm ( t ) = Re V e g t e ( ) m = Vm g ( t ) cos ( 2 f c t + m )
Vm =
2 2 + Ams Amc and m = tan 1 ( Ams Amc )
81

Quadrature amplitude modulation (QAM) (cont.)
We may select a combination of M1-level PAM and M2phase PSK to construct an M=M1M2 combined PAM-PSK signal constellation.
82

As in the case of PSK signals, the QAM signal waveforms may be represented as a linear combination of two orthonormal signal waveforms f1(t) and f2(t): sm ( t ) = sm1 f1 ( t ) + sm 2 f 2 ( t )
f1 ( t ) =
2
g ( t ) cos 2 f c t and f 2 ( t ) =
g ( t ) sin 2 f c t
1 1 sm = [ sm1 sm 2 ] = Amc g Ams g 2 2 The Euclidean distance between any pair of signal vectors is: 1 2 2 (e) = sm sn = d mn g ( Amc Anc ) + ( Ams Ans ) 2
83

Several signal space diagrams for rectangular QAM:
(e) d min = d 2 g
84

Multidimensional signals:
We may use either the time domain or the frequency domain or both in order to increase the number of dimensions. Subdivision of time and frequency axes into distinct slots:
85

Orthogonal multidimensional signals
Consider the construction of M equal-energy orthogonal signal waveforms that differ in frequency:
j 2 f c t sm ( t ) = Re s t e ( ) lm , m = 1, 2,..., M , 0 t T
= slm ( t ) =
2 cos [ 2 f c t + 2 mf t ] T 2 j 2 mf t e , m = 1, 2,..., M , 0 t T T
This type of frequency modulation is called frequency-shift keying (FSK).

86

Orthogonal multidimensional signals (cont.)
These waveforms have equal cross-correlation coefficients:
km
4.2-45
2 T = 2
j 2 ( m k ) ft
dt =
sin T ( m k ) f
r Re ( km ) =
sin [T (m k )f ] cos [T (m k )f ] T (m k )f sin [2T (m k )f ] = 2T (m k )f
T ( m k ) f
e j T ( m k )f
Note that Re(km)=0 when f=1/2T and mk. | km |=0 for multiple of 1/T.
87

Cross-correlation coefficient as a function of frequency separation for FSK signals:
88

For f=1/2T, the M-FSK signals are equivalent to the Ndimensional vectors:
s1 = s2 = 0
[ [
0 0 0 0 0 0 0 0
] ]
sN = 0 0 0
The distance between pairs of signals is:

(e) d km = 2 , for all m, k
89

Orthogonal signals for M=N=3 and M=N=2.
90

Binary Frequency-Shift Keying
In a binary FSK system, symbols 1 and 0 are distinguished from each other by transmitting one of two sinusoidal waves that differ in frequency by a fixed amount.
2 Eb cos ( 2 f i t ) , 0 t Tb si ( t ) = Tb elsewhere 0,
The transmitted frequency is

fi = nc + i Tb
for some fixed integer nc and i = 1, 2

91

The FSK signal is a continuous phase signal in the sense that phase continuity is always maintained, including the inter-bit switching times. This form of digital modulation is an example of continuousphase frequency-shift keying (CPFSK). The signal s1(t) and s2(t) are orthogonal
2 cos (2f i t ), i (t ) = Tb 0, 0 t Tb elsewhere
92

sij = si (t ) j (t ) dt
Tb 0
Tb
2 Eb 2 cos (2f i t ) cos (2f i t ) dt Tb Tb i= j i j
Eb , = 0,
Eb s1 = 0
0 s2 = E b
93

Biorthogonal signals
A set of M biorthogonal signals can be constructed from M/2 orthogonal signals by simply including the negatives of the orthogonal signals. The correlation between any pair of waveforms is either r=-1 or 0.
Signal space diagrams for M=4 and M=6 biorthogonal signals.

94

Simplex signals
For a set of M orthogonal waveforms {sm(t)} or their vector representation {sm} with mean of: 1 M s= sm M m =1 Simplex signals are obtained by translating the origin of the m orthogonal signals to the point s . ' sm = sm s , m = 1, 2,..., M The energy per waveform is:
s
' 2 m
2 1 1 = sm s = + = 1 M M M
2
95

Simplex signals (cont.)
The cross correlation of any pair of signals is (4.2-27): ' ' sm sn 1 M Re ( mn ) = ' = ' 11 M sm sn
1 = M 1 The set of simplex waveforms is equally correlated and requires less energy, by the factor 1-1/M, than the set of orthogonal waveforms.
96
Signal space diagrams for M-ary simplex signals.

Signal waveforms from binary codes
A set of M signaling waveforms can be generated from a set of M binary code words of the form: Cm = [ cm1 cm 2 cmN ] , m = 1, 2,..., M , cmj = 0 or 1. Each component of a code word is mapped into an elementary binary PSK waveform:
cmj = 1 smj ( t ) =
2 c cos 2 f c t , 0 t Tc Tc
2 c cos 2 f c t , 0 t Tc cmj = 0 smj ( t ) = Tc

where Tc=T/N and c=/N.
97

Signal waveforms from binary codes (cont.)
The M code words {Cm} are mapped into a set of M waveforms {sm(t)}. The waveforms can be represented in vector form as: sm = [ sm1 sm 2 smN ] , m = 1, 2,..., M , where smj = / N . N is called the block length of the code and is also the dimension of the M waveforms.
Signal space diagrams for signals generated from binary codes.

98

Signal waveforms from binary codes (cont.)
Each of the M waveforms has energy . Any adjacent signal points have a cross-correlation coefficient:
(1 2 N ) N 2 r = = N
d ( e ) = 2 (1 r ) = 4 / N
The corresponding distance is:
99
4.3.2 Linear Modulation with Memory

There are some modulation signals with dependence between the signals transmitted in successive symbol intervals. This signal dependence is usually introduced for the purpose of shaping the spectrum of the transmitted signal so that it matches the spectral characteristics of the channel.
Examples of baseband signals and the corresponding data sequence

100

rectangular pulse of polarity A and the binary digit 0 is represented by a rectangular pulse of polarity A. The NRZ modulation is memoryless and is equivalent to a binary PAM or a binary PSK signal in a carrier-modulated system.
NRZ : the binary information digit 1 is represented by a
NRZI: the signal is different from the NRZ signal in that

transitions from one amplitude level to another occur only when a 1 is transmitted. This type of signal encoding is called differential encoding. The encoding operation is described by the relation: bk = ak bk 1 where {ak} is the binary information sequence into the encoder, {bk} is the output sequence of the encoder.
101

NRZI (cont.)
When bk=1, the transmitted waveform is a rectangular pulse of amplitude A, and when bk=0, the transmitted waveform is a rectangular pulse of amplitude A. The differential encoding operation introduces memory in the signal. The combination of the encoder and the modulator operations may be represented by a state diagram (Markov chain):
Input bit
01 102

NRZI (cont.)
The state diagram may be described by two transition matrices corresponding to the two possible input bits {0,1}. When ak=0, the encoder stays in the same state and the state transition matrix for a zero is:
1 0 T1 = 0 1 where tij=1 if ak results in a transition from state i to state j, i=1,2, and j=1,2; otherwise tij=0. The state transition matrix for ak=1 is: 0 1 T2 = 1 0
103

NRZI (cont.)
The trellis diagram for the NRZI signal:
Delay modulation is equivalent to encoding the data sequence by a run-length-limited code called a Miller code and using NRZI to transmit the encoded data (will be shown in Chapter 9).
104

Delay modulation (cont.):
The signal of delay modulation may be described by a state diagram that has four states:
There are two elementary waveforms s1(t) and s2(t) and their negatives -s1(t) and -s2(t), which are used for transmitting the binary information.
105

Delay modulation
The state transition matrices can be obtained from the figure in previous page: s3 s1 0 0 0 1 0 0 0 1 s4 s1 When ak=0: T1 = 1 0 0 0 s1 s4 s2 s4 1 0 0 0 When ak=1:
0 0 T2 = 0 0 1 0 0 0 1 0 1 0 0 0 1 0
106
s1 s2 s3 s2 s2 s3 s4 s3

Modulation techniques with memory such as NRZI and Miller coding are generally characterized by a K-state Markov chain with stationary state probabilities {pi, i=1,2,,K} and transition probabilities {pij, i,j=1,2,,K}. Associated with each transition is a signal waveform sj(t), j=1,2,,K. Transition probability matrix (P) can be arranged in matrix form as: p1K p11 p12 p p p 22 2K P = 21 p KK pK 1 pK 2
107

The transition probability matrix can be obtained from the transition matrices {Ti} and the corresponding probabilities of occurrence of the input bits:
P = qiTi
i =1 2
where q1=P(ak=0) and q2=P(ak=1). For the NRZI signal with equal state probabilities p1=p2=0.5,
1 2 P= 1 2
1 2 1 1 0 1 0 1 + = 1 2 0 1 2 1 0 2
108

The transition probability matrix for the Miller-coded signal with equally likely symbols (q1=q2=0.5 or, equivalently, p1=p2=p3=p4=0.25) is: 1 1 0 0 2 2 0 0 1 1 2 2 P= 1 1 0 0 2 2 1 1 0 0 2 2 The transition probability matrix is useful in the determination of the spectral characteristics of digital modulation techniques with memory.
109
4.3.3 Non-linear Modulation Methods with Memory---CPFSK and CPM

Introduction
In this section, we consider a class of digital modulation methods in which the phase of the signal is constrained to be continuous. This constraint results in a phase or frequency modulator that has memory. The modulation method is also non-linear.
Continuous-phase FSK (CPFSK)

A conventional FSK signal is generated by shifting the carrier by an amount f n = 1 2 f I n , I n = 1, 3, , ( M 1), to reflect the digital information that is being transmitted. This type (conventional type) of FSK signal is memoryless.
110

Continuous-phase FSK (CPFSK) (cont.)
The switching from one frequency to another may be accomplished by having M=2k separate oscillators tuned to the desired frequencies and selecting one of the M frequencies according to the particular k-bit symbol that is to be transmitted in a signal interval of duration T=k/R seconds. The reasons why we have CPFSK: (or the defects of conventional FSK) Such abrupt switching from one oscillator output to another in successive signaling intervals results in relatively large spectral side lobes outside of the main spectral band of the signal. Consequently, this method requires a large frequency band for transmission of the signal.
111

Solution: To avoid the use of signals having large spectral side lobes, the information-bearing signal frequency modulates a single carrier whose frequency is changed continuously. The resulting frequency-modulated signal is phasecontinuous and, hence, it is called continuous-phase FSK (CPFSK) . This type (continuous-phase type) of FSK signal has memory because the phase of the carrier is constrained to be continuous.
112

In order to represent a CPFSK signal, we begin with a PAM signal: d (t ) = I n g (t nT )
n
d(t) is used to frequency-modulate the carrier. {In} denotes the sequence of amplitudes obtained by mapping k-bit blocks of binary digits from the information sequence {an} into the amplitude levels 1,3,,(M-1). g(t) is a rectangular pulse of amplitude 1/2T and duration T seconds.
113

Equivalent low-pass waveform v(t) is expressed as t 2 v(t ) = exp j 4Tf d d ( )d + 0 T fd is the peak frequency deviation, 0 is the initial phase of the carrier. The carrier-modulated signal may be expressed as 2 cos 2 f c t + ( t ; I ) + 0 s (t ) = T where (t;I) represents the time-varying phase of the carrier.
114

t t n 1
( t ; I ) = 4 Tf d d ( ) d = 4 Tf d I n g ( nT ) d n = 2 f d T
k =
+ 2 f d ( t nT ) I n
nTt(n+1)T
0 (t < 0) n 1 q ( t ) = t 2T ( 0 t T ) = h Ik k = h = 2 fdT 1 2 ( t > T ) Note that, although d(t) contains discontinuities, the integral of d(t) is continuous. Hence, we have a continuous-phase signal. n represents the accumulation (memory) of all symbols up to time nT ?. Parameter h is called the modulation index.
115
= n + 2 hI n q ( t nT )

Continuous-phase modulation (CPM)
CPFSK becomes a special case of a general class of continuous-phase modulated (CPM) signals in which the carrier phase is
( t ; I ) = 2
k =
I h q ( t kT ) ,
k k
nT t ( n + 1) T
when hk=h for all k, the modulation index is fixed for all symbols. when hk varies from one symbol to another, the CPM signal is called multi-h. (In such a case, the {hk} are made to vary in a cyclic manner through a set of indices.)
The waveform q(t) may be represented in general as the integral of some pulse g(t), i.e.,
q (t ) =
g ( )d
t 0
116

Continuous-phase modulation (CPM) (cont.)
If g(t)=0 for t >T, the CPM signal is called full response CPM. (Fig a. b.) If g(t)0 for t >T, the modulated signal is called partial response CPM. (Fig c. d.)
LREC, L=2
LREC, L=1, results in CPFSK
LRC, L=2
LRC, L=1
117

Continuous-phase modulation (CPM) (cont.)
The CPM signal has memory that is introduced through the phase continuity. For L>1, additional memory is introduced in the CPM signal by the pulse g(t). Three popular pulse shapes are given in the following table. LREC denotes a rectangular pulse of duration LT. LRC denotes a raised cosine pulse of duration LT. Gaussian minimum-shift keying (GMSK) pulse with bandwidth parameter B, which represents the -3 dB bandwidth of the Gaussian pulse.
118

Continuous-phase modulation (CPM) (cont.) Some commonly used CPM pulse shapes 1 LREC ( 0 1 LT )
g ( t ) = 2 LT 0 1 g ( t ) = 2 LT
( otherwise )
2 t 1 cos LT 0
LRC
( 0 1 LT ) ( otherwise )
1/ 2 1/ 2 T T GMSK g ( t ) = Q 2 B t ( ln 2 ) Q 2 B t + ( ln 2 ) 2 2 1 x2 / 2 Q (t ) = e dt t 2
119

Minimum-shift keying (MSK).
MSK is a special form of binary CPFSK (and, therefore, CPM) in which the modulation index h=1/2. The phase of the carrier in the interval nT t (n+1)T is
1 n 1 (t ; I) = I k + I n q(t nT ) 2 k = 1 t nT = n + I n , 2 T
From 4.3-54
nT t (n + 1)T
1 t nT s (t ) = A cos 2 f c t + n + I n 2 T
The modulated carrier signal is
From 4.3-52
nT t (n + 1)T
1 1 I n t n I n + n , = A cos 2 f c + 4T 2
120

Minimum-shift keying (MSK) (cont.)
The expression indicates that the binary CPFSK signal can be expressed as a sinusoid having one of two possible frequencies in the interval nT t (n+1)T. If we define these frequencies as 1 f1 = f c + 4T 1 f2 = fc 4T Then the binary CPFSK signal may be written in the form
1 i 1 si (t ) = A cos 2 fi t + n + n (1) , 2
121
i = 1, 2

Why binary CPFSK with h=1/2 is called minimum-shift keying (MSK)? Because the frequency separation f =f2f1=1/2T, and f =1/2T is the minimum frequency separation that is necessary to ensure the orthogonality of the signals s1(t) and s2(t) over a signaling interval of length T. The phase in the nth signaling interval is the phase state of the signal that results in phase continuity between adjacent interval.
122

Minimum-shift keying (MSK) (Haykin)
Consider a continuous-phase frequency-shift keying (CPFSK) signal, which is defined for the interval 0tTb as follows: 2 Eb cos[2f1t + (0 )] for symbol 1 Tb s (t ) = 2 Eb cos[2f t + (0 )] for symbol 0 2 T b Eb is the transmitted signal energy per bit. Tb is the bit duration. The phase (0), denoting the value of the phase at time t = 0, sums up the past history of the modulation process up to time t = 0.
123

Another useful way of representing the CPFSK signal s(t) is to express it in the conventional form of an angle-modulated signal as follows: 2 Eb s (t ) = cos[2f c t + (t )] (*) Tb
(t) is the phase of s(t). When the phase (t) is a continuous function of time, we find that the modulated signal s(t) itself is also continuous at all times, including the inter-bit switching times.
124

The phase (t) of a CPFSK signal increases or decreases linearly with time during each bit duration of Tb seconds h t , 0 t Tb (t ) = (0)
Tb
The plus (minus) sign corresponds to sending symbol 1 (0). We can find that h h fc + = f1 f = f2 c 2Tb 2Tb We thus get 1 f c = ( f1 + f 2 ) 2
125
h = Tb ( f1 + f 2 )

h is referred to as the deviation ratio. With h=1/2, the frequency deviation equals half the bit rate. This is the minimum frequency spacing that allows the two FSK signals representing symbols 1 and 0 to be coherently orthogonal. At time t = Tb: h for symbol 1 (Tb ) ( 0 ) = h for symbol 0 This is to say, the sending of symbol 1 increases the phase of a CPFSK signal s(t) by h radians, whereas the sending of symbol 0 reduces it by an equal amount.
126

Phase Trellis
127

Phase trellis of the sequence 1101000 (h =1/2)
From (*), we may express the CPFSK signal s(t): 2 Eb 2 Eb s (t ) = cos t cos 2 f t sin sin ( 2 f c t ) ( t ) ( ) ( c ) Tb Tb
S I cos ( 2 f c t ) SQ sin ( 2 f c t )
128
(**)

Consider first the in-phase component SI With the deviation ratio h=1/2, (t ) = (0 ) t , 0 t Tb 2Tb where the plus (minus) sign corresponds to symbol 1 (0). A similar result holds for (t) in the interval Tb t 0, except that the algebraic sign is not necessarily the same in both intervals. Since the phase (0) is 0 or , depending on the past history of the modulation process, we find that, in the interval Tb t Tb, the polarity of cos[(t)] depends only on (0), regardless of the sequence of 1s and 0s transmitted before or after t = 0.
129

Thus, for Tb t Tb, the in-phase component sI(t) consists of a half-cycle cosine pulse defined as follows:
sI ( t ) 2 Eb = ( t ) cos Tb t , 2 Eb ( 0 ) cos cos Tb 2Tb Tb t Tb t
2 Eb = cos Tb 2Tb
where the plus (minus) sign corresponds to (0) = 0 ((0)=).
130

Similarly, for 0 t 2Tb, the quadrature component sQ(t) consists of a half-cycle sine pulse, whose polarity depends only on (Tb), defined as follows:
2 Eb 2 Eb sQ ( t ) sin sin sin t ( t ) (Tb ) = Tb Tb 2Tb 2 Eb sin t , = Tb 2Tb 0 t 2Tb
where the plus (minus) sign corresponds to (Tb) = /2 ((Tb) = -/2).

131

MSK signal may assume any one of four possible forms: The phase(0)=0 and (Tb)=/2, corresponding to the transmission of symbol 1. The phase(0)= and (Tb)=/2, corresponding to the transmission of symbol 0. The phase(0)= and (Tb)=-/2, corresponding to the transmission of symbol 1. The phase(0)=0 and (Tb)=-/2, corresponding to the transmission of symbol 0.
132

Define the orthonormal basis functions: 2 1 (t ) = t cos cos (2f c t ), 0 t Tb Tb 2Tb
2 2 (t ) = sin 2T Tb b t sin (2f c t ), 0 t Tb
From (**), we have:
s(t ) = s11 (t ) + s22 (t ),
0 t Tb
where the coefficients s1 and s2 are related to the phase states (0) and(Tb), respectively.
133

s1 =
s2 =
Tb Tb
2Tb
s ( t )1 ( t ) dt = Eb cos ( 0 ) ,
Tb t Tb
s ( t )2 ( t ) dt = Eb sin (Tb ) , 0 t 2Tb Both integrals are evaluated for a time interval equal to twice the bit duration. Both the lower and upper limits of the product integration used to evaluate the coefficient s1 are shifted by the bit duration Tb with respect to those used to evaluate the coefficient s2. The time interval 0tTb, for which the phase states (0) and(Tb) are defined, is common to both integrals.
0
134

Signal space diagram for MSK system
135

In QPSK the transmitted symbol is represented by any one of the four message points, whereas in MSK one of two message points is used to represent the transmitted symbol at any one time.
136
137

Coherent detection of the MSK signal In the case of an AWGN channel, the received signal is given by x (t ) = s (t ) + w (t ) where s(t) is the transmitted MSK signal, and w(t) is the sample function of a white Gaussian noise process of zero mean and power spectral density N0/2. For the optimal decision of (0),
x1 =
Tb Tb
x ( t ) 1 ( t ) dt = s1 + w1 ,
Tb t Tb
( 0) = 0 If x1 > 0 ( 0) = If x1 < 0
138

Coherent detection of the MSK signal (cont.) For the optimum detection of (Tb),
x2 =
2Tb 0
x ( t ) 2 ( t ) dt = s2 + w2 ,
0 t 2Tb
If x2 > 0 (Tb ) = / 2 If x2 < 0 (Tb ) = / 2
Error probability of MSK Eb 1 Pe = erfc N 2 0 which is exactly the same as that for binary PSK and QPSK.
139

Block diagrams for MSK transmitter
140

Block diagrams for coherent MSK receiver
141

Power spectra of MSK signals
g ( f ) 32 Eb cos ( 2 Tb f ) SB ( f ) = 2 = 2 2 2 2 16 1 T T f b b
2
MSK does not produce as much interference outside the signal band of interest as QPSK.
142

MSK may also be represented as a form of four-phase PSK. The equivalent low-pass digitally modulated signal in the form
v(t ) =
n =
[I
2n
g (t 2nT ) jI 2 n +1 g (t 2nT T ) ]
g(t) is a sinusoidal pulse

t sin g (t ) = 2T 0 (0 t 2T ) (otherwise)
Problem 4.14
143

This type of signal is viewed as a four-phase PSK signal in which the pulse shape is one-half cycle of a sinusoid (0 ~ ). The even-numbered binary-valued (1) symbols {I2n} of the information sequence {In} are transmitted via the cosine of the carrier, while the odd-numbered symbols {I2n+1} are transmitted via the sine of the carrier. The transmission rate on the two orthogonal carrier components is 1/2T bits/s so that the combined transmission rate is 1/T bits/s. Note that the bit transitions on the sine and cosine carrier components are staggered or offset in time by T seconds.
144

For this reason, the signal s (t ) = A I 2 n g (t 2nT ) cos 2 f c t n =
+ I 2 n +1 g (t 2nT T ) sin 2 f c t n = is called offset quadrature PSK (OQPSK) or staggered quadrature PSK (SQPSK). Figure in next page illustrates the representation of an MSK signal as two staggered quadrature-modulated binary PSK signals. The corresponding sum of the two quadrature signals is a constant amplitude, frequency-modulated signal.
145

146

Compare the waveforms for MSK with OQPSK (pulse g(t) is rectangular for 0t2T) and with conventional QPSK (pulse g(t) is rectangular for 0t2T). All three of the modulation methods result in identical data rates. The MSK signal has continuous phase. The OQPSK signal with a rectangular pulse is basically two binary PSK signals for which the phase transitions are staggered in time by T seconds. Thus, the signal contains phase jumps of 90. The conventional four-phase PSK (QPSK) signal with constant amplitude will contain phase jumps of 180 or 90 every 2T seconds.
147

Compare the waveforms for MSK with OQPSK and QPSK (cont.)
148

149
4.4 Spectral Characteristics of Digitally Modulated Signals In most digital communication systems, the available channel bandwidth is limited. The system designer must consider the constraints imposed by the channel bandwidth limitation in the selection of the modulation technique used to transmit the information. From the power density spectrum, we can determine the channel bandwidth required to transmit the informationbearing signal.
150
4.4.1 Power Spectra of Linearly Modulation Signals

Beginning with the form j 2 f c t s ( t ) = Re v t e ( ) where (t) is the equivalent low-pass signal. Autocorrelation function (from 4.1-50) j 2 f c ss ( ) = Re ( ) e Power density spectrum (from 4.1-51) 1 ss ( f ) = [ ( f f c ) + ( f f c )] 2 First we consider the general form Random Variable
(t ) =
where the transmission rate is 1/T = R/k symbols/s and {In} represents the sequence of symbols.
151
n =
g (t nT )
Deterministic

Autocorrelation function 1 (t + ; t ) = E (t ) (t + ) 2 1 = E In I m g (t nT ) g (t + mT ) 2 n = m = We assume the {In} is WSS with mean i and the autocorrelation function ii (m) = 1 2 E I n I n+m
(t + ; t ) =
n = m =
( m n) g
ii m '=
(t nT ) g (t + mT ) let m ' = m n
= =
(m ') g
ii n = n =
(t nT ) g ( t + ( m '+ n ) T ) let m = m '
m =
( m ) g ii (t nT ) g (t + nT mT )
152

The second summation g (t nT ) g (t + nT mT )
n =
is periodic in the t variable with period T. Consequently, (t+;t) is also periodic in the t variable with period T. That is (t + T + ; t + T ) = (t + ; t ) In addition, the mean value of v(t), which is E [ (t ) ] = E I n g (t nT ) = i g (t nT ) n = n = is periodic with period T.
153

Therefore v(t) is a stochastic process having a periodic mean and autocorrelation function. Such a process is called a cyclostationary process or a periodically stationary process in the wide sense. In order to compute the power density spectrum of a cyclostationary process, the dependence of (t+;t) on the t variable must be eliminated. Thus, 1 ( ) = T
2 T 2
(t + ; t )dt
1 = ii (m) m = n = T 1 = ii (m) m = n = T

2 T 2
g (t nT ) g (t + nT mT )dt g (t ) g (t + mT )dt ( t = t nT )
T 2 nT
T 2 nT
154

We interpret the integral as the time-autocorrelation function of g(t) and define it as gg ( ) = g (t ) g (t + )dt convolution Consequently, 1 ( ) = ii (m) gg ( mT ) T m = The (average) power density spectrum of v(t) is in the form 1 2 ( f ) = G ( f ) ii ( f ) Applying equation 2.2-27 T where G( f ) is the Fourier transform of g(t), and ii( f ) denotes the power density spectrum of the information sequence ii ( f ) = ii (m)e j 2fmT
m =
155

The result illustrates the dependence of the power density spectrum of v(t) on the spectral characteristics of the pulse g(t) and the information sequence {In}. That is, the spectral characteristics of v(t) can be controlled by (1) design of the pulse shape g(t) and by (2) design of the correlation characteristics of the information sequence. Whereas the dependence of ( f ) on G( f ) is easily understood upon observation of equation, the effect of the correlation properties of the information sequence is more subtle. First of all, we note that for an arbitrary autocorrelation ii(m) the corresponding power density spectrum ii( f ) is periodic in frequency with period 1/T. (see next page)
156

In fact, the expression relating the spectrum ii(f ) to the autocorrelation ii(m) is in the form of an exponential Fourier series with the {ii(m)} as the Fourier coefficients.
ii (m) = T
1 2T
1 2T
ii ( f )e j 2fmT df
Second, let us consider the case in which the information symbols in the sequence are real and mutually uncorrelated. In this case, the autocorrelation function ii(m) can be expressed as (applying 2.2-5 and 2.2-6)
2 2 + i i ii (m) = 2 i
(m = 0) (m 0)
where i2 denotes the variance of an information symbol.

157

Substitute for ii(m) in equation, we obtain It may be viewed as ii ( f ) =
1 t nT = ( ) s Ts n = 2 where s = Ts
m =
j 2 fmT ( m ) e ii
n =
jns t
= +
2 i 2 i
2 i
m =
e j 2 fmT
the exponential Fourier series of a periodic train of impulses with each impulse having an area 1/T.
m (f ) = + T m = T
i2
The desired result for the power density spectrum of v(t) when the sequence of information symbols is uncorrelated.
m m ( f ) = G( f ) + 2 G( ) ( f ) T T m = T T
2
i2
i2
158

The expression for the power density spectrum is purposely separated into two terms to emphasize the two different types of spectral components. The first term is the continuous spectrum, and its shape depends only on the spectral characteristic of the signal pulse g(t). The second term consists of discrete frequency components spaced 1/T apart in frequency. Each spectral line has a power that is proportional to |G( f )|2 evaluated at f = m/T. Note that the discrete frequency components vanish when the information symbols have zero mean, i.e., i=0. This condition is usually desirable for the digital modulation techniques under consideration, and it is satisfied when the information symbols are equally likely and symmetrically positioned in the complex plane.
159

Example 4.4-1 To illustrate the spectral shaping resulting from
g(t), consider the rectangular pulse shown in figure. The Fourier transform of g(t) is sin fT jfT G ( f ) = AT e fT Hence sin fT G ( f ) = ( AT ) fT Thus
2 2
2 i 2
zero
sin fT 2 2 ( f ) = A T + A i ( f ) fT
160
Decays inversely as the square of the frequency

Example 4.4-2 As a second illustration of the spectral shaping
resulting from g(t), we consider the raised cosine pulse
2 T A 0t T g (t ) = 1 + cos t , 2 T 2 its Fourier transform is: A sin fT jfT G( f ) = e 2 fT (1 f 2T 2 )

Decays inversely as the f 6
nonzero
161

Example 4.4-3 To illustrate that spectral shaping can also be
accomplished by operations performed on the input information sequence, we consider a binary sequence {bn} from which we form the symbols I n = bn + bn 1 E[XiXj]=E[Xi]E[Xj] The {bn} are assumed to be uncorrelated random variables, each having zero mean and unit variance. Then the autocorrelation function of the sequence {In} is
ii (m) = E ( I n I n + m ) = E ( bn + bn 1 )( bn + m + bn + m 1 )
2 2 E bn 2 b b b + + n n 1 n 1 2 b E n + bn bn +1 + bn 1bn +1 + bn bn 1 = 2 E b b b b b + + n n 1 n n 2 n 1 + bn 1bn 2 E [bn bn + m + bn bn + m 1 + bn 1bn + m + bn 1bn + m 1 ]
( m = 0) 2 ( m = +1) = 1 ( m = -1) 0 ( otherwise )
E[(Xi-0)2]=1
(m = 0) (m = 1) (otherwise)
162

Example 4.4-3 (cont.)
Hence, the power density spectrum of the input sequence is (from 4.4-13) ii ( f ) = ii (m)e j 2 fmT
m =
= 2(1 + cos 2 fT ) = 4 cos 2 fT and the corresponding power density spectrum for the (low-pass) modulated signal is (from 4.4-12)
4 2 ( f ) = G ( f ) cos 2 fT T
163

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Wireless Information Transmission System Lab.

Chapter 4 Characterization of Communication Signals and Systems

4.2 Signal Space Representations

4.4 Spectral Characteristics of Digitally Modulated Signals

4.1.1 Representation of Band-Pass Signals

Spectrum of a band-pass signal.

Our object is to develop a mathematical representation of such signals.

4.1.1 Representation of Band-Pass Signals

4.1.1 Representation of Band-Pass Signals

4.1.1 Representation of Band-Pass Signals

4.1.1 Representation of Band-Pass Signals

4.1.1 Representation of Band-Pass Signals

4.1.1 Representation of Band-Pass Signals

The Fourier transform of s(t) is:

4.1.1 Representation of Band-Pass Signals

4.1.1 Representation of Band-Pass Signals

4.1.2 Representation of Linear BandPass Systems

Note that h ( t ) is real.

4.1.2 Representation of Linear BandPass Systems

4.1.3 Response of a Band-Pass System to a Band-Pass Signal

4.1.3 Response of a Band-Pass System to a Band-Pass Signal

From 4.1-21 and 4.1-28.

Sl ( f f c ) H l ( f f c ) = 0 4.1-26 1 R( f ) = S f f H f f S f f H + ( ) ( ) ( ) l c l c l c l ( f fc ) 2 Rl ( f ) Sl ( f ) H l ( f ) 1 Compared R f f R f f = + ( ) ( ) l c l c r t = with 4.1-21 l ( ) sl ( ) hl ( t ) d 2

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

4.1.4 Representation of Band-Pass Stationary Stochastic Processes Proof of xx ( ) = yy ( ) and xy ( ) = yx ( )

= xx ( ) cos 2 f c t cos 2 f c ( t + ) + yy ( ) sin 2 f c t sin 2 f c ( t + ) xy ( ) sin 2 f c t cos 2 f c ( t + ) yx ( ) cos 2 f c t sin 2 f c ( t + )

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

is symmetric about f = 0).

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

The band-pass white noise can be represented by:

4.1.4 Representation of Band-Pass Stationary Stochastic Processes

4.2 Signal Space Representations

The inner product of two n-dimensional vectors is defined as:

4.2.1 Vector Space Concepts

4.2.1 Vector Space Concepts

4.2.1 Vector Space Concepts

4.2.2 Signal Space Concepts

4.2.2 Signal Space Concepts

The Cauchy-Schwarz inequality is:

with equality when x2(t)=ax1(t), where a is any complex number.

4.2.3 Orthogonal Expansions of Signals

4.2.3 Orthogonal Expansions of Signals

e (t ) = s (t ) s (t ) The energy of the approximation error:

4.2.3 Orthogonal Expansions of Signals

4.2.3 Orthogonal Expansions of Signals

Under such condition, we may express s(t) as:

4.2.3 Orthogonal Expansions of Signals

4.2.3 Orthogonal Expansions of Signals

Then c12 f1(t) is subtracted from s2(t): f 2' ( t ) = s2 ( t ) c12 f1 ( t )

4.2.3 Orthogonal Expansions of Signals

4.2.3 Orthogonal Expansions of Signals

sk = [ sk1 sk 2 ... skN ]

4.2.3 Orthogonal Expansions of Signals

4.2.3 Orthogonal Expansions of Signals

4.2.3 Orthogonal Expansions of Signals

The Euclidean distance between a pair of signals is defined as:

When m=k=for all m and k, this expression simplifies to:

4.3 Representation of Digitally Modulated Signals

DC 04 Signal&System

DC 04 Signal&System