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Interpolation
Yao Wang
Polytechnic University, Brooklyn, NY11201
http://eeweb.poly.edu/~yao
Outline
1
T=0.1
Q=0.25
0.5
-0.5
-1
Sampling Quanti-
zation
xc(t) x[n] = xc(nT) x[ n]
Sampling Quantization
Period Interval
T Q
• Sampling: take samples at time nT
– T: sampling period;
x[ n] = x( nT ),−∞ < n < ∞
– fs = 1/T: sampling frequency
• Theorem:
– If xc(t) is bandlimited, with maximum frequency fm (or ωm =2π fm )
– and if fs =1/T > 2 fm or ωs =2π /T >2 ωm
– Then xc(t) can be reconstructed perfectly from x[n]= xc(nT) by
using an ideal low-pass filter, with cut-off frequency at fs/2
– fs0 = 2 fm is called the Nyquist Sampling Rate
• Physical interpretation:
– Must have at least two samples within each cycle!
Reconstructed
=original
Sampling under
Nyquist rate
ωs=1.5ωm<ωs0
Reconstructed
\= original
Aliasing: The reconstructed sinusoid has a lower frequency than the original!
Original signal
Sampling
impulse train The spectrum of the
sampled signal includes
the original spectrum and
its aliases (copies) shifted
Sampled signal to k fs , k=+/- 1,2,3,…
ωs>2 ωm The reconstructed signal
from samples has the
frequency components
upto fs /2.
Sampled signal
ωs<2 ωm When fs< 2fm , aliasing
(Aliasing effect) occur.
Spectrum of
cos(2πf0t)
-f0 0 f0
No aliasing
fs >2f0
fs -f0 >f0
Reconstructed
-fs -f0 -fs -fs+f0 -f0 0 f0 fs-f0 fs fs+f0
signal: f0
-fs/2 fs/2
With aliasing
f0<fs <2f0 (folding) 0
fs -f0 <f0
Reconstructed signal: fs -f0 -fs -f0 -fs -f0 -fs+f0 fs-f0 f0 fs fs+f0
With aliasing
fs <f0 (aliasing) -fs
0 fs
f0-fs <f0
Reconstructed signal: fs -f0 -fs -f0 -f0 -f0+fs f0-fs f0 fs+f0
Fmax=1/Tmin
Tmin
Pre-Filter Periodic
H (f) Sampling
x(t) x’(t) xd(n)
Sampling
period T
1.2
0.2
0
0 1 2 3 4 5 6 7 8 9 10
Original signal
Sampled signal
ωs>2 ωm
Ideal
reconstruction filter
(low-pass)
Reconstructed signal
(=Original signal)
Hr(f)
T
f
- fs/2 fs/2
T | f |< f s / 2 sin π t / T
H r ( f )= hr (t ) =
0 otherwise π t /T
∞ ∞
sin[ π (t − nT ) / T ]
xr (t ) = xs (t ) ∗ hr (t ) = ∑ x[n] hr (t − nT ) = ∑ x[n]
π (t − nT ) / T
n =−∞ n =−∞
1.2
0.8
Down-sampling by a
0.6
factor of 2 = take
0.4
every other sample
0.2
0 To avoid aliasing of
0 1 2 3 4 5 6 7 8 9 10
any high frequency
content in the
original signal,
1.2
should smooth the
1
original signal before
0.8 down-sampling --
0.6 Prefiltering
0.4
0.2
0
0 1 2 3 4 5 6 7 8 9 10
Low-pass filter
Gain: M M
Cut-off: 1/M
x [n] x’[n] xd [n] = x [nM]
1.2
0.8
0.6
0.4
0.2
0
0 1 2 3 4 5 6 7 8 9 10
1.2
0.8
0.6
0.4
0.2
0
0 1 2 3 4 5 6 7 8 9 10
Missing samples need to be filled from neighboring available samples using interpolation filter
Lowpass Filter
L Gain = L
Cutoff = 1 / L
x[n] xe [n] xi [n]
y psd-y
0.4 0
0.2
-20
0
-0.2 -40
-0.4
-60
5000 10000 15000 0 5000 10000
0.2 -20
-20 0.4
-40
0
0.2
-40 -60
-0.2
0 -80
-0.4
-60
2000 4000 6000 0 5000 10000 -10 0 10 0 0.2 0.4 0.6 0.8
-0.2 -40
0
-0.4 -100
-60
5000 10000 15000 0 5000 10000 -5 0 5 0 0.2 0.4 0.6 0.8
y psd-y
0.4 0
0.2
-20
0
-0.2 -40
-0.4
-60
5000 10000 15000 0 5000 10000
0.2 -10
-20 0.4
-20
0
0.2 -30
-40
-0.2
-40
0
-0.4 -60 -50
2000 4000 6000 0 5000 10000 0 0.2 0.4 0.6 0.8
y psd-y
0.4 0
0.2
-20
0
-0.2 -40
-0.4
-60
5000 10000 15000 0 5000 10000
0.2
-20 0.4
0 -50
0.2
-40
-0.2
0
-60 -100
1000 2000 3000 0 5000 10000 -10 0 10 0 0.2 0.4 0.6 0.8
y psd-y
0.4 0
0.2
-20
0
-0.2 -40
-0.4
-60
5000 10000 15000 0 5000 10000
-10
0.2 0.4
-20 -20
0 -30
0.2
-40
-0.2 -40
0
-50
-60
1000 2000 3000 0 5000 10000 -1012 0 0.2 0.4 0.6 0.8
• Sampling:
– Know the minimally required sampling rate:
• fs > 2 fmax , Ts < T0,min /2
• Can estimate T0,min from signal waveform
• Interpolation:
– Can illustrate sample-and-hold and linear interpolation from
samples.
– Understand why the ideal interpolation filter is a lowpass filter with
cutoff frequency at fs /2.
– Know the ideal interpolation kernel is the sinc function.
– Interpolation using the sinc kernel is NOT required