ELEN E4810: Digital Signal Processing
Topic 11:
Continuous Signals
1. Sampling and Reconstruction
2. Quantization
Dan Ellis
2013-12-04
1
1. Sampling & Reconstruction
x(t)
DSP must interact with an analog world:
A to D
ADC
x[n]
DSP
y[n]
Anti- Sample
alias and
filter hold
Sensor
Dan Ellis
D to A
y(t)
DAC
Reconstruction
filter
WORLD
2013-12-04
Actuator
2
Sampling: Frequency Domain
Sampling: CT signal → DT signal by
recording values at ‘sampling instants’:
Discrete
g[n] = ga (nT )
Continuous
Sampling period T
→ samp.freq. ≠samp = 2º/T rad/sec
What is the relationship of the spectra?
≠ in rad/second
i.e. relate Ga ( j) =
and
Dan Ellis
jt
g
t
e
a ( ) dt
G(e ) = g[n]e
j
CTFT
jn
DTFT
! in rad/sample
2013-12-04
3
Sampling
DT signals have same ‘content’ as
CT signals gated by an impulse train:
ga(t)
gp(t)
×
t
t
CT delta fn
p(t)
=
n=
(t nT )
T
t
‘sampled’ signal:
still continuous
- but discrete
values
gp(t) = ga(t)·p(t) is a CT signal with the
same information as DT sequence g[n]
Dan Ellis
2013-12-04
4
Spectra of sampled
signals
⇥
Given CT gp (t) =
CTFT Spectrum
n= ⇥
Gp (j ) = F{gp (t)} =
ga (nT ) · (t
by linearity
ga (nT )F{ (t nT )}
n
Gp (j ) =
Compare to DTFT
ga (nT )e
⇥n
G(ej ) =
g[n]e
⇥n
i.e. G(ej ) = Gp (j )|
Dan Ellis
nT )
T=
2013-12-04
j nT
j n
≠ ~ !/T
º/T = ≠samp/2
! ~ ≠T
º
5
Spectra of sampled signals
Also, note that
p(t ) = (t nT )
n
is periodic, thus has Fourier Series:
1
ck =
T
1
=
T
1
j ( 2T ) kt
p(t ) = e
T k=
j2 kt / T
p
t
e
dt
(
)
T / 2
T /2
shift in
But F {e j0 t x (t )} = X ( j ( 0 )) frequency
(
so G p ( j) = T1 k Ga j ( k samp )
)
- scaled sum of replicas of Ga(j≠)
shifted by multiples of sampling frequency ≠samp
Dan Ellis
2013-12-04
6
CT and DT Spectra
j
1
k 2
So: G (e
=
G
j
=
G
j
(
)
(
(
) p T = T k a T T ))
DTFT G ( e jT ) = 1
CTFT
G
j
k
(
)
(
)
T
a
samp
k
ga (t ) Ga ( j)
−≠M
×
p(t ) P ( j)
=
−2≠samp
( )
g[n] G e j
Dan Ellis
−2º
=
2≠samp
shifted/scaled copies
≠
2
samp
T
2013-12-04
have = T = 2
At
≠
≠samp
M/T
−4º
≠
≠M
−≠samp
g p (t ) G p ( j)
↓
ga(t) is bandlimited
@ ≠Μ
M
=
2º
4º
7
!
Avoiding aliasing
Sampled analog signal has spectrum:
Gp(j≠)
Ga(j≠)
−≠samp −≠M
−≠samp + ≠M
≠M
“alias” of “baseband”
signal
≠
≠samp
≠samp− ≠M
ga(t) is bandlimited to ± ≠M rad/sec
When sampling frequency ≠samp is large...
→ no overlap between aliases
→ can recover ga(t) from gp(t)
by low-pass filtering
Dan Ellis
2013-12-04
8
Aliasing & The Nyquist Limit
If bandlimit ≠M is too large, or sampling
rate ≠samp is too small, aliases will overlap:
Gp(j≠)
−≠M
−≠samp
≠M ≠samp
≠samp− ≠M
≠
Spectral effect cannot be filtered out
→ cannot recover ga(t)
Sampling theorum
Avoid by: samp M M samp 2 M
i.e. bandlimit ga(t) at ≤ samp Nyquist
2 frequency
Dan Ellis
2013-12-04
9
Anti-Alias Filter
To understand speech, need ~ 3.4 kHz
→ 8 kHz sampling rate (i.e. up to 4 kHz)
‘space’ for
Limit of hearing ~20 kHz
filter rolloff
→ 44.1 kHz sampling rate for CDs
Must remove energy above Nyquist with
LPF before sampling: “Anti-alias” filter
A to D
ADC
Anti-alias Sample
filter
& hold
Dan Ellis
2013-12-04
10
M
Sampling Bandpass Signals
Signal is not always in ‘baseband’
around ≠ = 0 ... may be at higher ≠:
A
−≠H −≠L
≠L ≠H
≠
Bandwidth ¢≠ = ≠H − ≠L
If aliases from sampling don’t overlap,
no aliasing distortion; can still recover:
A/T
−≠samp
≠samp/2 ≠samp
≠
2≠samp
Basic limit: ≠samp/2 ≥ bandwidth ¢≠
Dan Ellis
2013-12-04
11
Reconstruction
To turn g[n]
back to g^a(t):
g[n]
n
make a continuous
impulse train g^ p(t)
lowpass filter to
extract baseband
→ g^ a(t)
G(ej!)
º
!
G^ p(j≠)
g^
p(t)
t
^
Ga(j≠)
^
ga(t)
≠samp/2
≠
≠
t
Ideal reconstruction filter is brickwall
Dan Ellis
i.e. sinc - not realizable (especially analog!)
use something with finite transition band...
2013-12-04
12
2. Quantization
Course so far has been about
discrete-time i.e. quantization of time
Computer representation of signals also
quantizes level (e.g. 16 bit integer word)
Level quantization introduces an error
between ideal & actual signal → noise
Resolution (# bits) affects data size
→ quantization critical for compression
Dan Ellis
smallest data
coarsest quantization
2013-12-04
13
Quantization
x
Quantization is performed
in A-to-D:
x
t
A/D
n
T
Quantization has simple transfer curve:
5
Q{x} 4
3
2
1
0
-1
-2
-3
-4
-5
-5 -4 -3 -2 -1 0 1 2 3 4 5
Quantized signal
e = x - Q{x}
Quantization error
e = x xˆ
x
Dan Ellis
xˆ = Q{x}
2013-12-04
14
Quantization noise
Can usually model quantization as
additive white noise:
i.e. uncorrelated
with self or signal x
x[n]
+
-
^
x[n]
e[n]
bits ‘cut off’ by
quantization;
hard amplitude limit
Dan Ellis
2013-12-04
15
Quantization SNR
Common measure of noise is
Signal-to-Noise ratio (SNR) in dB:
signal power
2x
SNR = 10 log10 2 dB
e
noise power
When |x| >> 1 LSB, quantization noise
has ~ uniform distribution:
P(e[n])
2
e2 =
12
Dan Ellis
- /
2
+ /
2
(quantizer step = ")
2013-12-04
16
Quantization SNR
Now, σx2 is limited by dynamic range of
converter (to avoid clipping)
e.g. b+1 bit resolution (including sign)
output levels vary -2b·" .. (2b-1)"
–R
R
FS
! =
! !
where full-scale range
...! FS
b+1
2
2
RFS = 2
SNR = 10 log10
2
x
2
RFS
6b + 16.8
22(b+1) ·12
Dan Ellis
i.e. ~ 6 dB
SNR per bit
2013-12-04
20 log10
RFS
depends
on signal
17
x
Coefficient Quantization
Quantization affects not just signal
but filter constants too
.. depending on implementation
.. may have different resolution
Some coefficients
are very sensitive
to small changes
Dan Ellis
e.g. poles near
unit circle
high-Q pole
becomes
unstable
z-plane
2013-12-04
18
12/9 Project Presentations
10:15 Arthur Argall: Genomic Signal Processing
10:25 Yue Hou, Dongxue Liu:
Blind Signal Separation
10:35 Nathan Lin:
Spectral Domain Phase Microscopy
10:45 Minda Yang, Yitong Li:
Reconstruction from Neural Measurements
10:55 Yinan Wu: Mixed Channel Music
11:05 Maja Rudolph, Preston Conley:
Walking Pace Extraction from Video
11:15 Alan Zambeli-Ljepović, Tanya Shah, Sophie Wang:
Lung Sounds Analysis
Dan Ellis
2013-12-04
19