Interpolation means increasing the sampling rate,
or filling in in-between samples. Equivalent to sampling a
bandlimited analog signal LL
times faster. For the ideal interpolator,
X
1
ω=
X
0
Lωif|ω|<πL0ifπL≤|ω|≤π
X
1
ω
X
0
L
ω
ω
L
0
L
ω
(1)
We wish to accomplish this digitally. Consider
Equation 2 and
Figure 1.
ym=
X
0
mLifm=0±L±2L…0otherwise
y
m
X
0
m
L
m
0
±
L
±
2
L
…
0
(2)
The DTFT of
ym
y
m
is
Yω=∑m=-ω∞ymⅇ-ⅈωm=∑n=-∞∞
x
0
nⅇ-ⅈωLn=∑n=-∞∞xnⅇ-ⅈωLn=
X
0
ωL
Y
ω
m
ω
y
m
ω
m
n
x
0
n
ω
L
n
n
x
n
ω
L
n
X
0
ω
L
(3)
Since
X
0
ω
′
X
0
ω
′
is periodic with a period of
2π
2
,
X
0
Lω=Yω
X
0
L
ω
Y
ω
is periodic with a period of
2πL
2
L
(see
Figure 2).
By inserting zero samples between the samples of
x
0
n
x
0
n
, we obtain a signal with a scaled frequency response
that simply replicates
X
0
ω
′
X
0
ω
′
LL times over a
2π
2
interval!
Obviously, the desired
x
1
m
x
1
m
can be obtained simply by lowpass filtering
ym
y
m
to remove the replicas.
x
1
m=ym*
h
L
m
x
1
m
y
m
h
L
m
(4)
Given
H
L
m=1if|ω|<πL0ifπL≤|ω|≤π
H
L
m
1
ω
L
0
L
ω
In practice, a finite-length lowpass filter is designed using
any of the methods studied so far (
Figure 3).
Let
ym=
x
0
Lm
y
m
x
0
L
m
(Figure 4)
That is, keep only every
LLth
sample (
Figure 5)
In frequency (DTFT):
Yω=∑m=-∞∞ymⅇ-ⅈωm=∑m=-∞∞
x
0
Mmⅇ-ⅈωm=∑n=-∞∞
x
0
n∑k=-∞∞δn−Mkⅇ-ⅈωnM|n=Mm=∑n=-∞∞
x
0
n∑k=-∞∞δn−Mkⅇ-ⅈ
ω
′
n|
ω
′
=ωM=DTFT
x
0
n*DTFT∑δn−Mk
Y
ω
m
y
m
ω
m
m
x
0
M
m
ω
m
n
M
m
n
x
0
n
k
δ
n
M
k
ω
n
M
ω
′
ω
M
n
x
0
n
k
δ
n
M
k
ω
′
n
DTFT
x
0
n
DTFT
δ
n
M
k
(5)
Now
DTFT∑δn−Mk=2π∑k=0M−1Xkδ
ω
′
−2πkM
DTFT
δ
n
M
k
2
k
0
M
1
X
k
δ
ω
′
2
k
M
for
|ω|<π
ω
as shown in homework #1 , where
Xk
X
k
is the DFT of one period of the periodic sequence.
In this case,
Xk=1
X
k
1
for
k∈01…M−1
k
0
1
…
M
1
and
DTFT∑δn−Mk=2π∑k=0M−1δ
ω
′
−2πkM
DTFT
δ
n
M
k
2
k
0
M
1
δ
ω
′
2
k
M
.
DTFT
x
0
n*DTFT∑δn−Mk=
X
0
ω
′
*2π∑k=0M−1δ
ω
′
−2πkM=12π∫-ππ
X
0
μ
′
2π∑k=0M−1δ
ω
′
−
μ
′
−2πkMd
μ
′
=∑k=0M−1
X
0
ω
′
−2πkM
DTFT
x
0
n
DTFT
δ
n
M
k
X
0
ω
′
2
k
0
M
1
δ
ω
′
2
k
M
1
2
μ
′
X
0
μ
′
2
k
0
M
1
δ
ω
′
μ
′
2
k
M
k
0
M
1
X
0
ω
′
2
k
M
(6)
so
Yω=∑k=0M−1
X
0
ωM−2πkM
Y
ω
k
0
M
1
X
0
ω
M
2
k
M
i.e., we get
digital
aliasing.(
Figure 6)
Usually, we prefer not to have aliasing, so the downsampler
is preceded by a lowpass filter to remove all frequency
components above
|ω|<πM
ω
M
(
Figure 7).
This is easily accomplished by interpolating by a factor of
LL, then decimating by a factor
of MM (Figure 8).
The two lowpass filters can be combined into one LP filter
with the lower cutoff,
Hω=1if|ω|<πmax{LM}0ifπmax{LM}≤|ω|≤π
H
ω
1
ω
L
M
0
L
M
ω
Obviously, the computational complexity and simplicity of
implementation will depend on
LM
L
M
:
2/3
23
will be easier to implement than
1061/1060
10611060!