Analog & Digital Signal

Slides adapted from ME Angoletta, CERN
-0.2
-0.1
0
0.1
0.2
0.3
0 2 4 6 8 10
sampling time, t
k
[ms]
V
o
l
t
a
g
e

[
V
]
t
s
-0.2
-0.1
0
0.1
0.2
0.3
0 2 4 6 8 10
sampling time, t
k
[ms]
V
o
l
t
a
g
e

[
V
]
t
s
Analog & digital signals
Analog & digital signals
Continuous function Continuous function V of
continuous continuous variable t (time,
space etc) : V(t).
Analog
Discrete function Discrete function V
k
of
discrete discrete sampling variable t
k
,
with k = integer: V
k =
V(t
k
).
Digital
-0.2
-0.1
0
0.1
0.2
0.3
0 2 4 6 8 10
time [ms]
V
o
l
t
a
g
e

[
V
]
Uniform (periodic) sampling.
Sampling frequency f
S
= 1/ t
S
Digital vs analog procing
Digital vs analog procing
Digital Signal Processing (DSPing)
More flexible.
Often easier system upgrade.
Data easily stored.
Better control over accuracy
requirements.
Reproducibility.
Advantages Advantages
A/D & signal processors speed:
wide-band signals still difficult to
treat (real-time systems).
Finite word-length effect.
Obsolescence (analog
electronics has it, too!).
Limitations Limitations
Digital system example
Digital system example
ms
V
A
N
A
L
O
G

A
N
A
L
O
G

D
O
M
A
I
N
D
O
M
A
I
N
ms
V
Filter
Antialiasing
k
A
D
I
G
I
T
A
L

D
I
G
I
T
A
L

D
O
M
A
I
N
D
O
M
A
I
N
A/D
k
A
Digital
Processing
ms
V
A
N
A
L
O
G

A
N
A
L
O
G

D
O
M
A
I
N
D
O
M
A
I
N
D/A
ms
V
Filter
Reconstruction
Sometimes steps missing
- Filter + A/D
(ex: economics);
- D/A + filter
(ex: digital output wanted).
General scheme
Digital system implementation
Digital system implementation
Sampling rate.
Pass / stop bands.
KEY DECISION POINTS: KEY DECISION POINTS:
Analysis bandwidth, Dynamic range
No. of bits. Parameters.
1
2
3
Digital
Processing
A/D
Antialiasing
Filter
ANALOG INPUT ANALOG INPUT
DIGITAL OUTPUT DIGITAL OUTPUT
Digital format.
What to use for processing?
See slide DSPing aim & tools
Sampling
Sampling
How fast must we sample a continuous
signal to preserve its info content?
Ex: train wheels in a movie.
25 frames (=samples) per second.
Frequency misidentification due to low sampling frequency.
Train starts wheels go clockwise.
Train accelerates wheels go counter-clockwise.
1
Why? Why?
* Sampling: independent variable (ex: time) continuous discrete.
Quantisation: dependent variable (ex: voltage) continuous discrete.
Here well talk about uniform sampling.
* *
Sampling - 2
Sampling - 2
__
s(t) = sin(2f
0
t)
-1.2
-1
-0.8
-0.6
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
1.2
t
s(t) @ f
S
f
0
= 1 Hz, f
S
= 3 Hz
-1.2
-1
-0.8
-0.6
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
1.2
t
__
s
1
(t) = sin(8f
0
t)
-1.2
-1
-0.8
-0.6
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
1.2
t
__
s
2
(t) = sin(14f
0
t)
-1.2
-1
-0.8
-0.6
-0.4
-0.2
0
0.2
0.4
0.6
0.8
1
1.2
t
s
k
(t) = sin( 2 (f
0
+ k f
S
) t ) , k
s(t) @ f
S
represents exactly all sine-waves s
k
(t) defined by:
1
The sampling theorem
The sampling theorem
A signal s(t) with maximum frequency f
MAX
can be
recovered if sampled at frequency f
S
> 2 f
MAX
.
Condition on f
S
?
f
S
> 300 Hz
t) cos(100 t) sin(300 10 t) cos(50 3 s(t) + =
F
1
=25 Hz, F
2
= 150 Hz, F
3
= 50 Hz
F
1
F
2
F
3
f
MAX
Example
1
Theo
*
*
Multiple proposers: Whittaker(s), Nyquist, Shannon, Kotelnikov.
Nyquist frequency (rate) f
N
= 2 f
MAX
or f
MAX
or f
S,MIN
or f
S,MIN
/2
Naming gets
confusing !
Frequency domain (hints)
Frequency domain (hints)
Time & frequency Time & frequency: two complementary signal descriptions.
Signals seen as projected onto time or frequency domains.
1
Bandwidth Bandwidth: indicates rate of change of a signal.
High bandwidth signal changes fast.
Ear Ear + brain act as frequency analyser: audio spectrum
split into many narrow bands low-power sounds
detected out of loud background.
Example
Sampling low-pass signals
Sampling low-pass signals

-B 0 B f
Continuous spectrum
(a) Band-limited signal:
frequencies in [-B, B] (f
MAX
= B).
(a)

-B 0 B f
S
/2 f
Discrete spectrum
No aliasing
(b) Time sampling frequency
repetition.
f
S
> 2 B no aliasing.
(b)
1

0 f
S
/2 f
Discrete spectrum
Aliasing & corruption
(c)
(c) f
S
2 B aliasing ! aliasing !
Aliasing: signal ambiguity Aliasing: signal ambiguity
in frequency domain in frequency domain
Antialiasing filter
Antialiasing filter

-B 0 B f
Signal of interest
Out of band
noise
Out of band
noise

-B 0 B f
S
/2
f
(a),(b) Out-of-band noise can alias
into band of interest. Filter it before! Filter it before!
(a)
(b)

-B 0 B f
Antialiasing
filter
Passband
frequency
(c)
Passband: depends on bandwidth of
interest.
Attenuation A
MIN
: depends on
ADC resolution ( number of bits N).
A
MIN, dB
~ 6.02 N + 1.76
Out-of-band noise magnitude.
Other parameters: ripple, stopband
frequency...
(c) Antialiasing Antialiasing filter filter
1
ADC - Number of bits N
ADC - Number of bits N
Continuous input signal digitized into 2
N
levels.
-4
-3
-2
-1
0
1
2
3
-4 -3 -2 -1 0 1 2 3 4
000
001
111
010
V
V
FSR
Uniform, bipolar transfer function (N=3) Uniform, bipolar transfer function (N=3)
Quantization step Quantization step q =
V
FSR
2
N
Ex: V
FSR
= 1V , N = 12 q = 244.1 V
LSB LSB
Voltage ( = q)
Scale factor (= 1 / 2
N
)
Percentage (= 100 / 2
N
)
-1
-0.5
0
0.5
1
-4 -3 -2 -1 0 1 2 3 4
- q / 2
q / 2
Quantisation error Quantisation error
2
ADC - Quantisation error
ADC - Quantisation error
2
Quantisation Error e
q
in
[-0.5 q, +0.5 q].
e
q
limits ability to resolve
small signal.
Higher resolution means
lower e
q
.
-0.2
-0.1
0
0.1
0.2
0.3
0 2 4 6 8 10
time [ms]
V
o
l
t
a
g
e

[
V
]
Frequency analysis: why?
Frequency analysis: why?
Fast & efficient insight on signals building blocks.
Simplifies original problem - ex.: solving Part. Diff. Eqns. (PDE).
Powerful & complementary to time domain analysis techniques.
The brain does it?
time, t frequency, f
F
s(t) S(f) = F[s(t)]
analysis analysis
synthesis synthesis
s(t), S(f) :
Transform Pair
General Transform as General Transform as
problem problem- -solving tool solving tool
Fourier analysis - tools
Fourier analysis - tools
Input Time Signal Frequency spectrum
=
1 N
0 n
N
n k 2
j
e s[n]
N
1
k
c
~
Discrete
Discrete
DFS DFS
Periodic
(period T)
Continuous
DTFT
Aperiodic
Discrete
DFT DFT
n f 2 j
e
n
s[n] S(f)

+
=
=

0
0.5
1
1.5
2
2.5
0 2 4 6 8 10 12
time, t
k
0
0.5
1
1.5
2
2.5
0 1 2 3 4 5 6 7 8
time, t
k
=
1 N
0 n
N
n k 2
j
e s[n]
N
1
k
c
~
**
**
Calculated via FFT
**
dt
t f j2
e s(t) S(f)

+

=
dt
T
0
t k j
e s(t)
T
1
k
c

= Periodic
(period T)
Discrete
Continuous
FT FT
Aperiodic
FS FS
Continuous
0
0.5
1
1.5
2
2.5
0 1 2 3 4 5 6 7 8
time, t
0
0.5
1
1.5
2
2.5
0 2 4 6 8 10 12
time, t
Note: j =-1, = 2/T, s[n]=s(t
n
), N = No. of samples
A little history
A little history
Astronomic predictions by Babylonians/Egyptians likely via trigonometric sums.
1669 1669: Newton stumbles upon light spectra (specter = ghost) but fails to
recognise frequency concept (corpuscular corpuscular theory of light, & no waves).
18 18
th th
century century: two outstanding problems two outstanding problems
celestial bodies orbits: Lagrange, Euler & Clairaut approximate observation data
with linear combination of periodic functions; Clairaut,1754(!) first DFT formula.
vibrating strings: Euler describes vibrating string motion by sinusoids (wave
equation).
1807 1807: Fourier presents his work on heat conduction Fourier presents his work on heat conduction Fourier analysis born. Fourier analysis born.
Diffusion equation series (infinite) of sines & cosines. Strong criticism by peers
blocks publication. Work published, 1822 (Theorie Analytique de la chaleur).
A little history -2
A little history -2
19 19
th th
/ 20 / 20
th th
century century: two paths for Fourier analysis two paths for Fourier analysis - - Continuous & Discrete. Continuous & Discrete.
CONTINUOUS CONTINUOUS
Fourier extends the analysis to arbitrary function (Fourier Transform).
Dirichlet, Poisson, Riemann, Lebesgue address FS convergence.
Other FT variants born from varied needs (ex.: Short Time FT - speech analysis).
DISCRETE: Fast calculation methods (FFT) DISCRETE: Fast calculation methods (FFT)
1805 1805 - Gauss, first usage of FFT (manuscript in Latin went unnoticed!!!
Published 1866).
1965 1965 - IBMs Cooley & Tukey rediscover FFT algorithm (An algorithm for
the machine calculation of complex Fourier series).
Other DFT variants for different applications (ex.: Warped DFT - filter design &
signal compression).
FFT algorithm refined & modified for most computer platforms.
Fourier Series (FS)
Fourier Series (FS)
* * see next slide see next slide
A A periodic periodic function s(t) satisfying function s(t) satisfying Dirichlet Dirichlet s s conditions conditions * * can be expressed can be expressed
as a Fourier series, with harmonically related sine/cosine terms as a Fourier series, with harmonically related sine/cosine terms. .
[ ]
+
=
+ =
1 k
t) (k sin
k
b t) (k cos
k
a
0
a s(t)
a
0
, a
k
, b
k
: Fourier coefficients.
k: harmonic number,
T: period, = 2/T
For all t but discontinuities For all t but discontinuities
Note: {cos(kt), sin(kt) }
k
form orthogonal base of
function space.
=
T
0
s(t)dt
T
1
0
a
=
T
0
dt t) sin(k s(t)
T
2
k
b -
=
T
0
dt t) cos(k s(t)
T
2
k
a
(signal average over a period, i.e. DC term &
zero-frequency component.)
a
n
a
l
y
s
i
s
a
n
a
l
y
s
i
s
s
y
n
t
h
e
s
i
s
s
y
n
t
h
e
s
i
s
FS convergence
FS convergence
s(t) piecewise-continuous;
s(t) piecewise-monotonic;
s(t) absolutely integrable ,
<
T
0
dt s(t)
(a)
(b)
(c)
Dirichlet conditions
In any period:
Example:
square wave

T

(a)

(b)

T
s(t)
(c)
if s(t) discontinuous then
|a
k
|<M/k for large k (M>0)
Rate of convergence Rate of convergence
FS analysis - 1
FS analysis - 1
* Even & Odd functions
Odd :
s(-x) = -s(x)
x
s(x)
s(x)
x
Even :
s(-x) = s(x)
FS of odd* function: square wave.
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
2
0
0
2
1)dt ( dt
2
1
0
a =
+ =

0
0
2
dt kt cos dt kt cos
1
k
a =
=

{ }=
= =
=

k cos 1
k
2
...
0
2
dt kt sin dt kt sin
1
k
b -
=
even k , 0
odd k ,
k
4
1 2 T = =
(zero average) (zero average)
(odd function) (odd function)
... t 5 sin
5
4
t 3 sin
3
4
t sin
4
sw(t) +
+ =
[ ]
=
=
7
1 k
sin(kt)
k
b - (t)
7
sw
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
[ ]
=
=
5
1 k
sin(kt)
k
b - (t)
5
sw [ ]
=
=
3
1 k
sin(kt)
k
b - (t)
3
sw
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
[ ]
=
=
1
1 k
sin(kt)
k
b - (t)
1
sw
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
[ ]
=
=
9
1 k
sin(kt)
k
b - (t)
9
sw
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
[ ]
=
=
11
1 k
sin(kt)
k
b - (t)
11
sw
FS synthesis
FS synthesis
Square wave reconstruction Square wave reconstruction
from spectral terms from spectral terms
Convergence may be slow (~1/k) - ideally need infinite terms.
Practically, series truncated when remainder below computer tolerance
( error error). BUT BUT Gibbs Phenomenon.
Gibbs phenomenon
Gibbs phenomenon
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
[ ]
=
=
79
1 k
k 79
sin(kt) b - (t) sw
Overshoot exist @ Overshoot exist @
each discontinuity each discontinuity
Max overshoot pk-to-pk = 8.95% of discontinuity magnitude.
Just a minor annoyance.
FS converges to (-1+1)/2 = 0 @ discontinuities, in this case in this case.
First observed by Michelson, 1898. Explained by Gibbs.
FS time shifting
FS time shifting
-1.5
-1
-0.5
0
0.5
1
1.5
0 2 4 6 8 10
t
s
q
u
a
r
e

s
i
g
n
a
l
,

s
w
(
t
)
2
FS of even function: FS of even function:
/2 /2- -advanced s advanced square quare- -wave wave

f
1
3f
1
5f
1
7f
1
f
f
1
3f
1
5f
1
7f
1
f
r
k

k

4/
4/3
0
0
a =
0 =
k
b -
=
even. k , 0
11... 7, 3, k odd, k ,
k
4
9... 5, 1, k odd, k ,
k
4
k
a
(even function)
(zero average)
p
h
a
s
e
p
h
a
s
e
a
m
p
l
i
t
u
d
e
a
m
p
l
i
t
u
d
e
Note: amplitudes unchanged BUT BUT
phases advance by k/2.
Complex FS
Complex FS
Complex form of FS (Laplace 1782). Harmonics
c
k
separated by f = 1/T on frequency plot.
r
a
b
= arctan(b/a)
r = a
2
+ b
2
z = r e
j
2
e e
cos(t)
jt jt
+
=
j 2
e e
sin(t)
jt jt
=

Eulers notation:
e
-jt
= (e
jt
)* = cos(t) - jsin(t) phasor
Note Note: c
-k
= (c
k
)*
( ) ( )
k
b j
k
a
2
1
k
b j
k
a
2
1
k
c

= + =
0
a
0
c =
Link to FS real Link to FS real coeffs coeffs. .
=
=
k
t k j
e
k
c s(t)
=
T
0
dt
t k j -
e s(t)
T
1
k
c
a
n
a
l
y
s
i
s
a
n
a
l
y
s
i
s
s
y
n
t
h
e
s
i
s
s
y
n
t
h
e
s
i
s
FS properties
FS properties
Time Frequency Time Frequency
* *
Homogeneity as(t) aS(k)
Additivity s(t) + u(t) S(k)+U(k)
Linearity as(t) + bu(t) aS(k)+bU(k)
Time reversal s(-t) S(-k)
Multiplication * s(t)u(t)
Convolution * S(k)U(k)
Time shifting
Frequency shifting S(k - m)
=

m
m)U(m) S(k
t d ) t
T
0
u( ) t s(t
T
1

S(k) e
T
t k 2
j
s(t)
T
t m 2
j
e
+
) t s(t
FS - oddities
FS - oddities
k = - , -2,-1,0,1,2, + ,
k
= k,
k
=
k
t, phasor turns anti-clockwise.
Negative k phasor turns clockwise (negative phase
k
), equivalent to negative time t,
time reversal.
Negative frequencies & time reversal
Fourier components { Fourier components {u u
k k
} form orthonormal base of signal space: } form orthonormal base of signal space:
u
k
= (1/T) exp(jkt) (|k| = 0,1 2, +) Def.: Internal product :
u
k
u
m
=
k,m
(1 if k = m, 0 otherwise). (Remember (e
jt
)* = e
-jt
)
Then c
k
= (1/T) s(t) u
k
i.e. (1/T) times projection projection of signal s(t) on component u
k
Orthonormal base
=
T
o
*
m k m k
dt u u u u
Careful Careful: phases important when combining several signals!
0 50 100 150 200
k f
W
k
/W
0
10
-3
10
-2
10
-1
1
2
W
k
= 2 W
0
sync
2
(k )
W
0
= ( s
MAX
)
2

=
+ =

1 k
0
W
k
W
1
0
W W
FS - power
FS - power
FS convergence FS convergence ~1/k ~1/k
lower frequency terms
W
k
= |c
k
|
2
carry most power.
W W
k k
vs. vs.
k k
: Power density spectrum : Power density spectrum.
Example
sync(u) = sin( u)/( u)
Pulse train, duty cycle Pulse train, duty cycle = 2 = 2 / T / T

T
2
t
s(t)
b
k
= 0 a
0
= s
MAX
a
k
= 2s
MAX
sync(k )
Average power W Average power W:
s(t) s(t)
T
o
dt
2
s(t)
T
1
W =

=

+ + =
=
=
1 k
2
k
b
2
k
a
2
1
2
0
a
k
2
k
c W
Parseval Parseval s Theorem s Theorem
FS of main waveforms
FS of main waveforms
Discrete Fourier Series (DFS)
Discrete Fourier Series (DFS)
N consecutive samples of s[n] N consecutive samples of s[n]
completely describe s in time completely describe s in time
or frequency domains. or frequency domains.
DFS generate periodic c
k
with same signal period
=
=
1 N
0 k
N
n k 2
j
e
k
c s[n]
~
Synthesis: finite sum band-limited s[n]
Band-limited signal s[n], period = N.
m k,
1 N
0 n
N
-m) n(k 2
j
e
N
1
=
Kroneckers delta
Orthogonality in DFS:
s
y
n
t
h
e
s
i
s
s
y
n
t
h
e
s
i
s
=
1 N
0 n
N
n k 2
j
e s[n]
N
1
k
c
~
Note: Note: c
k+N
= c
k
same period N
i.e. time periodicity propagates to frequencies! i.e. time periodicity propagates to frequencies!
DFS defined as: DFS defined as:
~ ~ ~ ~
a
n
a
l
y
s
i
s
a
n
a
l
y
s
i
s
DFS analysis
DFS analysis
DFS of periodic discrete DFS of periodic discrete
1 1- -Volt square Volt square- -wave wave
+ =
=
otherwise ,
N
k
sin
N
kL
sin
N
N
1) (L k
j
e
2N,... N, 0, k ,
N
L
k
c
~

0.6
0 1 2 3 4 5 6 7 8 9 10 k
1
0 2 4 5 6 7 8 9 10 n
k

-0.4
0.2
0.24 0.24
0.6 0.6
0.24
1
0.24
-0.2
0.4
0.2
-0.4
-0.2
0.4
0.2
0.6
c
k

~
a
m
p
l
i
t
u
d
e
a
m
p
l
i
t
u
d
e
p
h
a
s
e
p
h
a
s
e
Discrete signals Discrete signals periodic frequency spectra. periodic frequency spectra.
Compare to continuous rectangular function
(slide # 10, FS analysis - 1)

-5 0 1 2 3 4 5 6 7 8 9 10 n
0 L N
s[n]
1
s[n]: period N N, duty factor L/N L/N
DFS properties
DFS properties
Time Frequency Time Frequency
Homogeneity as[n] aS(k)
Additivity s[n] + u[n] S(k)+U(k)
Linearity as[n] + bu[n] aS(k)+bU(k)
Multiplication * s[n] u[n]
Convolution * S(k)U(k)
Time shifting s[n - m]
Frequency shifting S(k - h)
1 N
0 h
h) - S(h)U(k
N
1
=

1 N
0 m
m] u[n s[m]
S(k) e
T
m k 2
j
s[n]
T
t h 2
j
e
+
DFT Window characteristics
DFT Window characteristics
Finite discrete sequence spectrum convoluted with rectangular window spectrum.
Leakage amount depends on chosen window & on how signal fits into the window.
Resolution: capability to distinguish different tones. Inversely proportional to main-
lobe width. Wish: as high as possible. Wish: as high as possible.
(1)
(1)
Several windows used ( Several windows used (application application- -
dependent dependent): Hamming, ): Hamming, Hanning Hanning, ,
Blackman, Kaiser ... Blackman, Kaiser ...
Rectangular window
Peak-sidelobe level: maximum response outside the main lobe.
Determines if small signals are hidden by nearby stronger ones.
Wish: as low as possible. Wish: as low as possible.
(2)
(2)
Sidelobe roll-off: sidelobe decay
per decade. Trade-off with (2).
(3)
(3)
Sampled sequence
In time it reduces end-
points discontinuities.
Non
windowed
Windowed
DFT of main windows
DFT of main windows
Windowing reduces leakage by
minimising sidelobes magnitude.
Some window functions
DFT - Window choice
DFT - Window choice
Window type -3 dB Main-
lobe width
[bins]
-6 dB Main-
lobe width
[bins]
Max sidelobe
level
[dB]
Sidelobe roll-off
[dB/decade]
Rectangular
0.89 1.21 -13.2 20
Hamming
1.3 1.81 - 41.9 20
Hanning
1.44 2 - 31.6 60
Blackman
1.68 2.35 -58 60
Common windows characteristics
NB: Strong DC component can shadow nearby small signals. Remove NB: Strong DC component can shadow nearby small signals. Remove it! it!
Far & strong interfering components high roll-off rate.
Near & strong interfering components small max sidelobe level.
Accuracy measure of single tone wide main-lobe
Observed signal Observed signal Window wish list Window wish list
DFT - Window loss remedial
DFT - Window loss remedial
Smooth data Smooth data- -tapering windows cause information loss near edges. tapering windows cause information loss near edges.
Attenuated inputs get next
windows full gain & leakage
reduced.
Usually 50% or 75% overlap
(depends on main lobe width).
Drawback: increased
total processing time.
Solution:
sliding (overlapping) DFTs.
2 x N samples (input signal)
DFT #1
DFT #2
DFT #3
DFT AVERAGING
DFT - parabolic interpolation
DFT - parabolic interpolation
Parabolic interpolation often enough to find position of
peak (i.e. frequency).
Other algorithms available depending on data.
198 199 200 201 202 203
1.962
1.963
1.964
1.965
1.966
1.967
1.968
199 200 201 202 203 204
0.974
0.975
0.976
0.977
Rectangular window
Hanning window
Systems spectral analysis (hints)
Systems spectral analysis (hints)
System analysis: measure input System analysis: measure input- -output relationship output relationship. .

DIGITAL LTI
SYSTEM
h[n]
x[n] y[n]
H(f H(f) : LTI transfer function ) : LTI transfer function
=
= =
0 m
h[m] m] x[n h[n] x[n] y[n] x[n] h[n]
X(f) H(f) Y(f) = X(f) H(f)
DIGITAL
LTI
SYSTEM
0 n
[n]
1
0 n
h[n]
h[t] = impulse response
Linear Time Invariant Linear Time Invariant
y[n] predicted from { x[n], h[t] }
Transfer function can be estimated by Y(f) / X(f)
Estimating H(f) (hints)
Estimating H(f) (hints)
(f) X X(f) (f) G
*
xx
=
Power Spectral Density of x[t]
(FT of autocorrelation).
(f) X Y(f) (f) G
*
yx
=
Cross Power Spectrum of x[t] & y[t]
(FT of cross-correlation).
It is a check on
H(f) validity!
xx
yx
*
*
G
G
(f) X X(f)
(f) X Y(f)
X(f)
Y(f)
H(f) =
= =
Transfer Function
(ex: beam !)

Analog & Digital Signal

Diunggah oleh

Informasi Dokumen

Judul Asli

Hak Cipta

Format Tersedia

Bagikan dokumen Ini

Bagikan atau Tanam Dokumen

Opsi Berbagi

Apakah menurut Anda dokumen ini bermanfaat?

Apakah konten ini tidak pantas?

Hak Cipta:

Format Tersedia

Analog & Digital Signal

Diunggah oleh

Hak Cipta:

Format Tersedia

Slides adapted from ME Angoletta, CERN

Anda mungkin juga menyukai