Professional Documents
Culture Documents
Communication 1
Communication 1
system analysis
Associate Prof. Dr. Ho Van Khuong
Office: Dept. of Telecom. Eng.
Email: khuong.hovan@yahoo.ca
Outlines
• Signal models
• Signal classifications
• Generalized Fourier series
• Fourier series
• The Fourier transform
• Power spectral density and correlation
• Signal and linear system
• Sampling theory
• The Hilbert transform
• Discrete Fourier transform and fast Fourier transform
Notes
• A signal is defined as the time history of some
quantity, usually a voltage or current.
• A system is a combination of devices and
networks (subsystems) chosen to perform a
desired function.
• Communication systems are complicated,
requiring a great deal of analysis and
experimentation with trial subsystems before
actual building of the desired system.
• Mathematical models for signals and systems.
Signal models
• Deterministic and random signals
– Deterministic signals: completely specified
functions of time.
• Sinusoidal: x(t) = A cos(ω0t), -∞ < t < ∞
• Unit rectangular pulse:
1, t ≤ 0.5
Π (t ) =
0, otherwise
– Random signals: take on random values at any
given time instant.
Signal models
• Periodic and aperiodic signals
– Periodic signal x(t): if and only if
x(t+T0) = x(t), -∞ < t < ∞ (*)
where the constant T0 is the period.
• Fundamental period is the smallest number satisfying
(*).
• Ex: x(t) = A cos(ω0t), -∞ < t < ∞
– Aperiodic signal: not satisfy (*).
Signal models
• Phasor signals and spectra
– Consider a useful periodic signal with T0 = 2π/ω0
x ( t ) Ae
j (ω0t +θ )
Rotating
= , −∞ < t < ∞
phasor
amplitude frequency (radians per phase in radians
second) or f0 = ω0/2π (Hz)
Phasor Ae jθ
– Relationship between the rotating phasor and the
real, sinusoidal signal
x ( t ) ) Re ( Ae (
+ θ ) Re ( =
x ( t ) A cos (ω0t= )
j ω t +θ )
= 0
1 1 1 j (ω0t +θ ) 1 − j (ω0t +θ )
A cos (ω0t + θ=
) x ( t ) + x * (=
t) Ae + Ae
2 2 2 2
Signal models
1
=x ( t ) 2sin 10π t − π
6
1
( t ) 2sin 10π t − π + cos ( 20π t )
y=
6
Signal models
• Singularity functions
– Unit impulse function (or delta function) δ(t)
∞ ∞
∫ x ( t ) δ ( t ) dt = x ( 0 )
−∞
∫ x ( t ) δ ( t − t ) dt =
−∞
0 x (t )
0
continuous at t = 0 continuous at t = t0
∫ δ ( t − t ) dt=
0 1, t1 < t0 < t2 1, t1 < t < t2
t1 x (t ) =
δ ( t − t0 )= 0, t ≠ t0 0, otherwise
Signal models
• Singularity functions
– Properties of δ(t)
• δ(at) = (1/|a|)δ(t), a is a constant.
• δ(-t) = δ(t) t
x ( t0 ) , t1 < t0 < t2
2
• Sifting property ∫ x ( t ) δ ( t − t0 ) dt =
0, otherwise
t1 undefined, t = t or t
0 1 2
( ) ( n)
( ) ( ) x ( t0 ) , t1 < t0 < t2
n ( n)
∫ x
t1
t δ t − t 0 dt = −1
x2 ( t ) = Au ( t )
x3 ( t ) = Ae
− j (ω0t +θ )
t −1/4 , t ≥ t0 ≥ 0
x4 ( t ) =
0, otherwise
Signal classifications
• Periodic signal with the period T0
T0 + t0
1
∫ x p ( t ) dt
2
P=
T0 t0
x p ( t ) A cos (ω0t + θ )
=
Generalized Fourier series
• Representation of signals as a series of
orthogonal functions.
• Consider the problem of representing a time
function x(t) on a T-s interval (t0, t0+T)
– A basis function set φn(t): φ1(t), φ2(t), …, φn(t) are
linearly independent (i.e., no one of them can be
expressed as a weighted sum of the other N-1).
– The approximation of x(t) Independent of time
N
xa ( t ) = ∑ X nφn ( t )
n =0
Generalized Fourier series
– The choice of
1
X n = ∫ x ( t ) φn∗ ( t ) dt
cn T
is to minimize
∫ x (t ) − x (t )
2
= ε=
Error N a dt
T
where φn(t) is assumed to be orthogonal
t0 +T0
cn , n = m
∫ φm ( t ) φ (=
*
t ) dt c=
nδ mn (all m and n)
0, n ≠ m
n
t0
Kronecker delta function
T n =0
– φn(t) is complete if
lim ( ε N )min = 0
N →∞
∫ x (t )
2
for any signal that is integrable square dt < ∞
T
– In the sense that the error is zero
∞
x ( t ) = ∑ X nφn ( t )
n =0
– Parseval’s theorem
∞
∫ x ( t ) dt = ∑ cn X n
2 2
T n =0
Example
sin (π t ) , 0 ≤ t ≤ 2 xa(t)?
x (t ) =
0, otherwise Error?
where
t0 +T0
1
Xn = ∫ x ( t )e − jnω0t dt
T0 t0
Fourier series
• Complex exponential Fourier series
– Note:
• The right-hand side of (*) is periodic with period T0,
since it is the sum of periodic rotating phasors with
harmonic frequencies.
• If x(t) is periodic with period T0, the Fourier series is an
accurate representation for x(t) for all t.
– Example: Find the complex exponential Fourier
series for the signal x(t) = cos(ω0t) + sin2(2ω0t)
where ω0 = 2π/T0.
Fourier series
• Symmetry properties of the Fourier
coefficients • Magnitude of X is even
n
• Argument of Xn is odd
– x(t) is real: X n* = X − n
– x(t) is even/odd: Xn is purely real/imaginary and is
an even/odd function of n.
– Signals with odd half-wave symmetry, i.e.
1
− x (t )
x t ± T0 =
2
where T0 is the period of x(t). Then
X n = 0, n = 0, ±2, ±4,...
Fourier series
• Trigonometric form of the Fourier series
– x(t) is real ∞
) X 0 + ∑ 2 X n cos ( nω0t + ∠X n )
x ( t=
n =1
∞ ∞
X 0 + ∑ An cos ( nω0t ) + ∑ Bn sin ( nω0t )
x (t ) =
=n 1=n 1
t0 +T0 t0 +T0
2 2
An = ∫ x ( t ) cos ( nω0t ) dt Bn = ∫ x ( t ) sin ( nω t ) dt
0
T0 t0
T0 t0
∞
=x (t ) ∑Xe
n = −∞
n
jnω0t
, t0 < t < t0 + T0
∞
) X 0 + ∑ 2 X n cos ( nω0t + ∠X n )
x ( t=
n =1
Fourier series
T0
τ=
4
• Line spectra
Pulse train
∞
t − nT0 − 0.5τ
x (t ) = ∑ ∏
n = −∞
A
τ
T0
τ= ; T0 same as in (a)
8
T0
τ= ;τ same as in (a)
4
Fourier transform of nonperiodic signals
• x(t) is nonperiodic but is an energy signal.
• The Fourier transform of x(t)
∞
X(f )=
ℑ x ( t ) =
∫ x ( t )e − j 2π ft
dt
−∞
• Time-delay x ( t − t ) ↔ X ( f ) e
0
− j 2π ft0
• Modulation x ( t ) cos ( 2π f t ) ↔ 12 X ( f − f ) + 12 X ( f + f )
0 0 0
Differentiation d x ( t ) ↔ ( j 2π f ) X ( f )
n
• n
n
dt
• Integration ∫ x ( λ ) d λ ↔ ( j 2π f ) X ( f ) + 1 X ( 0 ) δ ( f )
t
−1
−∞
2
∞ ∞
• Convolution ∫ x ( λ )x (t − λ ) d λ = ∫ x (t − λ )x ( λ ) d λ ↔ X ( f ) X ( f )
1 2 1 2 1 2
−∞ −∞ ∞
• Multiplication x ( t ) x ( t ) ↔ X ( f ) * X ( f ) =∫ X ( λ )X ( f − λ ) d λ
1 2 1 2 1 2
−∞
Fourier transform of nonperiodic signals -
Transform theorems
3. A ↔ Aδ ( f )
4. Ae j 2π f t ↔ Aδ ( f − f 0 )
0
Fourier transform of nonperiodic signals -
Transform theorems
Pulse-type signal
∞
=X(f) ∑ f s P ( nf s ) δ ( f − nf s ) P ( f ) = ℑ p ( t )
n = −∞
n = −∞ n = −∞
∞ ∞
∑ p ( t − mT ) ↔ ∑
m = −∞
s
n = −∞
f s P ( nf s ) δ ( f − nf s )
∞ ∞
∑
m = −∞
f s ∑ P ( nf s ) e j 2π nf s t
p ( t − mTs ) =
n = −∞
Power spectral density and correlation
• For power signals, power spectral density S(f)
∞ T
1
=P ∫ S (=
f ) df x ( t ) lim
=2
∫ x 2 ( t ) dt
T →∞ 2T
−∞ −T
(τ )
φ= ∫ x (λ ) x (λ +τ =
) dλ lim
T →∞ ∫ x (λ ) x (λ +τ ) dλ
−∞ −T
y(t) = H[x(t)]
• Linear time-invariant (LTI) system
– Linear: y(t) = H[α1x1(t)+α2x2(t)]
= α1H[x1(t)]+α2H[x2(t)]
= α1y1(t)+α2y2(t)
– Time-invariant: y(t-t0) = H[x(t-t0)]
Signals and linear systems
• Impulse response and superposition integral
– The impulse response h(t) of an LTI system is the
response of the system to an impulse applied at
t=0: h(t) = H[δ(t)]
– LTI system:
N N
x (t )
= ∑ α δ ( t − t )
n
n 1=
n linearity
→ y (t ) ∑α h (t − t )
=
time invariance
n 1
n n
– Superposition integral
∞ ∞
y (t ) = x (t ) ∗ h (t ) = ∫ x ( λ ) h ( t − λ )d λ = ∫ x ( t − λ ) h ( λ )d λ
−∞ −∞
Signals and linear systems
• Stability
– A LTI system is bounded-input, bounded-output
(BIBO) stable if every bounded input results in a
bounded output.
– A system is BIBO stable if and only if
∞
∫ h ( t ) dt < ∞
−∞
Signals and linear systems
• Causality
– A system is causal if it does not anticipate the
input.
– For a time-invariant causal system, the impulse
response
h(t) = 0, t < 0
Signals and linear systems
• Transfer (frequency-response) function
– Applying the convolution theorem of Fourier
transform
∞ ∞
y (t ) = x (t ) ∗ h (t ) = ∫ x ( λ ) h ( t − λ )d λ = ∫ x ( t − λ ) h ( λ )d λ
−∞ −∞
X ( f ) = ℑ{ x ( t )}
H ( f ) = ℑ{h ( t )}
Y ( f ) = ℑ{ y ( t )}
Y( f )= H( f )X ( f )
Transfer (frequency-
response) function of
the system
Signals and linear systems
• Symmetry properties of H(f)
H ( f ) = H ( f ) e j∠H ( f )
Fourier transform
1 dθ ( f )
Tg ( f ) = −
2π df
– For distortionless system,
θ(f )= −2π ft0 → Tg ( f ) = t0
– Group delay is the delay that a group of two or more
frequency components undergo in passing through a
linear system.
– The delay of a single-frequency component as the
input is defined as the phase delay θ(f)
Tp ( f ) = −
2π f
Signals and linear systems
• Nonlinear distortion
– Consider a nonlinear system with the input-output
characteristic:
y ( t ) a1 x ( t ) + a2 x 2 ( t ) ; a1 and a2 are constants
=
– The output
= is y ( t ) a1 A1 cos (ω1t ) + A2 cos (ω2t )
desired output
1
2
( 1
)
+ a2 A12 + A22 + a2 A12 cos ( 2ω1t ) + A22 cos ( 2ω2t )
2
harmonic distortion
{ }
+ a2 A1 A2 cos (ω1 + ω2 ) t + cos (ω1 − ω2 ) t
intermodulation distortion
Signals and linear systems
• Nonlinear distortion
Signals and linear systems
• Ideal filter
– B is the single-sided bandwidth of the filter
– Lowpass
H LP ( f=
) 0 (
H Π f / 2 B ) e − j 2π ft0
– Highpass
( f ) H 0 1 − Π ( f / 2 B ) e− j 2π ft0
H HP =
– Bandpass
H BP ( f=
) H1 ( f − f0 ) + H1 ( f + f0 ) e− j 2π ft0
where H1 ( f=) H 0Π ( f / B )
Signals and linear systems
• Ideal filter
Signals and linear systems
• Ideal filter
=hLP ( t ) 2 BH 0 sin c 2 B ( t − t0 )
• Questions:
– What are the restrictions on x(t) and TS to allow
perfect recovery of x(t) from xδ(t)?
– How is x(t) recovered from xδ(t)?
Sampling theory
• Theorem: if a signal x(t) contains no frequency
components for frequencies above f = W Hz,
then it is completely described by
instantaneous sample values uniformly spaced
in time with period TS < 1/2W. The signal can
be exactly reconstructed from the sampled
waveform by passing it through an ideal
lowpass filter with bandwidth B, where W < B
< fS-W with fS=1/TS. The frequency 2W is
referred to as the Nyquist frequency.
Sampling theory
• Spectrum of the sampled signal
∞
Xδ ( f ) fs
= ∑ X ( f − nf )
n = −∞
s
f − j 2π ft0
H ( f ) = H 0Π e , W ≤ B ≤ fS − W
2B
• The output spectrum with xδ(t) at the input, is
Y ( f ) = f S H 0 X ( f ) e − j 2π ft0
Distortionless
y ( t ) f S H 0 x ( t − t0 )
= recovery of
x(t) from xδ(t)
Sampling theory
• If the conditions of the sampling
theorem are not satisfied (either
because x(t) is not bandlimited
or because fS<2W), the
distortion (referred to as
aliasing in (a)) at the output of
the reconstruction filter is
inevitable.
• Error due to the nonideal
frequency response
characteristics of practical filters
(see (b)).
Sampling theory
• Theorem: if a signal has a spectrum of W Hz
and upper frequency limit fu, then a rate fs at
which the signal can be sampled is 2fu/m,
where m is the largest integer not exceeding
fu/W. All higher sampling rates are not
necessarily usable unless they exceed 2fu.
• Ex.: fs = 2(3)/2 = 3 samples per second
∞
X δ ( f ) 3 ∑ X ( f − 3n )
=
n = −∞
{
x ( t ) = Re x ( t ) e j 2π f0t }
= Re { x ( t )} cos ( 2π f 0t ) − Im { x ( t )} sin ( 2π f 0t )
inphase component quadrature component
{
h ( t ) = Re h ( t ) e j 2π f0t }
x ( t ) = Re { x ( t ) e j 2π f 0 t
}
y (t ) =
1
2
{
Re y ( t ) e j 2π f0t }
• Ex: find the complex envelope of the filter
output t
x ( t ) = Π cos ( 2π f 0t )
−ατt
h ( t ) = α e u ( t ) cos ( 2π f 0t )
Discrete Fourier transform (DFT)
• DFT N −1
=Xk ∑ n =
x e
n =0
− j 2π nk / N
, k 0,1,..., N − 1