Sub-band Processing
There exist many applications in modern signal processing
where it is advantageous to separate a signal into different
frequency ranges called
sub-bands. The
spectrum might be partitioned in the uniform manner
illustrated in
Figure 1, where the
sub-band width
Δ
k
=2πM
Δ
k
2
M
is identical for each sub-band and the band centers are
uniformly spaced at intervals of
2πM
2
M
.
Alternatively, the sub-bands might have a logarithmic
spacing like that shown in
Figure 2.
For most of our discussion, we will focus on uniformly
spaced sub-bands.
The separation into sub-band components is intended to make
further processing more convenient. Some of the most
popular applications for sub-band decomposition are audio
and video source coding (with the goal of efficient storage
and/or transmission).
Figure 3 illustrates the use of
sub-band processing in MPEG audio coding. There a
psychoacoustic model is used to decide how much quantization
error can be tolerated in each sub-band while remaining
below the hearing threshold of a human listener. In the
sub-bands that can tolerate more error, less bits are used
for coding. The quantized sub-band signals can then be
decoded and recombined to reconstruct (an approximate
version of) the input signal. Such processing allows, on
average, a 12-to-1 reduction in bit rate while still
maintaining "CD quality" audio. The psychoacoustic model
takes into account the
spectral masking
phenomenon of the human ear, which says that high energy in
one spectral region will limit the ear's ability to hear
details in nearby spectral regions. Therefore, when the
energy in one sub-band is high, nearby sub-bands can be
coded with less bits without degrading the perceived quality
of the audio signal. The MPEG standard specifies
32-channels of sub-band filtering. Some psychoacoustic
models also take into account "temporal masking" properties
of the human ear, which say that a loud burst of sound will
temporarily overload the ear for short time durations,
making it possible to hide quantization noise in the time
interval after a loud sound burst.
In typical applications, non-trivial signal processing takes
place between the bank of analysis filters and the bank of
synthesis filters, as shown in
Figure 4. We will focus, however, on filterbank
design rather than on the processing that occurs between the
filterbanks.
Our goals in filter design are:
-
Good sub-band frequency separation
(i.e., good "frequency selectivity").
-
Good reconstruction (i.e.,
yn≈xn-d
y
n
x
n
d
for some integer delay
dd) when the sub-band
processing is lossless.
The first goal is driven by the assumption that the sub-band
processing works best when it is given
access to cleanly separated sub-band signals, while the
second goal is motivated by the idea that the sub-band
filtering should not limit the reconstruction performance
when the sub-band processing (e.g., the
coding/decoding) is lossless or nearly lossless.