Analog & Digital Signals

Analog & digital signals
Analog
Digital
Discrete function Vk of
discrete sampling variable tk,
with k = integer: Vk = V(tk).
0.3
0.3
0.2
0.2
Voltage [V]
Voltage [V]
Continuous function V of
continuous variable t (time,
space etc) : V(t).
0.1
0
-0.1
-0.2
0.1
0
ts ts
-0.1
-0.2
4
6
time [ms]
10
6
8
2
4
sampling time, tk [ms]
Uniform (periodic) sampling.

Sampling frequency fS = 1/ tS
Slides adapted from ME Angoletta, CERN
10
Digital vs analog procing

Digital Signal Processing (DSPing)
Advantages
Limitations
Often easier system upgrade.
A/D & signal processors speed:

wide-band signals still difficult to
treat (real-time systems).
Data easily stored.
Finite word-length effect.
Better control over accuracy

requirements.
Obsolescence (analog
electronics has it, too!).
More flexible.
Reproducibility.
Digital system example

General scheme
ms
Antialiasing
Sometimes steps missing
ms
A
(ex: economics);
- D/A + filter
(ex: digital output wanted).
A
k
V
Digital
Processing
D/A
Filter
Reconstruction
ms
ANALOG
DOMAIN
ms
A/D
DIGITAL
DOMAIN
- Filter + A/D
Filter
ANALOG
DOMAIN
Digital system implementation

ANALOG INPUT
Antialiasing
Filter
KEY DECISION POINTS:

Analysis bandwidth, Dynamic range
Pass / stop bands.
Sampling rate.
A/D
No. of bits. Parameters.
Digital
Processing
DIGITAL OUTPUT
Digital format.
What to use for processing?
See slide DSPing aim & tools
2
3
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.
Train starts
wheels go clockwise.
Train accelerates
wheels go counter-clockwise.
Why?
Frequency misidentification due to low sampling frequency.
* Sampling: independent variable (ex: time) continuous
discrete.
Quantisation: dependent variable (ex: voltage) continuous discrete.

Here well talk about uniform sampling.
Sampling - 2
1.2
__ s(t) = sin(2f t)
0
0.8
0.6
s(t) @ fS
0.4
0.2
f0 = 1 Hz, fS = 3 Hz
tt
-0.2
-0.4
-0.6
__ s (t) = sin(8f t)
0
1
-0.8
-1
__ s (t) = sin(14f t)
0
2
-1.2
s(t) @ fS represents exactly all sine-waves sk(t) defined by:

sk (t) = sin( 2 (f0 + k fS) t ) , k
The sampling theorem
A signal s(t) with maximum frequency fMAX can be

Theo* recovered if sampled at frequency f > 2 f
S
MAX .
* Multiple proposers: Whittaker(s), Nyquist, Shannon, Kotelnikov.
Naming gets
confusing !
Nyquist frequency (rate) fN = 2 fMAX or fMAX or fS,MIN or fS,MIN/2
Example
s(t) = 3 cos(50 t) + 10 sin(300 t) cos(100 t)
F1
F2
F1=25 Hz, F2 = 150 Hz, F3 = 50 Hz
Condition on fS?
F3
fS > 300 Hz
fMAX
Frequency domain (hints)
Time & frequency:

frequency two complementary signal descriptions.
Signals seen as projected onto time or frequency domains.
Example
Ear + brain act as frequency analyser: audio spectrum
split into many narrow bands
low-power sounds
detected out of loud background.
Bandwidth:
Bandwidth indicates rate of change of a signal.
High bandwidth
signal changes fast.
Sampling low-pass signals

Continuous spectrum
(a)
(a) Band-limited signal:

frequencies in [-B, B] (fMAX = B).
-B
(b)
Discrete spectrum
No aliasing
(b) Time sampling
frequency
repetition.
fS > 2 B
-B
B fS/2
Discrete spectrum
Aliasing & corruption
(c)
fS/2
no aliasing.
(c) fS
f
2B
aliasing !
Aliasing: signal ambiguity

in frequency domain
Antialiasing filter
1
(a)
(a),(b) Out-of-band noise can alias
Signal of interest
Out of band
noise
Out of band
noise
-B
(b)
into band of interest. Filter it before!
(c) Antialiasing filter

Passband: depends on bandwidth of
interest.
Attenuation AMIN : depends on

(c)
-B
ADC resolution ( number of bits N).
B fS/2
AMIN, dB ~ 6.02 N + 1.76

Antialiasing
filter
Passband
frequency
Out-of-band noise magnitude.

Other parameters: ripple, stopband
frequency...
-B
ADC - Number of bits N
Continuous input signal digitized into 2N levels.

Uniform, bipolar transfer function (N=3)
1113
2
Quantization step q =
1
0
-4
-3
-2
-1
Ex: VFSR = 1V , N = 12
-1
V FSR
2N
q = 244.1 V
010
-2
001
-3
Voltage ( = q)
000
-4
LSB
VFSR
Scale factor (= 1 / 2N )
Percentage (= 100 / 2N )
q/2
0.5
-4
-3
-2
-1
-0.5
-q/2
Quantisation error
-1
ADC - Quantisation error
2
0.3
Quantisation Error eq in
[-0.5 q, +0.5 q].
Voltage [V]
0.2
0.1
0
0
-0.1
-0.2
time [ms]
10
eq limits ability to resolve

small signal.
Higher resolution means
lower eq.
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
analysis
General Transform as
problem-solving tool
frequency, f
F
S(f) = F[s(t)]
s(t)
s(t), S(f) :
Transform Pair
synthesis
Fourier analysis - tools

Input Time Signal
Frequency spectrum
2.5
2
1.5
1
Periodic
0.5
0
0
time, t
Continuous
2.5
(period T)
Aperiodic
2
1.5
1
FS
Discrete
FT
Continuous
T
1
c k = s(t) e j k t dt
T
0
j2 f t
+
S(f) = s(t) e
dt
0.5
0
0
time, t
10
12
2.5
2
Periodic
1.5
1
0.5
(period T)
0
0
time, tk
Discrete
2.5
Aperiodic
2
1.5
1
0.5
0
0
time, tk
6
10
12
2kn
N
1
j
1
~
N
ck = s[n] e
N
n =0
DFS** Discrete
DTFT
Continuous
DFT** Discrete
Note: j =-1, = 2/T, s[n]=s(tn), N = No. of samples
S(f) =
s[n] e j 2 f n
n=
j
1 N1
~
ck = s[n] e
N
n =0
**
2kn
N
Calculated via FFT
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 theory of light, & no waves).
18th century:
century 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 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
19th / 20th century:
century two paths for Fourier analysis - Continuous & Discrete.
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)
1805 - Gauss, first usage of FFT (manuscript in Latin went unnoticed!!!

Published 1866).
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)

A periodic function s(t) satisfying Dirichlets conditions * can be expressed
as a Fourier series, with harmonically related sine/cosine terms.
is
s
+
he
t
n s(t) = a0 + [ak cos (k t) bk sin (k t)]
sy
k =1 For all t but discontinuities
a0, ak, bk : Fourier coefficients.

k: harmonic number,
T: period, = 2/T
s
si
y
T
al
n
1
(signal average over a period, i.e. DC term &
a a = s(t)dt
0
zero-frequency component.)
T
0
T
2
ak = s(t) cos(k t) dt
Note: {cos(kt), sin(kt) }k
T
0
form orthogonal base of
T
function space.
2
- bk = s(t) sin(k t) dt
T
0
* see next slide
FS convergence
Dirichlet conditions
(a) s(t) piecewise-continuous;
(b) s(t) piecewise-monotonic;
In any period:
(c) s(t) absolutely integrable ,
s(t) dt <
Example:
square wave
Rate of convergence
if s(t) discontinuous then

|ak|<M/k for large k (M>0)
s(t)
T
(a)
(b)
(c)
FS analysis - 1
T = 2 = 1
2
a0 =
dt + ( 1)dt = 0
2
0
2
ak = cos kt dt cos kt dt = 0

(zero average)
(odd function)
2
1
{ 1 cos k } =
- bk = sin kt dt sin kt dt = ... =
k
4
k , k odd
=
0 , k even
4
4
4
sw(t) = sin t +
sin 3 t +
sin 5 t + ...
3
5
1.5
square signal, sw(t)
FS of odd* function: square wave.
1
0.5
0
-0.5
10
-1
-1.5
* Even & Odd functions

s(x)
Even :
s(-x) = s(x)
x
s(x)
Odd :
s(-x) = -s(x)
FS synthesis
Square wave reconstruction
from spectral terms
1.5
7
3
15
911
sw1
(t)
sin(kt)
(t)===
sin(kt)
sin(kt)
]]]]
[[--[b-bkbkksin(kt)
7
3
5
11
9(t)
kkk==1=11
1
0.5
0
-0.5
-1
-1.5
0
Convergence may be slow (~1/k) - ideally need infinite terms.

Practically, series truncated when remainder below computer tolerance
( error).
error BUT Gibbs Phenomenon.
10
Gibbs phenomenon
1.5
sw 79 (t) =
79
[- bk sin(kt)]
k =1
Overshoot exist @
each discontinuity
0.5
0
-0.5
-1
-1.5
0
First observed by Michelson, 1898. Explained by Gibbs.

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.
case
10
FS time shifting
4
k , k odd, k = 1, 5, 9...
ak = 4
, k odd, k = 3, 7, 11...
k
0
, k even.
- bk = 0
(even function)
Note: amplitudes unchanged BUT

phases advance by k/2.
(zero average)
1
0.5
0
-0.5
10
-1
-1.5
rk
4/
4/3
f1
3f1
5f1
7f1
f1
3f1
5f1
7f1
ph
as
e
a 0= 0
1.5
am
pl
it
ud
e
FS of even function:
/2-advanced square-wave
Complex FS
Eulers notation:
e-jt = (ejt)* = cos(t) - jsin(t)
phasor
e jt + e jt
cos(t) =
2
e jt e jt
sin(t) =
2 j
s
T
si
1
y
l
a c k = s(t) e - j k t dt
n
a
T
is
s
he
t
n
sy s(t) =
Complex form of FS (Laplace 1782). Harmonics

ck separated by f = 1/T on frequency plot.
jk t
c
e
k
k =
z=re
Note:
Note c-k = (ck)*
Link to FS real coeffs.
c 0 = a0
ck =
1
1
(ak + j bk ) = (a k j b k )
2
2
r = a2 + b2
= arctan(b/a)
FS properties
Time
Homogeneity
as(t)
Additivity
s(t) + u(t)
Linearity
as(t) + bu(t)
Time reversal
Multiplication *
Convolution *
Time shifting
Frequency
aS(k)
S(k)+U(k)
aS(k)+bU(k)
s(-t)
S(-k)
s(t)u(t)
T
1
s(t t ) u( t ) dt
T
0
s(t t )
Frequency shifting e
+j
2 m t
T s(t)
S(k m)U(m)
m =
S(k)U(k)
e
2 k t
T
S(k)
S(k - m)
FS - oddities
Orthonormal base
Fourier components {uk} form orthonormal base of signal space:
T
*
uk = (1/T) exp(jkt) (|k| = 0,1 2, +) Def.: Internal product : uk um = uk um
dt
uk um = k,m (1 if k = m, 0 otherwise).
(Remember (ejt)* = e-jt )
Then ck = (1/T) s(t) uk i.e. (1/T) times projection of signal s(t) on component uk
Negative frequencies & time reversal

k = - , -2,-1,0,1,2, + ,
k = k, k = kt, phasor turns anti-clockwise.
Negative k phasor turns clockwise (negative phase k ), equivalent to negative time t,

time reversal.
Careful:
Careful phases important when combining several signals!
FS - power
Average power W :
1
W =
T
Parsevals Theorem
W=
ck
k =
1
= a0 2 +
2
s(t) 2 dt s(t) s(t)
FS convergence ~1/k
Example
Pulse train, duty cycle = 2 / T
s(t)
2
T
bk = 0
a0 = sMAX
ak = 2sMAX sync(k )
lower frequency terms
ak 2 + bk 2
k =1
Wk = |ck|2 carry most power.

Wk vs. k: Power density spectrum.
spectrum
2
1
10
-1
10
-2
10
-3
Wk/W0
Wk = 2 W0 sync2(k )
kf
0
50
W0 = ( sMAX)2
sync(u) = sin( u)/( u)
100
150
200
W = W0 1+ k
k =1 W0
FS of main waveforms
Discrete Fourier Series (DFS)

Band-limited signal s[n], period = N.
DFS defined as:

is
2 k n
s
y
N
1
l
j
a
1
N
an ~
ck =
s[n] e
n =0
~
~
Note: ck+N = ck same period N
i.e. time periodicity propagates to frequencies!
s
si
e
2 k n
th
N
1
j
n
~
sy s[n] =
ck e N
k =0
DFS generate periodic ck

with same signal period
Orthogonality in DFS:
2 n(k -m)
N
1
j
1
N
e
= k,m
N
n =0
Kroneckers delta
N consecutive samples of s[n]

completely describe s in time
or frequency domains.
Synthesis: finite sum band-limited s[n]
DFS analysis
DFS of periodic discrete
1-Volt square-wave
s[n]: period N, duty factor L/N
0 1 2 3 4 5 6 7 8 9 10
0
L
N
am
pl
it
ud
e
-5
ck
0.24
0.24
0.2
0.6
0.6
0.6
0.6
0.24
0.24
0 1 2 3 4 5 6 7 8 9 10
ph
as
e
,
k = 0, + N, 2N,...
~
ck =
k (L 1)
kL
j
sin
N
N
e
, otherwise
N
k
sin
s[n]
0.4
0.2
Discrete signals periodic frequency spectra.

Compare to continuous rectangular function
(slide # 10, FS analysis - 1)
0.4
0.2
4 5 6 7 8 9 10
-0.2
-0.4
n
-0.2
-0.4
DFS properties
Time
Homogeneity
as[n]
Additivity
s[n] + u[n]
Linearity
as[n] + bu[n]
Multiplication *
s[n] u[n]
N1
Frequency
aS(k)
S(k)+U(k)
aS(k)+bU(k)
1 N1
S(h)U(k - h)
N h=0
s[m] u[n m]
Convolution *
S(k)U(k)
m =0
Time shifting
s[n - m]
Frequency shifting
+j
2 h t
T s[n]
2 k m
T
S(k)
S(k - h)
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.
(1) Resolution: capability to distinguish different tones. Inversely proportional to mainlobe width. Wish: as high as possible.
(2) Peak-sidelobe level: maximum response outside the main lobe.

Determines if small signals are hidden by nearby stronger ones.
Wish: as low as possible.
(3) Sidelobe roll-off: sidelobe decay
(2)
(1)
(3)
per decade. Trade-off with (2).
Several windows used (applicationdependent): Hamming, Hanning,

Blackman, Kaiser ...
Rectangular window
DFT of main windows

Windowing reduces leakage by
minimising sidelobes magnitude.
In time it reduces endpoints discontinuities.
Sampled sequence
Non
windowed
Windowed
Some window functions

DFT - Window choice

Common windows characteristics
Window type
-3 dB Mainlobe width
-6 dB Mainlobe width
[bins]
1.21
Max sidelobe
level
Sidelobe roll-off
[dB/decade]
Rectangular
[bins]
0.89
Hamming
1.3
1.81
- 41.9
20
Hanning
1.44
- 31.6
60
Blackman
1.68
2.35
-58
60
Observed signal
[dB]
-13.2
20
Window wish list
Far & strong interfering components
Near & strong interfering components
Accuracy measure of single tone
high roll-off rate.

small max sidelobe level.
wide main-lobe
NB: Strong DC component can shadow nearby small signals. Remove it!
DFT - Window loss remedial

Smooth data-tapering windows cause information loss near edges.
Solution:
sliding (overlapping) DFTs.
Attenuated inputs get next
windows full gain & leakage
reduced.
2 x N samples (input signal)

DFT #1
DFT #2
DFT #3
Usually 50% or 75% overlap

(depends on main lobe width).
Drawback: increased
total processing time.
DFT AVERAGING
DFT - parabolic interpolation

Rectangular window
Hanning window
1.968
0.977
1.967
1.966
0.976
1.965
1.964
0.975
1.963
1.962
198
199
200
201
202
203
0.974
199
200
201
202
Parabolic interpolation often enough to find position of

peak (i.e. frequency).
Other algorithms available depending on data.
203
204
Systems spectral analysis (hints)

System analysis: measure input-output relationship.
Linear Time Invariant
x[n]
DIGITAL LTI
SYSTEM
[n]
y[n]
h[n]
x[n]
h[n]
y[n] = x[n] h[n] =
DIGITAL
LTI
SYSTEM
h[n]
0
h[t] = impulse response
x[n m] h[m]
y[n] predicted from { x[n], h[t] }
m =0
X(f)
H(f)
Y(f) = X(f) H(f)
H(f) : LTI transfer function
Transfer function can be estimated by Y(f) / X(f)
Estimating H(f)
G xx (f) = X(f) X* (f)
G yx (f) = Y(f) X* (f)
(hints)
Power Spectral Density of x[t]

(FT of autocorrelation).
Cross Power Spectrum of x[t] & y[t]

(FT of cross-correlation).
Y(f) Y(f) X* (f) G yx

H(f) =
=
=
*
X(f) X(f) X (f) G xx
Transfer Function
(ex: beam !)
It is a check on
H(f) validity!

Analog & Digital Signals

Uploaded by

Document Information

Original Title

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Analog & Digital Signals

Uploaded by

Copyright:

Available Formats

Analog & digital signals

Uniform (periodic) sampling.

Slides adapted from ME Angoletta, CERN

Digital vs analog procing

Often easier system upgrade.

A/D & signal processors speed:

Data easily stored.

Finite word-length effect.

Better control over accuracy

Slides adapted from ME Angoletta, CERN

Digital system example

Sometimes steps missing

Slides adapted from ME Angoletta, CERN

Digital system implementation

KEY DECISION POINTS:

Pass / stop bands.

No. of bits. Parameters.

Slides adapted from ME Angoletta, CERN

How fast must we sample * a continuous

* Sampling: independent variable (ex: time) continuous

Quantisation: dependent variable (ex: voltage) continuous discrete.

Slides adapted from ME Angoletta, CERN

s(t) @ fS represents exactly all sine-waves sk(t) defined by:

Slides adapted from ME Angoletta, CERN

The sampling theorem

A signal s(t) with maximum frequency fMAX can be

Nyquist frequency (rate) fN = 2 fMAX or fMAX or fS,MIN or fS,MIN/2

F1=25 Hz, F2 = 150 Hz, F3 = 50 Hz

Frequency domain (hints)

Time & frequency:

Slides adapted from ME Angoletta, CERN

Sampling low-pass signals

(a) Band-limited signal:

(b) Time sampling

Aliasing: signal ambiguity

Slides adapted from ME Angoletta, CERN

(a),(b) Out-of-band noise can alias

into band of interest. Filter it before!

(c) Antialiasing filter

Attenuation AMIN : depends on

ADC resolution ( number of bits N).

AMIN, dB ~ 6.02 N + 1.76

Out-of-band noise magnitude.

Slides adapted from ME Angoletta, CERN

ADC - Number of bits N

Continuous input signal digitized into 2N levels.

Slides adapted from ME Angoletta, CERN

ADC - Quantisation error

eq limits ability to resolve

Slides adapted from ME Angoletta, CERN

Frequency analysis: why?

Fourier analysis - tools

Note: j =-1, = 2/T, s[n]=s(tn), N = No. of samples

Calculated via FFT

Slides adapted from ME Angoletta, CERN

Slides adapted from ME Angoletta, CERN

Fourier extends the analysis to arbitrary function (Fourier Transform).

Dirichlet, Poisson, Riemann, Lebesgue address FS convergence.

DISCRETE: Fast calculation methods (FFT)

1805 - Gauss, first usage of FFT (manuscript in Latin went unnoticed!!!

FFT algorithm refined & modified for most computer platforms.

Slides adapted from ME Angoletta, CERN

Fourier Series (FS)