DC 03 Digital Modulation Scheme

Digital M d l i S h Di i l Modulation Schemes
Wireless Information Transmission System Lab. Institute of Communications Engineering g g National Sun Yat-sen University Yat-
Table of Contents
Representation of Band-Pass Signals and Systems Representation of Band Pass Signals Band-Pass Representation of Linear Band-Pass Systems Response of a Band-Pass System to a Band-Pass Signal Representation of Band-Pass Stationary Stochastic Processes Signal Space Representations Vector Space Concepts/Signal Space Concepts Orthogonal Expansions of Signals Representation of Digitally Modulated Signals Memoryless Modulation Methods Non-linear Modulation Methods with Memory-CPFSK and CPM Power S t P Spectrum of Di it ll M d l t d Si l f Digitally Modulated Signals Power Spectral Density of Linearly Modulated Signals
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Representation of Band-Pass Signals and Systems Band
The channel over which the signal is transmitted is limited in bandwidth to an interval of frequencies centered about the carrier. Signals d h Si l and channels (systems) that satisfy the condition l ( ) h if h di i that their bandwidth is much smaller than the carrier frequency are termed narrowband band-pass signals and f d b db d i l d 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.
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. p p g
Our object is to develop a mathematical representation of such signals. g

4
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

= F 1 2u ( f ) F 1 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 th pre-envelope of s(t). l f () 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:
Real-Valued R lVl d
1 1 s (t ) = s (t ) = t
s ( ) t
A filter, called a Hilbert transformer, is defined as:
1 h (t ) = , <t < t The i l Th signal s (t ) may be viewed as the output of the Hilbert b i d th t t f th Hilb t transformer when excited by the input signal s(t). The frequency response of this filter is:
j 1 1 j 2 ft j 2 ft H ( f ) = h (t ) e dt = e dt = 0 t j
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( f > 0) ( f = 0) ( f < 0)
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 0 ) /2 0. 90 degrees phase shifter for all frequencies in the input signal. The analytic signal s+(t) is a band-pass signal. To obtain an y g () p g 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 c t
sl ( t ) = x ( t ) + jy ( t ) j 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 band-pass low-frequency 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 bandpass signal s(t). i l () s ( t ) + j s ( t ) = s ( t ) e s(t) can also be written as:
l j 2 fct
s ( t ) = Re x ( t ) + jy ( t ) e j 2 fc t = Re sl ( t ) e j 2 fc t The low pass signal sl(t) is usually called the complex envelope p g () y p p 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:
x (t )
y (t )
j 2 f t + ( t ) s ( t ) = Re sl ( t ) e j 2 fc t = Re a ( t ) e c = 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). s(t)
Three equivalent representations of band-pass signals: s ( t ) = x ( t ) cos 2 f c t y ( t ) sin 2 f c t i

= Re sl ( t ) e j 2 fct = a ( t ) cos 2 f c t + ( t )
The F i t Th Fourier transform of s(t) i f f ( ) is:

S ( f ) = s (t ) e
j 2 ft
dt =
1 Re ( ) = + 2
Re sl ( t ) e j 2 fct e j 2 ft dt
1 S ( f ) = sl ( t ) e j 2 fct + sl ( t ) e j 2 fct e j 2 ft dt 2 1 = Sl ( f f c ) + Sl ( f f c ) 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
2 j 4 f t 1 2 j 4 f c t * * sl ( t ) = a ( t ) e j ( t ) c = 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 4 f c t + 2 ( t ) dt 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 neglected. g 2 1 = sl ( t ) dt 2
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Representation of Linear Band-Pass Systems Band
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 ), * j 2 ( 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 ) ( f > 0) Define H l ( f f c ) = ( f < 0) 0 0 * then H l ( f f c ) = * H ( f ) = H ( f )

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(f (f
< 0)
> 0)
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 = 2 Re hl ( t ) e j 2 fc t 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
H l ( x ) e j 2 xt dx e j 2 fc t = hl* ( t ) e j 2 fc t
hl(t) i the inverse Fourier transform of Hl( f ) is th i F i t f f ). In general, the impulse response hl(t) of the equivalent low-pass system is complex valued complex-valued.
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We have shown that narrow band band-pass signals and systems can be represented by equivalent low-pass signals and systems 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 band-pass 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. r ( t ) = Re rl ( t ) e j 2 fc t
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Response of a Band-Pass System to a Band-Pass Signal BandBand
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 = Sl ( f f c ) + Sl ( f f c ) H l ( f f c ) + H l ( f f c ) 2
S l ( f f c ) H l ( f f c ) = 0
4.1-27 41 2
Sl ( f f c ) H l ( f f c ) = 0 4.1-26 1 R ( f ) = Sl ( f f c ) H l ( f f c ) + Sl ( f f c ) H l ( f f c ) 2 Rl ( f ) Sl ( f ) H l ( f ) 1 Compared = Rl ( f f c ) + Rl ( f f c ) r ( t ) = s ( ) h ( t ) d l with 4 1-21 4.1 21 l l 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.
Representation of Band-Pass Stationary BandStochastic Processes
In this section, we extend the representation to sample functions of a band-pass stationary stochastic process. In band pass 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 h ()i l f i f id 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 f q y 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: n ( t ) = a ( t ) cos 2 f c t + ( t )
= x ( t ) cos 2 f c t y ( t ) sin 2 f c t = Re z ( t ) e j 2 fc t

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 ( )
xy ( ) = yx ( )
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Proved next.
Proof of xx ( ) = yy ( ) and xy ( ) = yx ( ) Autocorrelation function of n(t) is: A t l ti f ti f (t) i 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 + )
by using: cos A cos B = 1 cos ( A B ) + cos ( A + 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 xx ( ) + yy ( ) cos 2 f c + xx ( ) yy ( ) cos 2 f c ( 2t + ) = 2 2 1 1 yx ( ) xy ( ) sin 2 f c yx ( ) + xy ( ) sin 2 f c ( 2t + ) y y y y 2 2 Since n(t) is stationary, the right-hand side must be independent of t. Q.E.D. As A a result, xx ( ) = yy ( ) and xy ( ) = yx ( ) Q E D lt Therefore, nn ( ) = xx ( ) cos 2 f c yx ( ) sin 2 f c Note that thi N t th t this equation is identical in form to: ti i id ti l i f t
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) 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 ( ) y y 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] () Re[ Therefore, the autocorrelation function nn() of the band-pass stochastic process is uniquely determined from the p q y 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 ) = =
Re zz ( ) e j 2 fc e j 2 f d
1 zz ( f f c ) + zz ( f f c ) 2
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Properties of the quadrature components
yx ( ) = xy ( )
(2.2-10) (2 2 10) (4.1-41)

zz ( ) = xx ( ) + yy ( ) jxy ( ) + j yx ( )
1 2
xy ( ) = yx ( )
xy ( ) = xy ( )
xy ( ) is an odd function of and xy ( 0 ) = 0. =>Uncorrelated 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
(4.1-54) (4 1 54)
is symmetric about f = 0).
If n(t) i Gaussian, x(t) and y(t +) are jointly Gaussian. For is G i d ) j i l G i F =0, they are statistically independent, and the joint PDF is:
p ( x, y ) = 1
x2 + y 2
2 2
, 2 = xx ( 0 ) = yy ( 0 ) = nn ( 0 )
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Representation of white noise
White i i Whit noise is a stochastic process that is defined to have a flat t h ti th t i d fi d t h fl t (constant) power spectral density over the entire frequency range. range This type of noise can t be expressed in terms of cant quadrature components, as a result of its wideband character. In the demodulation of narrowband signals in noise, it is g , 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 band pass filter.
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Representation of white noise (cont.)
The i Th noise resulting from passing the white noise process lti f i th hit i 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 ) = Re z ( t ) e j 2 fc t
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= x ( t ) cos 2 f c t y ( t ) sin 2 f c t
Representation of white noise (cont.)
The Th equivalent l i l t low-pass noise z(t) h a power spectral i (t) has t l density: 1 N0 f 2 B zz ( f ) = 1 0 f > B 2 sin B zz ( ) = N 0 and zz ( ) = N 0 ( )
The power spectral density for white noise and band-pass white noise is symmetric about f =0 so yx()=0 for all =0, .
4.1-54
zz ( ) = xx ( ) = yy ( ) (from 4.1-48)
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Vector Space Concepts
We will demonstrate that signals have characteristics that are similar to vectors and develop a vector representation for signal waveforms. Vector Space Concepts A vector v in an n-dimensional space is characterized by its n p [v y p components [ 1 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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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
vi2
i =1
A set of m vectors is said to be orthonormal if the vectors are t f t i id t b th l th t 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 g 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 i called an eigenvector and i the corresponding h is ll d d is h di eigenvalue.
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Gram-Schmidt procedure for constructing a set of orthonormal vectors. th l t
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 '2 Normalize the vector u2 to unit length. u 2 = u '2 ' u3 ' Selecting v3: u3 = 3 - ( v 3 u1 ) u1 - ( v 3 u 2 ) u 2 u3 = ' u3 By continuing this procedure, we construct a set of orthonormal vectors.
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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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Signal Space Concepts
The triangle inequality for two signals is:

x1 ( t ) + x2 ( t ) x1 ( t ) + x2 ( t )
The Cauchy-Schwarz inequality i Th C h S h i lit is:
x1 ( t ) x2 ( t ) dt
x1 ( t ) dt
2
12
x2 ( t ) dt
2
12
with equality when x2(t)=ax1(t), where a is any complex number. number
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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 o these u ct o s, .e., co b at o of t ese functions, i.e.,
s ( t ) = sk f k ( t )
k =1 33
The approximation error incurred is: e (t ) = s (t ) s (t ) The energy of the approximation error: 2 2 K e = s ( t ) s ( t ) dt = s ( t ) sk f k ( t ) dt (4.2-24) k =1 To minimize the energy of the approximation error, the error optimum coefficients in the series expansion of s(t) may be found by: y
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 meanll k l f i i h b d h square-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 h Si th f ti th l 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
s f ( t ) s ( t )dt
k =1 k k
2 = s sk
k =1
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When the minimum mean square approximation error min=0,
s = s = s ( t ) dt
k =1 2 k 2
Under U d such condition, we may express s(t) as: h diti
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 =0, functions {fn(t)} is said to be complete.
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Example 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 = 0 s ( t ) cos T dt , bk = T 0 s ( t ) sin T dt 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. g gy 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) ti th j ti f 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 g g k 1 f k' ( t ) where f k' ( t ) = sk (T ) cik f i ( t ) fk (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. constructed
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Once we h O have constructed the set of orthonormal waveforms t t d th t f th l f {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 = sk ( t ) dt = skn = sk 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} 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. 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 N th l th t { ill t i th i ti l configuration and their lengths will be invariant to the choice of orthonormal functions {fn(t)}.
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Representation of Digitally Modulated Signals
Digitally modulated signals, which are classified as linear, are conveniently expanded in terms of two orthonormal basis functions of the form:
2 2 f1 ( t ) = cos 2 f c t f2 (t ) = sin 2 f c t T T If sllm(t) is expressed as sllm(t)=xl(t)+jyl(t) sm(t) may be expressed (t) x (t), 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 d transmission system F i l d l f b d data i i
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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. interest The modulation process making the transmission possible involves switching ( y g) the amplitude, frequency, or phase of g (keying) p q y p a sinusoidal carrier in some fashion in accordance with the incoming data. There Th are three basic signaling schemes: amplitude-shift keying th b i i li h lit d hift k i (ASK), frequency-shift Keying (FSK), and phase-shift keying ( (PSK). )
44
Illustrative waveforms for the three basic forms of signaling binary information (a) ASK (b) PSK (c) FSK information. FSK.
45
Unlike ASK signals, both PSK and FSK signals have a constant envelope. This property makes PSK and FSK signals impervious p p p y g p to amplitude nonlinearities. In practice, we find that PSK and FSK signals are preferred to ASK signals f passband d transmission over nonlinear i l for b d data i i li 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.
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Memoryless Modulation Methods
Pulse-amplitude-modulated (PAM) signals
Double-sideband Double sideband (DSB) signal waveform may be represented as: sm ( t ) = Re Am g ( t ) e j 2 fc t
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 interval.
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Pulse-amplitude-modulated (PAM) signals (cont.)
The Th M PAM signals h i l have energies: i 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 g , m = 1,2,..., M 2 Digital PAM is also called amplitude shift keying (ASK) amplitude-shift (ASK). sm = Am
48
Signal Si l space diagram for digital PAM signals: di f di i l i l
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Gray encoding: Th mapping of k i f G d The i f information bits to the i bi h M=2k possible signal amplitudes may be done in a number of ways. The preferred assignment is one in which the ways 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: i l i

(e) d min = d 2 g
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Phase-modulated signals (Binary Phase-Shift Keying)
In I a coherent bi h binary PSK system, the pair of signals s1( ) and h i f i l (t) d 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.
51
To T ensure that each transmitted bit contains an integral h h i d bi i i l 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
52
53
Phase-modulated signals (Quadriphase-Shift Keying)
Quadriphase-Shift Keying (QPSK) Q di h Shif K i

2E 2E cos 2 f c t + ( 2i 1) , si ( t ) = T 4 0, 0t T e sew e e elsewhere
where i = 1, 2, 3, 4; E is the transmitted signal energy per symbol, and T i the symbol duration. b l d is h b ld i Equivalently
2E 2E cos ( 2i 1) cos ( 2 f ct ) sin ( 2i 1) sin ( 2 f ct ) T T 4 4
54
si ( t ) =
Defined D fi d a pair of quadrature carriers: i f d i
1 (t ) =
2 cos (2f c t ), T
0tT
2 sin (2f c t ), 0tT T There are four message points, and the associated signal vectors are d fi d by defined b E cos (2i 1) 4 (t ) = si , i = 1,2,3,4 E sin (2i 1) 4
2 (t ) =
55

Each E h possible value of the phase corresponds to a unique dibi ibl l f h h d i dibit. For example, we may choose the Gray coding.
56
Signal Si l space diagram of coherent QPSK system di f h
57
58
Phase-modulated signals (M-ary PSK)
The Th M signal waveforms are represented as: i l f d

sm ( t ) = Re g ( t ) e j 2 ( m 1) M e j 2 fc t , m = 1, 2,..., M , 0 t T 2 = g ( t ) cos 2 f c t + m 1) ( M 2 2 = g ( t ) cos m 1) cos 2 f c t g ( t ) sin ( ( m 1) sin 2 f c t M M
Digital phase modulation is usually called phase-shift keying (PSK).
59
Signal Si l space diagram for octaphase shift keying (i.e., M 8) di f h hif k i (i M=8)
60
The i l Th signal waveforms have equal energy: f h l 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: [ g y 2 2 2 2 cos sin sm = ( m 1) ( m 1) , m = 1, 2,..., M M g M g
61
The E lid Th Euclidean distance between signal points is: di t b t i l i t i (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 2 12
2 d min = g 1 cos M Average probability of symbol error for coherent M A b bili f b l f h M-ary PSK PSK: E Pe erfc sin N M 0
(e)
62
Phase-modulated signals (Offset QPSK)
The Th carrier phase changes by 180 degrees whenever both the i h h b d h b th th in-phase and quadrature components of the QPSK signal changes sign sign. This can result in problems for power amplifiers. The problem may be reduced by using offset QPSK. QPSK In offset QPSK, the bit stream responsible for generating the quad atu e co po e t s de ayed ( .e. o set) quadrature component is delayed (i.e. offset) by half a symbol a sy bo interval with respect to the bit stream responsible for generating the in-phase component.
1 (t ) =
2 cos (2f c t ), T
63
0tT
The t b i f ti Th two basis functions of offset QPSK are defined by f ff t d fi d b
1 (t ) =
2 cos (2f c t ), T
0tT
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 d H degrees phase t h transitions i offset QPSK iti in ff t occur twice as frequently.
64
65
Phase-modulated signals (/4-Shifted QPSK)
The Th carrier phase used for the transmission of successive i h d f th t i i f i symbols is alternately picked from one of the two QPSK constellations in the following figure and then the other. other
66
It follows that a /4-shifted QPSK signal may reside i any f ll th t /4 hift d i l id in one of eight possible phase states:
67
Attractive f t Att ti features of the /4-shifted QPSK scheme f th /4 hift d h The phase transitions from one symbol to the next are restricted to /4 and 3/4 3/4. Envelope variations due to filtering are significantly reduced. reduced /4-shifted QPSK signals can be noncoherently detected, t e eby co s de ab y s p y g t e ece ve des g . thereby considerably simplifying the receiver design. Like QPSK signals, /4-shifted QPSK can be differently encoded, in which case we should really speak of /4y p shifted DQPSK . /4-DQPSK is adopted in IS-54/136.
68
Quadrature amplitude modulation (QAM)
Quadrature PAM or QAM Th bandwidth efficiency of Q d QAM: The b d id h ffi i f 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. f The signal waveforms may be expressed as:
sm ( t ) = Re ( Amc + jAms ) g ( t ) e j 2 fct , m = 1, 2,..., M , 0 t T 12 = Amc g ( t ) cos 2 f c t Ams g ( t ) sin 2 f c t

sm ( t ) = Re Vm e j m g ( t ) e j 2 fc t = Vm g ( t ) cos ( 2 f c t + m ) Vm =
2 2 Amc + Ams and m = tan 1 ( Ams Amc )
69
Quadrature amplitude modulation (QAM) (cont.)
We W may select a combination of M1-level PAM and M2l bi i f l l d phase PSK to construct an M=M1M2 combined PAM-PSK signal constellation constellation.
70
As in th A i the case of PSK signals, the QAM signal waveforms f i l th i l f 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 ) =
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) d mn = sm sn = g ( Amc Anc ) + ( Ams Ans ) 2
71
Several signal space diagrams for rectangular QAM: S l i l di f l QAM
(e) d min = d 2 g
72
Multidimensional signals:
We W may use either the time domain or the frequency domain ith th ti d i th f d i or both in order to increase the number of dimensions. Subdivision of time and frequency axes into distinct slots:
73
Orthogonal multidimensional signals
Consider th C id the construction of M equal-energy orthogonal t ti f l th l signal waveforms that differ in frequency:
sm ( t ) = Re slm ( t ) e j 2 fc t , 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 t Thi type of frequency modulation is called f ff d l ti i ll d frequency-shift hift keying (FSK).
74
Orthogonal multidimensional signals (cont.)
These waveforms have equal cross-correlation coefficients: Th f h l l ti ffi i t
k km
4.2-45
2 T = 2
j 2 ( m k ) ft
dt =
sin T ( m k ) f
T ( m k ) f
e j T ( m k )f
sin [T (m k )f ] r Re ( km ) = cos [T (m k )f ] T (m k )f sin [2T (m k )f ] = 2T (m k )f
Note that Re(km)=0 when f=1/2T and mk. => Orthogonal The minimum frequency separation f that guarantees orthogonality i f 1/2T h li is f=1/2T.
75
Biorthogonal signals
A set of M bi h f biorthogonal signals can be constructed from M/2 l i l b df orthogonal signals by simply including the negatives of the orthogonal signals 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.

76
Simplex signals
For F a set of M orthogonal waveforms {sm(t)} or their vector t f th l f { th i t representation {sm} with mean of:
1 M s= 1 sm M m= Simplex signals are obtained by translating the origin of the m orthogonal signals to the point s . ' sm = sm s , m = 1 2 M 1, 2,...,
The energy per waveform is:
' 2 m
2 1 1 = sm s = + = 1 M M M
2
77
Simplex signals (cont.)

The Th cross correlation of any l ti f 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. f =
78
Signal space diagrams for M-ary simplex signals.
NonNon-linear Modulation Methods with Memory
Introduction
In thi I this section, we consider a class of digital modulation ti id l f di it l d l ti methods in which the phase of the signal is constrained to be continuous. continuous This constraint results in a phase or frequency modulator that has memory. y The modulation method is also non-linear. A conventional FSK signal is generated by shifting the carrier by an amount f n = 1 f I n , I n = 1, 3, , ( M 1), 2 to reflect the digital information that is being transmitted. This type ( yp (conventional type) of FSK signal is memoryless. yp ) g y
79
Continuous phase Continuous-phase FSK (CPFSK)
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 Suc b up sw c g o o e osc o ou pu o another in successive signaling intervals results in relatively large spectral side lobes outside of the main spectral band of the signal. signal Consequently, this method requires a large frequency band for transmission of the signal.
80
Solution: S l ti To avoid the use of signals having large spectral side lobes, the information-bearing signal frequency modulates a single information bearing carrier whose frequency is changed continuously. The resulting frequency-modulated signal is phase-continuous and, hence, it is called continuous-phase FSK (CPFSK) . This type (continuous phase type) of FSK signal has memory (continuous-phase because the phase of the carrier is constrained to be continuous.
81
In d t I order to represent a CPFSK signal, we begin with a PAM t i l b i ith 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). o e p ude eve s , ( ). g(t) is a rectangular pulse of amplitude 1/2T and duration T seconds.
82
Equivalent l E i l t low-pass waveform v(t) is expressed as f (t) i d 2 j 4Tf t d ( )d + v(t ) = exp p fd 0 T fd is the peak frequency deviation, 0 is the initial phase of the carrier. carrier The carrier-modulated signal may be expressed as 2 cos 2 f c t + ( t ; I ) + 0 s (t ) = T
(4.3-52)
where (t;I) represents the time-varying phase of the carrier.

83
( t ; I ) = 4 Tf d d ( ) d = 4 Tf d I n g ( nT ) d f f n
t t
= 2 f d T
0 (t < 0) (4.3-54) (4 3 54) n 1 q ( t ) = t 2T ( 0 t T ) n = h Ik h = 2 fdT k = 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.
84
= n + 2 hI n q ( t nT )
k =
n 1
+ 2 f d ( t nT ) I n
nTt(n+1)T Tt( +1)T
Continuous-phase modulation (CPM)
CPFSK b becomes a special case of a general class of i l f l l f 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. (I such a case, the {hk} are made to vary i a cyclic manner l i h (In h h d in li 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
85
Continuous-phase modulation (CPM) (cont.)

If g(t)=0 for t >T, the CPM signal is called full response CPM. (Fig a. b.) 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, LREC L=1 results in CPFSK
LRC, L=2 LRC, L=1
86
Continuous-phase modulation (CPM) (cont.)
The Th CPM signal has memory that is introduced through the phase i lh th t i i t d d th h th h continuity. For L>1 additional memory is introduced in the CPM signal by L>1, the pulse g(t). Three popular pulse shapes are given in the following table. table LREC denotes a rectangular pulse of duration LT. LRC denotes a raised cosine pulse of duration LT LT. Gaussian minimum-shift keying (GMSK) pulse with bandwidth parameter B, which represents the -3 dB 3 bandwidth of the Gaussian pulse.
87
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
t
Q (t ) =
1 x2 / 2 e dt 2
88
Minimum-shift keying (MSK).

MSK is a special form of binary CPFSK (and, therefore, CPM) in which (and therefore 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 From 4.3-52 2 T 1 1 I n t n I n + n , nT t (n + 1)T = A cos 2 f c + 4T 2

89
The Th modulated carrier signal is d l d i i li
Minimum-shift keying (MSK) (cont.)
The Th expression indicates that the bi i i di t th t th binary CPFSK signal can be i l b expressed as a sinusoid having one of two possible frequencies in the interval nT t (n+1)T If we define these frequencies as (n+1)T. 1 f1 = f c + 4T 1 f2 = fc 4T 4T
Then the binary CPFSK signal may be written in the form 1 si (t ) = A cos 2 fi t + n + n (1)i 1 , 2
90
i = 1, 2
Why binary CPFSK with h 1/2 is called minimum-shift Wh bi ith h=1/2 i ll d i i hift keying (MSK)? Because the frequency separation f =f2f1=1/2T and f =1/2T, =1/2T is the minimum frequency separation that is necessary g y g () () 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.
91
Compare th waveforms for MSK with OQPSK and QPSK ( t ) C the f f ith d (cont.)
92
93
Spectral Characteristics of Digitally Modulated Signals
In most digital communication systems, the available channel bandwidth is limited. 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.
94
Power Spectra of Linearly Modulated Signals
Beginning with the form s ( t ) = Re v ( t ) e j 2 fct where (t) is the equivalent low-pass signal. Autocorrelation function (from 4.1-50) ss ( ) = Re ( )e j 2 fc Power density spectrum (from 4.1-51) 1 ss ( f ) = [ ( f f c ) + ( f f c )] 2 Random Variable First we consider the general form
(t ) =
where the transmission rate is 1/T = R/k symbols/s and {In} represents the sequence of symbols.
95
n =
g (t nT ) T
Deterministic
Autocorrelation function 1 (t + ; t ) = E (t ) (t + ) 2 1 = E I n 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 =
(t nT ) g ( t + ( m '+ n ) T ) let m = m '
m =
ii (m) g (t nT ) g (t + nT mT )
n =
96
The second summation 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.
n =
g (t nT ) g (t + nT mT ) T T T
97
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 p p power density spectrum of a cycloy p y stationary 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
98
We interpret the integral as the time-autocorrelation function of g(t) and define it as gg ( ) = g (t ) g (t + )dt convolution Consequently, 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 g( the power density spectrum of the information sequence
ii ( f ) =
m =
ii (m)e j 2fmT
99
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) , p () y( ) 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. ti f th i f ti i btl 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)
100
In fact, the expression relating the spectrum ii(f ) to the ( ) p 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, uncorrelated case the autocorrelation function ii(m) can be expressed as (applying 2.2-5 and 2.2-6) i2 + i2 (m = 0) ii (m) = 2 i (m (m 0)
where i2 denotes the variance of an information symbol.
101
Substitute for ii(m) in equation, we obtain ii ( f ) =

m =
ii (m)e j 2 fmT
1 ( t nTs ) = Ts n = 2 where s = Ts
n =
jns t
= i2 + i2
2 i
m =
e j 2 fmT
It may be viewed as the exponential Fourier p series of a periodic train of impulses with each impulse having an area 1/T.
m = + (f T ) T m =
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 ) + G( T ) ( f T ) T T m =
2 102
2 i
2 i 2
The expression for the power density spectrum is purposely separated into two terms to emphasize the two different types of p p yp spectral components. The first term is the continuous spectrum, and its shape depends only on th spectral characteristic of the signal pulse g(t). the t l h t i ti f th i l l (t) The second term consists of discrete frequency components spaced 1/T apart in frequency. Each spectral line has a power that is frequency 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, consideration and it is satisfied when the information symbols are equally likely and symmetrically positioned in the complex plane.
103
Example To illustrate the spectral shaping resulting from g(t),

consider the rectangular pulse shown in figure The Fourier figure. transform of g(t) is
sin fT jfT e G ( f ) = AT fT Hence 2 2 2 sin fT G ( f ) = ( AT ) fT Thus
2
zero
sin fT + A2 i2 ( f ) ( f ) = i2 A2T fT
104
Decays inversely as the y y square of the frequency
Example As a second illustration of the spectral shaping resulting

from g(t) we consider the raised cosine pulse g(t),
A 2 T g (t ) = 1 + cos 0t T t , 2 T 2 its Fourier transform is: A sin fT G( f ) = e jfT 2 fT (1 f 2T 2 )

Decays inversely as the f
6
nonzero
105
Example 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 y 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 + 2bn bn 1 + bn 1 2 E bn + bn bn +1 + bn 1bn +1 + bn bn 1 = 2 E bnbn 1 + bnbn 2 + bn 1 + bn 1bn 2 E [bn bn + m + bnbn + m 1 + bn 1bn + m + bn 1bn + m 1 ]
( m = 0) 2 ( m = +1) = 1 ( m = -1) 0 th i ( otherwise )
E[(Xi-0)2]=1
(m = 0) (m = 1) (otherwise)
106
Example (cont.)
Hence, th power density spectrum of the input sequence is (from H the d it t f th i t i (f 4.4-13) ii ( f ) =
m =
ii (m)e j 2 fmT
= 2(1 + cos 2 fT ) ( f = 4 cos 2 fT and the corresponding power density spectrum for the (low-pass) d th di d it t f th (l ) modulated signal is (from 4.4-12)
4 2 ( f ) = G ( f ) cos 2 fT T
107

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Digital M d l i S h Di i l Modulation Schemes

Representation of Band-Pass Signals and Systems Band

Representation of Band-Pass Signals and Systems Band

Spectrum of a band-pass signal. p p g

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

Representation of Band-Pass Signals and Systems Band

Representation of Band-Pass Signals and Systems Band

A filter, called a Hilbert transformer, is defined as:

Representation of Band-Pass Signals and Systems Band

sl ( t ) = x ( t ) + jy ( t ) j 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

Representation of Band-Pass Signals and Systems Band

Representation of Band-Pass Signals and Systems Band

sl(t) can be also be written as:

s(t) can be represented as:

Representation of Band-Pass Signals and Systems Band

Three equivalent representations of band-pass signals: s ( t ) = x ( t ) cos 2 f c t y ( t ) sin 2 f c t i

The F i t Th Fourier transform of s(t) i f f ( ) is:

Representation of Band-Pass Signals and Systems Band

The energy in the signal s(t) is defined as:

Representation of Band-Pass Signals and Systems Band

Representation of Linear Band-Pass Systems Band

Note that h ( t ) is real.

H ( f ) ( f > 0) Define H l ( f f c ) = ( f < 0) 0 0 * then H l ( f f c ) = * H ( f ) = H ( f )

Representation of Linear Band-Pass Systems Band

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 = 2 Re hl ( t ) e j 2 fc t where

Representation of Linear Band-Pass Systems Band

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

The output of the system in the frequency domain is:

From 4.1-21 and 4.1-28.

Sl ( f f c ) H l ( f f c ) = 0 4.1-26 1 R ( f ) = Sl ( f f c ) H l ( f f c ) + Sl ( f f c ) H l ( f f c ) 2 Rl ( f ) Sl ( f ) H l ( f ) 1 Compared = Rl ( f f c ) + Rl ( f f c ) r ( t ) = s ( ) h ( t ) d l with 4 1-21 4.1 21 l l 2

Representation of Band-Pass Stationary BandStochastic Processes

Representation of Band-Pass Stationary BandStochastic Processes

Representation of Band-Pass Stationary BandStochastic Processes

Proof of xx ( ) = yy ( ) and xy ( ) = yx ( ) Autocorrelation function of n(t) is: A t l ti f ti f (t) i nn ( ) = E n ( t ) n ( 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 + )

by using: cos A cos B = 1 cos ( A B ) + cos ( A + B ) 2

Representation of Band-Pass Stationary BandStochastic Processes

Representation of Band-Pass Stationary BandStochastic Processes

Representation of Band-Pass Stationary BandStochastic Processes

Representation of Band-Pass Stationary BandStochastic Processes

Properties of the quadrature components

(2.2-10) (2 2 10) (4.1-41)

is symmetric about f = 0).

Representation of Band-Pass Stationary BandStochastic Processes

Representation of white noise

Representation of Band-Pass Stationary BandStochastic Processes

Representation of white noise (cont.)

The band-pass white noise can be represented by:

Representation of Band-Pass Stationary BandStochastic Processes

Representation of white noise (cont.)

Vector Space Concepts

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

Vector Space Concepts

Vector Space Concepts

Vector Space Concepts

Gram-Schmidt procedure for constructing a set of orthonormal vectors. th l t

Signal Space Concepts

Signal Space Concepts

The triangle inequality for two signals is:

The Cauchy-Schwarz inequality i Th C h S h i lit is:

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

Orthogonal Expansions of Signals

Suppose that s(t) is a deterministic, real-valued signal with finite energy: 2 s = s ( t ) dt