Digital transformation of the sampling rate of signals, or
signal processing with different sampling rates in the system.
-
Sampling-rate conversion:
CD to DAT format change, for example.
-
Improved D/A, A/D conversion: oversampling
converters; which reduce performance requirements on
anti-aliasing or reconstruction filters
-
FDM channel modulation and processing:
bandwidth of individual channels is much less than the
overall bandwidth
-
Subband coding of speech and images: Eyes and
ears are not as sensitive to errors in higher frequency
bands, so many coding schemes split signals into different
frequency bands and quantize higher-frequency bands with
much less precision.
This procedure is motivated by an analog-based method: one
conceptually simple method to change the sampling rate is to
simply convert a digital signal to an analog signal and
resample it! (Figure 1)
H
aa
Ω=1if|Ω|<π
T
1
0otherwise
H
aa
Ω
1
Ω
T
1
0
h
aa
t=sinπ
T
1
tπ
T
1
t
h
aa
t
T
1
t
T
1
t
Recall the ideal D/A:
x
a
′
t=∑n=-∞∞
x
0
nsinπt-n
T
0
T
0
πt-n
T
0
T
0
x
a
′
t
n
x
0
n
t
n
T
0
T
0
t
n
T
0
T
0
(1)
The problems with this scheme are:
- A/D, D/A, filters cost money
- imperfections in these devices introduce errors
Digital implementation of rate-changing according to this
formula has three problems:
-
Infinite sum: The solution is to truncate. Consider
sinct≈0
sinc
t
0
for
t<
t
1
t
t
1
,
t>
t
2
t
t
2
: Then
m
T
1
-n
T
0
<
t
1
m
T
1
n
T
0
t
1
and
m
T
1
-n
T
0
>
t
2
m
T
1
n
T
0
t
2
which implies
N
1
=⌈m
T
1
-
t
2
T
0
⌉
N
1
m
T
1
t
2
T
0
N
2
=⌊m
T
1
-
t
1
T
0
⌋
N
2
m
T
1
t
1
T
0
x
1
m=∑n=
N
1
N
2
x
0
n
sinc
T
′
m
T
1
-n
T
0
x
1
m
n
N
1
N
2
x
0
n
sinc
T
′
m
T
1
n
T
0
This is essentially lowpass filter design using a boxcar
window: other finite-length filter design methods could be
used for this.
-
Lack of causality: The solution is to delay by
max{|N|}
N
samples. The mathematics of the analog portions
of this system can be implemented digitally.
x
1
m=
h
aa
t*
x
a
′
t|t=m
T
1
=∫-∞∞∑n=-∞∞
x
0
nsinπm
T
1
-τ-n
T
0
T
0
πm
T
1
-τ-n
T
0
T
0
sinπτ
T
1
πτ
T
1
dτ
x
1
m
t
m
T
1
h
aa
t
x
a
′
t
τ
n
x
0
n
m
T
1
τ
n
T
0
T
0
m
T
1
τ
n
T
0
T
0
τ
T
1
τ
T
1
(2)
x
1
m=∑n=-∞∞
x
0
nsinπ
T
′
m
T
1
-n
T
0
π
T
′
m
T
1
-n
T
0
|
T
′
=max{
T
0
T
1
}=∑n=-∞∞
x
0
n
sinc
T
′
m
T
1
-n
T
0
x
1
m
T
′
T
0
T
1
n
x
0
n
T
′
m
T
1
n
T
0
T
′
m
T
1
n
T
0
n
x
0
n
sinc
T
′
m
T
1
n
T
0
(3)
So we have an all-digital formula for
exact digital-to-digital rate changing!
-
Cost of computing
sinc
T
′
m
T
1
-n
T
0
sinc
T
′
m
T
1
n
T
0
: The solution is to precompute the table of
sinct
sinc
t
values. However, if
T
1
T
0
T
1
T
0
is not a rational fraction, an infinite number of
samples will be needed, so some approximation will have to
be tolerated.
Rate transformation of any rate to any other rate can be
accomplished digitally with arbitrary precision (if some
delay is acceptable). This method is used in practice in
many cases. We will examine a number of special cases and
computational improvements, but in some sense everything
that follows are details; the above idea is the central
idea in multirate signal processing.
Useful references for the traditional material (everything
except PRFBs) are Crochiere and Rabiner,
1981 and Crochiere and Rabiner,
1983. A more recent tutorial is Vaidyanathan; see also Rioul
and Vetterli. References to most of the original papers
can be found in these tutorials.
-
R.E. Crochiere and L.R. Rabiner. (1981, March). Interpolation and Decimation of Digital Signals: A Tutorial Review. Proc. IEEE, 69(3), 300-331.
-
R.E. Crochiere and L.R. Rabiner. (1983). Multirate Digital Signal Processing. Englewood Cliffs, NJ: Prentice-Hall.
-
P.P Vaidyanathan. (1990, January). Multirate Digital Filters, Filter Banks, Polyphase Networks, and Applications: A Tutorial. Proc. IEEE, 78(1), 56-93.
-
O. Rioul and M. Vetterli. (1991, October). Wavelets and Signal Processing. IEEE Signal Processing Magazine, 8(4), 14-38.