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Multirate digital filters and filter banks find application in com- digital filter bankdesigns.Ashort summaryof someofthese
munications, speech processing, image compression, antenna sys- developments was reported recently by this author at an
tems, analog voice privacy systems, and in the digital audio indus-
try. During the last several years there has been substantial progress IEEE international conference [5]. The purpose of this article
in multirate system research. This includes design of decimation i s to provide a self-contained and more complete exposure
and interpolation filters, analysis/synthesis filter banks (also called to many recent contributions on multirate systems, includ-
quadrature mirror filters, or QMFJ, and the development of new ing filter bank design.
sampling theorems. First, the basic concepts and building blocks
in multirate digital signal processing (DSPJ, including the digital
As mentioned in [3], multirate systems find application
polyphase representation, are reviewed. Next, recent progress as in communications, speech processing, spectrum analysis
reported by several authors in this area is discussed. Several appli- [6],radar systems, and antenna systems. In this tutorial, two
cations are described, including the following: subband coding of sections are devoted to a review of applications. In Section
waveforms, voice privacy systems, integral and fractional sampling 111, we point out applications in digital audio systems, in
rate conversion (such as in digital audio), digital crossover net-
works, and multirate coding of narrow-band filter coefficients. The subband coding techniques (used in speech and image
M-band QMF bank is discussed in considerable detail, including compression), and in analog voice privacy systems (for stan-
an analysis of various errors and imperfections. Recent techniques dard telephone communications). In Section V-E, appli-
for perfect signal reconstruction in such systems are reviewed. The cations of special transfer functions (such as complemen-
connection between QMF banks and other related topics, such as
tary functions) in digital audio is reviewed. In Section IX,
block digital filtering and periodically time-varying systems, based
on a pseudo-circulant matrix framework, is covered. Unconven- several unconventional applications of multirate systems
tional applications of the polyphase concept are discussed. and polyphase theory are indicated. These include a) deri-
vation of new sampling theorems for efficient compression
I. INTRODUCTION of signals, b) derivation of new techniques for efficient cod-
In recent years there has been tremendous progress in ing of impulse response sequences of narrow band filters,
the multirate processing of digital signals. Unlike the sin- c) design of FIR filters with adjustable multilevel responses,
gle-rate system, the sample spacing in a multirate system and d) adaptive filtering in subbands.
can vary from point to point [I], [2]. This often results in more
efficient processing of signals because the sampling rates A. Paper Outline
at various internal points can be kept as small as possible. In section II, basic tools, such as decimators, interpola-
Unfortunately, thisalso results in the introduction of a new tors, decimation and interpolation filters, and digital filter
type of error, i.e., aliasing, which should somehow be can- banks, are reviewed, along with the interconnection prop-
celed eventually. erties of the building blocks. In section I l l , some applica-
The basic building blocks in a multirate digital signal pro- tions of multirate DSP are indicated, in digital audio sys-
cessing (DSP) system are decimators and interpolators. In tems, in subband coding, and in voice privacy systems.
1981, an excellent tutorial article on decimation and inter- Section IV reviews the digital polyphase decomposition due
polation appeared in [3]. Subsequent to this a text on the to Bellanger, along with applications such as the uniform
subject of multirate systems has also been published by the DFT filter bank. The concept of multilevel polyphase
same authors [4]. Since then, a number of new develop- decomposition is also introduced here as a tool for efficient
ments have taken place in the area, particularly in multirate implementation of fractional decimation filters. Several
special types of filter banks, such as Nyquist filters, power-
Manuscript received October12,1988; revised Junel3,1989.This complementary systems and Euclidean filter-banks, are
work was supported in part by the National Science Foundation studied in section V. In section VI, the two-band QMF bank
under grants DCI 8552579 and MIP 8604456.
The author is with the Department of Electrical Engineering, Cal- i s studied in sufficient detail along with procedures for
ifornia Institute of Technology, Pasadena, CA 91125, USA. eliminating aliasing i n such systems. Procedures for elim-
I E E E Log Number 8933329. ination of amplitude and/or phase distortion are discussed.
xk = XnWkn, 0 5 k I M - 1. (1a)
ll=O
which says that the output at time n is equal to the input
The inverse DFT (IDFT) i s given by at time Mn. As a consequence, only the input samples with
sample numbers equal to a multiple of M are retained. This
sampling-rate reduction by a factor of M i s demonstrated
(1b)
in Fig. 2 for the case of M = 2. The L-fold interpolator i s char-
I ,I ,
i s given by
1 M-'
yD(z) = - X(Z"MWk) (3b)
~ ... _,, M k=O
01234 which means MY,(e/") ',= ,::E X(e/'"-2"k"M ). The term with
Fig. 2. Demonstration of decimation for M = 2. k = 0 i s indeed the M-fold stretched version of X(e/"). The
M - 1 terms with k > 0 are uniformly shifted versions of
acterized by the input-output relation this stretched version. These M terms together make up a
function with period 27r in w, which i s the basic property
of the Fourier transform of any sequence [8]. Fig. 4(c) dem-
onstrates this effect for M = 2. The terms with k > 0 are
called the aliasing terms. As long as x(n) i s bandlimited to
IwI < d M , there is no overlap of these terms with the k =
That is, the output yI(n) is obtained by inserting L - 1zero-
0 term.
valued samples between adjacent samples of x(n), as dem-
The fundamental difference between aliasing and imag-
onstrated in Fig. 3for L = 2. The decimator and interpolator
ing is important to notice. Aliasing can cause loss of infor-
mation because of the possible overlap of the shifted ver-
sionsof the stretched version of X(e1"). Imaging, on the other
hand, does not lead to any loss of information (which iscon-
sistent with the fact that no time-domain samples are lost).
B. Interconnections
Fig. 5 shows acascade connection which is often encoun-
tered in filter-bank systems. The signal v(n) here i s equal to
Fig. 3. Demonstration of interpolation for L = 2.
X(n)*v(n)
are linear systems even though they are time-varying [4], [SI,
[IO].
The z transform of the interpolator output yl(n) i s given
by [41: -n/M n/M
Fig. 5. Effect of decimation followed by interpolation.
Y/(Z) = X(ZL). (3-3)
This means YI(e/") = X(eluL)i.e., YI(e/") is an L-fold com-
x(n) whenever n i s a multiple of M, and zero otherwise. The
pressed version of X(e"?, as demonstrated in Fig. 4(b). The
transform-domain relation is
appearance of multiple copies of the basic spectrum in Fig.
1 M-'
4 is called the imaging effect and the extra copies are the
images created by the interpolator.
V(Z) = -
M k=O
c
X(zWk) (4)
(Appendix A) that the systems of Fig. 6(a) and 6(b) are iden-
tical if and only if L and M are relatively prime.
Decimation Filters and Interpolation Filters: In most
applications a decimator i s preceded by a bandlimiting fil-
ter H(z)whose purpose i s to avoid aliasing. For example, a (C) (d)
low-pass filter with stopband edge U, = TIM can serve as Fig. 9. Noble identitiesfor multiratesystems. (a) Decimator
such a filter. The cascade shown in Fig. 7(a) is commonly followed by transfer function G(z). (b) Equivalent cascade.
(c) Example of transfer function preceding. (d) Equivalent
called a decimation filter. An interpolation filter, on the
cascade.
other hand, i s a device which follows an interpolator (Fig.
7(b)), the purpose being to eliminate the images. The low-
pass filter of Fig. 7(c) again serves as an example (with L = on (3b), that this cascade is equivalent to the one in Fig. 9(b)
M). provided G(z) is a rational transfer function (i.e., a ratio of
polynomials in z-'). In a similar manner, the two cascades
in Figs. 9(c) and 9(d) are equivalent (provided G(z)i s rational),
-y(n)
as can be proved from (3a). These identities are very val-
The decimation filter
uable in practically all applications for efficient implemen-
(a) tation of filters and filter banks. We shall call these the
-y(n)
"noble identities."
The interpOlatiOnfilter
C. Analysis and Synthesis Banks
(b)
These are the two basic types of filter banks. An analysis
- -+Hli (eJw; bank is a set of analysis filters H k ( Z )which splits a signal into
w M subband signals xk(n) as shown in Fig. 10(a).What we do
M m
(C)
(C)
w
-&
3
0 x with the subband signals depends on the application, aswe
shall see in sections Ill, VI, and IX. Next, a synthesis bank
(Fig. 10(b)) consists of M synthesis filters Fk(Z), which com-
bine M signals yk(n) (possibly from an analysis bank) into
a reconstructed signal i(n). There are several types of filter
banks, i.e., the complementary type, the Nyquist type, etc.,
to be described in Section V along with applications.
(C) Uniform DFT Filter Banks: An analysis bank with M filters
Fig. 8. Decimation by rational fraction of MIL. (a) General (M > 1) i s said to be a uniform DFTfilter bank if all the filters
structure. (b)Exampleof bandlimited signal. (c) Effect of frac- are derived from H&) according to Hk(z)= H,(zWk),0 5
tional decimation of this signal ( L = 2, M = 3). k 5 M - 1. Here H&) i s called the prototype filter. Note
~
that Hk(e/") = Ho(el(w-2*k'M) ), which means that the fre-
minimum over-sampling
quency responses of HJz) are uniformly shifted versions sampling rate
of Ho(elw).Fig. I O W shows a typical set of responses, where
Ho(z)i s taken to be low pass. More details can be found in
* : =
-44 -22 22
4
44
H
kz:/:
88
section IV-C and in [4] and [ I l l .
which can then be interpolated and recombined through Once again Ek(z)are called polyphase components and the
the synthesis filters. coefficients ek(n) and are given by
The polyphase decomposition, which originated from the In other words, the impulse response sequence { h ( n ) }has
work by Bellanger et a/. [ I l l , i s very fundamental to many now been grouped into M subsequences ek(n).The kth sub-
applications in multirate DSP. These include efficient real- sequence ek(n) i s mererly the M-fold decimated version of
time implementation of decimation and interpolation fil- h(n + k). Note that for a given H(z), the quantity Ek(z)
ters, fractional sampling-rate changing devices, uniform depends on M. For example Eo(z) in (7) i s totally different
DFT filter banks, and perfect-reconstruction analysiskyn- from that in (IO). A second subscript (such as Ek,M(~)),which
thesis systems. More recently, the polyphase idea has been would make this distinction clear, i s avoided in this paper
applied for the derivation of new sampling theorems [30], in the interest of simplicity. In most applications M is fixed,
and for recovering bandlimited signalsfrom non-uniformly and there i s no room for confusion.
sampled versions. In this section the basic concept is intro- An important property of the representation (IO) is given
duced and some of the aforementioned applications are by the following relation:
elaborated. Section IX deals with the remaining applica- M-I
tions. To avoid repetitions, results presented in [3], [4] are c
k=O
H(ZWk) = ME,(zM) (12)
reviewed only briefly here.
which follows from the property (IC)of the quantity W. The
A. The Polyphase Decomposition z transform Eo(z)of eo(n) is thus seen to be equal to
H ( z " ~ W ~ ) I and
M , since eo(n)i s nothing but the M-fold deci-
Let H(z) = E:=-, h(n)z-" be a transfer function repre-
mated version of h(n), this gives a proof of the formula (3b)!
senting a digital filter. We can rewrite H(z) in the form
Before getting into applications, we shall mention a sec-
+ h(-4)z4 + h ( - 2 ) 1 + h(0)
H(z) = [ . . . ond type of polyphase decomposition which i s notationally
more convenient in some situations. This is given by
+ h(2)zT2+ h ( 4 ) ~ +- ~ . .I *
M-I
+ z-'[ - + h(-3)z4 + h(-l)z2
* * H(z) = c
k =O
z - ( ~ - ' - ~ )k(ZM).
R (13)
+ h(1) + h(3)z-Z + . . .]. (5)
It i s evident that RJz) i s related to Ek(z)by Rk(z) = EM-' -&),
If we use the abbreviations which is just a renumbering of the components. The rep-
m m resentations (IO) and (13) will be, respectively, called Type
Eo(z) = c
n=-m
h(2n)z-", El(z) =
n=-m
h(2n + l)z-" (6) 1and Type 2 polyphase decompositions. Correspondingly,
Ek(z)and Rk(z),0 5 k 5 M - 1 are called Type 1 and Type
we can re-express (5) in the form 2 polyphase components. It i s important to notice that the
polyphase decomposition can be applied for any sequence
+
H(z) = Eo(z2) z-'E1(z2). (7) x(n) and i s not restricted to impulse response coefficients.
Basically, we have merely grouped the impulse-response
coefficients h(n) into even numbered samples eo(n)= h(2n) B. Application in Sampling-Rate Alteration
and odd-numbered samples e,(n) = h(2n 1). Denoting the + In Section Il-B we mentioned three standard kinds of
z-transforms of eo(n)and el(n) by Eo(z)and E,(z), respectively, sampling-rate alterations, namely decimation, interpola-
we obtain the relation (7). This representation will be called tion, and fractional decimation. Each of these involves adig-
the two-component polyphase decomposition of H(z). The ita1filter H(z), which unfortunatelyoperates at apointwhere
quantities Eo(z)and E,(z) are called the polyphase compo- the sampling rate i s highest (see Figs. 7(a), 7(b) and 8(a)).
nents of H(z). As an example, let H(z) = 1 2z-' 5z-' + + + These filtering operations can be made more efficient as
~ . H(z) = 1 5z-'
4 ~ - Then + z-'[2 + +
4z-*], so that Eo(z)= described next.
+
1 5z-' and E,(z) = 2 4z-l. + Decimation Filters: Consider again the decimation filter
Note that the representation of (7) holds whether H(z) i s of Fig. 7(a)with M = 2. Supposewe implement H(z)(assumed
FIR or IIR. For example, let FlRof 1engthN)directlyasin Fig.14.Thesystem hastocom-
-l puteonlytheeven numbered output samples y(2n). Foreach
computed sample we require N multiplications and N - 1
additions.Assumethatx(n) isarriving in real-timeatthe rate
By using the identity 1 - x = (1 - x')/(l + x), we can rewrite
(8) as
so that Eo(z) = 1/(1 - a'z-') and El(z) = a41 - a'z-'). Fig. 14. Direct implementation of FIR decimation filter.
X ( ) Z ? J L [ xo(n)
2 -1
EM-&) xM-l(n)
~.
band filters, and power-complementary filter-banks. We = z - ~ H ( z )then
, the relation (18)translatestothewell-known
shall now review these systems and indicate some appli- relation OI[ ] HJZ) - H,(-z) = z - ~ .
cations. An Mth band filter [35] is a logical extension of a half-band
filter. We shall define it as a zero-phase FIR filter with
A. Half-band Filters, M t h Band Filters, and Nyquist Filters impulse response satisfying
In most applications, these are linear-phase FIR filters.
The impulse response h(n)i s assumed to be Hermitian-sym- (20)
metric with respect t o n = 0 (i.e., H(z) = E,"=- K h(n)z-" with
h(n) = h*(-n)), so that these are, i n fact, zero-phase filters. The constant c i s usually taken to be 1/M. The property of
A half-bandfilter [35], [16, page 1401by definition, i s a zero- (20) i s also called the Nyquist property, for historical rea:
phase filter whose impulse response satisfies
sons [37]. Representing H(z) in the polyphase form of (IO),
it is clear that the component Eo(z)i s equal to the constant
h(2n) =
Ic
0
n = O
n # O
(16)
sin ( d M )
h(n) = w(n)
7rn
2n
n 2n
(C)
2n
0 n 2n
(g)
Alias-term
a t Output
of F,,(z)
" 0 n 2n
(h)
Alias-term
a t Output
of F1(z)
0 n 2n
(i)
Fig. 23. Illustration of various Fourier transforms in two-channel QMF bank. Here hor-
izontal axis represents W. (a) Typical input. (b) Transform. (c) Aliasing effect. (d) Imaging
effect. (e) Using xl. (9 Using vl. (g) Using yl. (h) Alias-term at output of F,,(z). (i) Alias-term
at output of Fl(z).
Summarizing, the reconstructed signal R(n)in Fig. 13 suf- further. On the other hand, we shall see that the first three
fers from three distortions: aliasing distortion (ALD), ampli- errors can be completely eliminated at finite cost by careful
tudedistortion (AMD),and phasedistortion (PHD). In addi- choice of the filters Hk(z)and Fk(z).Such systems which are
tion, we have the nonlinear coding error created by the alias-free and satisfy (31) are called perfect-reconstruction
encoding (or quantization) of the decimated signals. Since (PR) systems.
coding error cannot be eliminated, we shall not discuss it Elimination of Phase Distortion: For any pair of analysis
T(z) = [ H ~ ( z-
) H;(-z)] = ~ Z - ' ~ ( Z ~ ) , ( Z ~ )(36) Summarizing, a two-channel FIR QMF bank with filters
restricted as in (34) can have PR property if and only if H,(z)
which automatically has linear phase as long as Ho(z)is cho- i s a trivial transfer function of the form (37).
sen to have linear phase (the RHS of (36) is obtained bywrit- In practice, frequency responses are required to be more
0.
-10.B08
-20. e00
-38.888
-40.000
-50.000
NORMALIZED FREQUENCY
(a)
2. 000
iL 1.600
0
..
Y
c
p:
0
c 1.20~
v)
c
U
e. I I
e. 0.1~0 0.200 0.300 0.400 e.50~
NORHALIZEO FREQUENCY
(b)
Fig. 24. Three design examples for FIR QMF bank, demonstrating amplitude distortion.
(a) Response of analysis filters H,,(z). (b) Corresponding amplitude distortions.
0. 8.100 8.200
NORNRLIZEO
e.190
FREOUENCV
nn 8.400 8.w
(a)
2o.eoe .
*
a
-I
w
e,
a.
=I
18.080 3
. with equalizer
without
E 12.810 equalizer
w
6.889
e. I
0. e.ia8 0.2~9 13.308 e.ue 0.501
NORNALIZED FREOUENCV
(b)
Fig. 26.Design example for IIR all-pass-based QMF bank. (a) Magnitude responseof H,(z).
(b) Group delay of complete QMF bank with and without equalizer.
R(4) = 0.666467,
(d
el = 0.7791~, e2 = 0.1896r, e3 = 0.3808~, Fig. 27. Pertainingto design of perfect-reconstruction sys-
= 0.5759~, tems. (a) Simple perfect-reconstruction system. (b) Redraw-
ing two-band QMF bank. (c) Here P ( z ) = R(z)E(z).
and /3 = 0.659.
There exist a number of digital filter structures for imple-
menting all-pass filters [52], I531 which have the following Ho(z)i s as in (37). Let us, therefore, not assume at the begin-
property: the transfer function remains (stable and) all-pass, ning any relationship between filters.
despite quantization of the multipliers. In other words, the Suppose each of the analysis filters i s written in the form
transfer function i s structurallyall-pass (i.e., all-pass despite +
of (7), i.e., H&) = Eko(z2) z-'Ekl(z2),k = 0,l. Similarly let
multiplier values). This i s very desirable: if we implement each synthesis filter be written in the form (13), i.e., Fk(z)=
the QMF bank as in Fig. 25, with the functions Eo(z)and El(z) +
z-'Rok(z2) Rlk(z2). We can collect these equations neatly
constrained to be structurally all-pass, then ALD and AMD in matrix-form as follows:
are eliminated regardless of quantization of the structure
coefficients.
See the end of section VI for a discussion of implemen-
tation complexity and for a comparison with other types of
two-channel QMF banks of comparable performance. and
-_____ ~ _ _
As explained earlier, it i s not possible to obtain a PR system for some c # 0. The property of (48) implies that H(z) is uni-
based on this, because Ho(z) would then become con- tary on the unit circle (see section I), i.e., Ht(e'") H(e"? =
strained as in (37). c l for all W.
To obtain an FIR PR system with good filters Ho(z)and A constant unitary matrix i s a trivial example of a lossless
H,(z),we havetogive up theconstraint (34). Smith and Barn- system. A scalar stable all-pass function H(z), which by def-
well [54] have shown how this can be done, based on spec- inition satisfies 1 H(e'")l = constant, i s another example. AS
tral factorization of half-band filters. We shall review this a third example, if Hk(z), 0 5 k 5 M - 1 i s a power com-
same approach from a different viewpoint which enables plementary set, then the M x 1 vector [Ho(z) Hl(z) * e e
us to extend the perfect reconstruction property for arbi- HM-l(z)]T i s a lossless system. A lossless example with p =
trary M and, in addition, gives efficient polyphase lattice r i s offered by the 2 x 2 orthogonal matrix [A-;]. When p
structures for robust implementation. Consider the task of = r, an FIR lossless systems H(z) satisfies the property det
satisfying the general condition (45) for perfect reconstruc- H(z) = ~ ! z - ~where
, d i s a constant and K i s an integer [62],
tion.Thus let us forcethe productP(z)to be d z - K / b y requir- [64]. Moreover, the inverse i s given by H-'(z) = fi(z)/c, as
ing R(z) = d ~ - ~ E - ' ( z This
) . means that R(z) and hence Fk(z) seen from (48). In other words, the inverse i s not only guar-
are, in general, IIR filters. The synthesis filters will be stable anteed to exist, but also can be found from H(z) simply by
provided the determinant of (z) i s a minimum phase poly- writing down fi(z). For example, let
nomial,
To obtain a PR system with only FIR filters, we must work
harder. The strategy woud be to constrain det f(z) to be a
delay. A subclass of FIR matrices for which the determinant
H(z) = [: :1:: ; 1: 1.
We can verify that this is lossless (because (48) is satisfied
(49)
Historically, lossless systems are well known in classical Returning now to the QMF bank, suppose we restrict f(z)
electrical network theory [56]-[58]. An electrical network to be FIR and lossless. We can now choose R ( z ) as
composed of lossless elements (such as inductors and R(z) = z-Ki(z) = cz-Kf-'(z) (51)
capacitors but no resistors) i s said to be lossless.The imped-
ance matrix Z(s) of a multiterminal lossless electrical net- so that perfect reconstruction i s guaranteed. The delayz-K
work satisfies Re [Z(jt?)] = 0. The scattering matrix of a mul- ensures causalityof the synthesis bank. For example, if (z)
+
titerminal network i s defined as T(s) = [Z(s) - I][Z(s) I ] - ' . i s as in (49), then R ( z ) should be
For a lossless network, the property of Z(jt?) translates into
the unitariness of T(jt?).An excellent (and perhaps the only
one of its kind) reference on lossless systems is the text by
Belevitch [58].
R(z) =
[
1 +z-'
-1 + z - l
-1 + z - '
1 +z-'
With (z) as in (49) and R(z) as in (52), we have
I- (52)
I \ A - f
so that
Method 2
(Smith-Barnwell [54],
VaidyanathanlHoang Method 3
Method 1 b51) (VaidyanathadNguyen [67]) Method 4
(Johnston'sTable) Perfect-Reconstruction Perfect-Reconstruction IIR Based
[41, [MI System System WI, [431
Relation between H,(z) = Ho(-z) H,(z) = -z-Lfi,(-z) not explicit; implicitly same as method 1
filters F ~ ( z )= HoM, Fo(z)= z-'fio(z), det E(z) = delay, and
F~(z= ) -H~(-z) F,(z) = z-'fi,(z) R(z) = cz-*E-'(z)
L = order of Ho(z)
Phase response of linear nonlinear linear nonlinear since H o ( z )
Ho ( I ) IIR
Other crucial features f i o ( z ) ~ o ( is
z )a (zero- 2H0(z)= ao(zZ)
of Ho(z) phase FIR) half-band + Z - ~ ~ , ( Zao(z),
~),
filter a,(z) IIR allpass
Distortions in QMF ALD canceled ALD canceled ALD canceled ALD canceled
bank AMD minimized AMD eliminated AMD eliminated AMD eliminated
PHD eliminated PHD eliminated PHD eliminated PHD equalized
Notation for filter N, = length of N, = length of Ho(z) N3 = length of Ho(z) N, = order of H,(z)
length or order as Ho (z)
relevant
Number of MPU to N, (direct-form, Nz (lattice, polyphase) N3 N, - 1 + equalizer
implement entire polyphase) - (lattice, polyphase)
2
analysis/synthesis overhead (all-pass
system polyphase)
Group delay of entire N, - 1 NZ - 1 N, - 1 complicated
analysis/synthesis
system
Note: ALD = aliasing distortion; A M D = amplitude distortion; PHD = phase distortion.
Table 2 Comparison of Design Examples for Three FIR Two-Channel QMF Banks. In
All Three Methods, H,(z) has A, = 38 dB and w , = 0.586 IT
- __ __-
linear periodically time-varying (LPTV) systems. In the fol- The M X M matrix H(z) in (64) i s called the alias-component
lowing parts of this section we shall review this relation. matrix. If it can be inverted, it i s indeed possible to find the
synthesis filters Fk(z),0 5 k 5 M - 1to cancel aliasing com-
A. Alias-Free QMF Banks pletely. The difficultywith this approach i s that even if Hk(z)
are FIR, the inverse of H(z)could be IIR, so that Fk(z)are, in
Fig. 29(a) shows the M band generalization of the QMF
general, neither FIR nor stable. If we multiply all Fk(z)bythe
bank. It i s said to be a maximally decimated structure
LCM of the denominators of Fk(z),0 5 k 5 M - 1 to get a
because the decimation ratio M i s equal to the number of
new set of synthesis filters, aliasing continues to be absent
bands.Todistinguish this from tree structures(section VIII),
(as seen from (64)), but T(z)changes. In any case, the result-
the scheme of Fig. 29(a) is called a parallel structure. Once
ing set of synthesis filters can have very high orders. For
these reasons, inversion of H(z)i s usually not used as a pro-
cedure for the design of alias-free QMF banks.
If we approach this problem in terms of the polyphase
matrices (z)and R(z)as in section VI-B, we have much bet-
ter luck. In fact, a convenient procedure for design of
M-band FIR perfect-reconstruction QMF banks based on
lossless (z)has been developed [31], [63], [64], and will be
summarized in Section VIII.
Suppose we express Hk(z)and Fk(z)in terms of Type 1 and
Type 2 polyphase components, respectively:
M-1
(b) Hk(Z) = c
n=o
z -nEkn(ZM),
Fig. 29. (a) M-band maximally decimated parallel QMF
M-1
bank. (b) Redrawing of Fig. 29(a) in terms of polyphase com-
ponent matrices.
1: Ho(ZWM-l) H,(z
..
~
(b)
Fig. 31. (a) Scalar transfer function S(z).(b) Its blocked ver-
sion with block length M.
(b)
where xdn) = x(nM + k), 0 I k s M - 1.Then x(n) is called Fig. 32. Representationof arbitrary linear periodicallytime
varying system using polyphase framework. (a) The LPTV
the blocked version of the scalar signal x(n). Similarly, the system. (b) Redrawing based on components defined in
blocked version y(n) of y(n) is equation (69).
]
y(n) = [yM-l(n) yM-2(n) .. * yo(n)lT (67)
i
Gdz) GOl(Z) G02(z)
where yk(n) = y(nM + k), 0 I kI M - 1. The integer M i s
called the block length. It can be shown [731-[75] that the P(z) = z-lclz(z) G,o(z) Gll(4 . (70)
two vector-sequences y(n) and x(n) are related by a transfer z -1G2,(Z) z -lG*2(Z) G20(z)
matrix, i.e.,
In other words, the kth row of P(z) is a k-times circularly
V(z)= f(z) X(z). (68) shifted version of the kth row of G(z) with a delay 2 - l
attached t o the elements that spill over. Here i s the third
The M x M matrix p(z) i s called the blocked version of S(z).
question we raise.
Implementing the filterS(z1 by direct implementation of P(z)
i s called block digital filtering. Theadvantageof such imple- Question 3: Let Fig. 32(b) represent an arbitrary LPTV sys-
mentation i s that we increase the available parallelism by tem. What i s a set of necessary and sufficient conditions on
a factor of M. Block filtering therefore offers higher imple- p(z) so that the system is actually time-invariant?
mentation speed due to increased parallelism.
Wecan seethat the blocking operation (68) i s veryclosely D. Pseudo-circulant Matrices
related to polyphase decomposition. Indeed, the compo-
Thethreequestions raised in the preceding havethesame
nents of (66) are merely M-fold decimated versions of x(n
+ k),0 5 k 5 M - 1. It can be seen that the implementation answer! The answer is that, f ( z ) should have an algebraic
form called the pseudo-circulant form. Thus, if (and only
of S(z) using the blocking approach can be given a flow-
graph representation as in Fig. 31(b). The close relation if)P(z) i s pseudo-circulant,thefollowing aretrue: a)theQMF
between the blocking scheme and QMF banks can be seen bank is alias-free; b) f(z) i s the blocked version of a scalar
filter; and c) the LPTV system of Fig. 32(b) represents an LTI
from this representation. Here is the second question we
system. This result is proved in [70]. In this section we shall
raise in this section (we shall discuss the answer in the fol-
merely state the meaning of "pseudo-circulant." Taking M
lowing text).
= 3, for convenience, a 3 x 3 pseudo-circulant has the form
Question 2: Suppose we are given an arbitrary M x M
transfer matrix f ( z ) .What i s a set of necessary and sufficient
conditions on p(z) such that it i s the blocked version of some
scalar filter S(z)!
f(z) = [
k!;'2(z)
Pdz)
Po(z)
P, (z) z - PJZ)
:El.
Po(z)
(71)
C. Linear Periodically Time-Varying (1PTV) Systems Thus P(z) i s a circulant matrix [76] with the exception that
theentries below the main diagonal are multiplied with z -'.
Consider a linear system whose coefficients vary period- Pseudo-circulants have also been observed explicitly or
ically with time, with period M. For 0 I k IM - 1, letAk(z) implicitly in the context of block processing by other
represent the transfer function of the system if its coeffi- authors [75], [77]-[80].
cients were frozen to be their values at time -k. For each Assuming that f(z) is pseudo-circulant, what is the scalar
k, let u s represent Ak(Z) in terms of i t s M polyphase com- filter S(z) of which f(z) i s the blocked version? The answer
ponents: Ak(z) = C!:Jz-"Gk,(zM). Defining G(z) = [Gkn(Z)], is
we can represent the LPTV system as in Fig. 32(a). By defin-
M-1
ing the components
s(z) = z-kPk(ZM). (72)
k=O
(69) I n other words, the 0th row of f ( z )contains the type-1 poly-
phase components of s ( ~ ) Next,
. given a QMF bank with
pseudo-circulant f(z) (so that aliasing i s canceled), the dis-
we can redraw Fig. 32(a) as in Fig. 32(b). The relation of (69)
tortion function T(z) of (64) i s given by
between the'kth rows of P(z) and G(z)i s quite interesting:
thus, with M = 3, we see that f(z) i s given by T(z) = ~ - (S(Z).
~ - l ) (73)
-
It can also be shown that P(z) i s lossless if and only if S(z) computations of sines and cosines per iteration, which
is stable all-pass. Thus an alias-free QMF bank i s free from results in a very slow optimization procedure.
AMD if and only if the pseudo-circulant i s lossless, and the It i s possible to obtain a more convenient characteriza-
blocked version of a scalar filter i s lossless if and only if the tion of FIR lossless matrices that i s free from cosines and
scalar filter is stable all-pass! sines. The following result is proved in [62] and [63] and
works for the real as well as complex coefficient case.
VIII. M-BAND PERFECT
RECONSTRUCTION SYSTEMS
Theorem. An M x M causal FIR transfer matrix F(z) of
Perfect reconstruction in M-band FIR QMF banks has degree K is lossless if and only if it can be expressed in the
been shown to be possible in the past [14], [31], [50]. We shall form
now summarize some techniques for this. E(z) = vK(z) vK-I(z) vl(z)/fO (75)
Consider the M-band QMF bank of Fig. 29(a) again. As
where Vn(z)are M x M matrices of the form
seen in Section VILA, there are M - 1 alias components in
k(n), and these components are removed if and only if (64) VJZ) = [I - v, v:, + z - l v , v:] (76)
holds. If perfect reconstruction i s also desired, we have to in which v, are M x 1 column vectors of unit norm, and Ho
force T(z) to be a delay. It remains to solve for the synthesis i s an M x M constant unitary matrix.
filters Fk(z) in (64). Once again we wish to avoid inversion The term degree in the preceding statement indicates
of the alias-component matrix. So we lean on the polyphase the smallest number of scalar delays (i.e., z - elements)
representation of Fig. 29(b), which is completely equivalent required to implement the system. It can be shown [62], [63]
to Fig. 29(a). that the form (76) for V,(z) with unit-norm v, automatically
As we did in section VI-B, we can choose R(z) = E-(z) to guarantees that Vn(z)is (causal, FIR, and) lossless of degree
obtain perfect reconstruction. To avoid inversion of F(z), we one. If we write down the state-space representation of the
wish to force it to be lossless so that (z) = cF-(z). This system implemented as in (75), the system matrix R turns
means that the choice of R(z), as in (51), will ensure the PR out to be unitary [82]. In other words, the structure for (z)
property. Assuming that H ~ ( z(and
) hence (z)) are FIR, this obtained in this way i s an orthogonal structure in the
also ensures that R(z), and hence Fk(z)are FIR. Moreover, same sense as those in [64] are. This property is important
this choice of R(z) i s equivalent to choosing the synthesis because it implies certain useful properties for finite word-
filter coefficients as [31] length implementations, such as automatic Oe2 scaling of
fk(n) = &;(no - 1 - n), 0 k 5 M -1 internal nodes, and reduced round-off noise [83].
5 (74)
It is important to note that for the case of real coefficient
where noi s the length of the longest analysis filter, and a FIR lossless matrices, the unit-norm vectors v, in (76) and
i s a nonzero constant which can be scaled to be unity. the unitary matrix Ho in (75) are real.
It remains to show how E(z)can be forced to be lossless. It i s possible to optimize the unit-norm vectors v, in (76)
In the M = 2 case, we had the factorization result (60) for and the unitary matrix Ho in (75) such that the sum of stop-
lossless matrices. Similar factorization theorems have also band energies of the analysis filters Hk(z)i s minimized.This
been developed [62]-[64] for the M x M case. A structure issue is handled in much greater detail in [63]. In fact, it is
has been developed in [64] for real coefficient FIR lossless possible to completely avoid rotation angles in the expres-
transfer matrices based on a state-space approach. The sion (75) (and this helps to speed up the optimization of the
structure is such that every M x M real coefficient FIR loss- analysis filter responses) because the unitary matrix Ho can
less system can be realized using this structure. Conversely, also be expressed as a product of M - 1matrices of the form
regardless of the values of the parameters in the structure [ I - 2w, w;] (where w, are unit-norm column vectors), fol-
(which are angles of rotation), the transfer matrix remains lowed by a diagonal unitary matrix. The factor [I -.
(FIR and) lossless. The development in [64] is based on the 2w, 4 1 , which i s a unitary matrix, i s thewell-known House-
discrete-time lossless bounded real lemma [81] which holder form.
relates the losslessness of (z)to the state-space descrip- The degree-one lossless building blocks described by(76)
tion (see end of section I for state-space notations). Accord- can be implemented with 2M multipliers and one scalar
ing to [81], a stable transfer matrix(z) i s lossless if and only delay (see Fig. 33). For the real coefficient case, Ho can be
if there exists a structure (with minimum number of delays)
such that the system matrix
R = r
C D
1 Fig. 33. Implementationof degree-oneM x M FIR lossless
i s unitary. Acomplete parameterization of all lossless matri- building block using 2M multipliers and one scalar delay.
ces of a given degree can therefore be obtained merely by
parameterizing all unitary matrices of a given size. The implemented using (r)planar rotation angles (see [64] and
development in [64] makes further use of the fact that F(z) [84]), each of which is equivalent to four real multiplica-
i s FIR, so that A has all eigenvalues equal to zero. Because tions. The matrix H, then requires 4 (r)
multiplications so
of the unitariness of the R matrix, the structures in [64] are that the total number of real multipliers in a real-coefficient
also said to be orthogonal filter structures. FlRanalysis bankwith losslessF(z),as in(75), isequal to2MK
The only disadvantage of the structure in [64] i s that the + 2M(M - 1).
parameters in the structure are sines and cosines of angles. Robustness of Perfect Reconstruction Under Coefficient
If we wish to optimize the analysis filter responses by opti- Quantization: Recall that as long as the polyphase matrix
mizing these angles, then the procedure involves several of theanalysis bank has the form (75),wherevnareunit-norm
~- ~ _ _
designed by constraining (z) to be lossless as just the M band FlRanalysisbankwith lossless polyphase matrix
described, and optimizing the parameters of v, and H,. The F(z)implemented as in (75).Assuming thecoefficients of the
optimization was carried out using appropriate software in filters to be real (so that v, and Hoare real), we have a total
[86]. Initialization of theoptimization wasdoneasdescribed +
of 2MK 2M(M - 1)multipliers for the analysis bank. With
in [63]. To obtain perfect reconstruction, the FIR synthesis F(z) as in (751, the analysis filters have maximum length N
filters were obtained as in (74) with no = 56. As a second +
= KM M, so that the number of multipliers becomes 2(N
example, Fig. 34(b) shows the responses of the analysis fil- +
- 1) 2(M - Recognizingthat these multiplications can
ters Ho(z),l+,(~),and H2(z)for the M = 5 case.The remaining be performed at the lower rate (by moving the decimators
two filters are given by H&z) = H1(-z) and H4(z)= Ho(-z), to the left of F(z)in Fig. 29(b)), the number of MPUs for the
and are therefore not shown (this relation is not necessary +
analysis bank is equal to 2(N - 1)lM 2(M - 1)21M= 2(N
for perfect reconstruction, nor i s it automatic; see [64]). The +
- 1)lM 2M. In comparison, consider the uniform DFT
FIR matrix F(z) was again constrained to be lossless. Once analysis bank of Fig. 21, which we found to be very efficient
again, if synthesis filters are chosen according to (74) with in section IV. Assuming that the analysis filters have length
no = 44, we have perfect reconstruction, because of the N each, the total number of multiplications i s equal to N
losslessness of F(z)which was forced during the design of plus the cost of IDFT. The number of MPUs i s equal to
Hk(z)s.The table of filter coefficients for these two exam- N/M plus the DFT cost, which depends on M. For N >> M
ples can be found in [63] and [64], respectively. Finally, Fig. we see that the perfect reconstruction FIR system with loss-
34(c) shows the analysis filter responses for a three-channel less&) implementedas in (75)isonlyabouttwotimes more
QMF bank, where the analysis filters have linear phase; the expensive than the uniform DFT filter bank with same filter
matrix F(z) i s not lossless here, but still has a determinant orders! In fact, if we impose certain symmetry conditions
equal toadelay.Thetableof filter coefficientsforthis exam- among the analysis filters Hk(z)in the perfect reconstruc-
ple can be found in [87l. tion system (as elaborated in [go]), then the number of MPUs
Methods Based on Tree-Structures: When M i s a power for the perfect reconstruction system i s reduced by a factor
of two, we can obtain the QMF bank by repeated use of the of two, so that it becomes equal to that of the uniform DFT
two-channel QMF bank. See [54] for details. If the two-chan- polyphase analysis bank. Summarizing, the computational
ne1 building block i s designed to have the PR property, then complexity of the FIR perfect-reconstruction analysis bank
so does the M channel tree-structured design. Since the based on lossless polyphase matrices can be made com-
design of the filters Ho(z)and Hl(z) i s easier than a direct parable to that of the uniform DFT polyphase analysis bank
M-band design, tree structures are often considered attrac- for same N, whenever N >> M.
tive. However, for agiven M, tree structures are not optimal Unsuitability o f the Uniform DFT QMF Bank for Perfect
in terms of filter lengths and result in greater group delay Alias Cancellation: The uniform DFT FIR analysis bank
of the overall QMF system, compared to nontree designs. shown in Fig.21 iscomputationallyveryefficient,and iswell
Such group delays are of concern in long-distance com- suited for spectrum analysis. However, it is usually not well
munication links, where echos are possible. suited for QMF bank applications requiring perfect recon-
Methods Based on Approximations: Some authors have struction (hence perfect alias cancellation in particular).This
developed methods for designing the filters Hk(z)such that can be seen by referring to [12], which derives the necessary
the three distortions (ALD, AMD, and PHD) are approxi- and sufficientconditionsontheFlRsynthesisfiltersforalias
mately eliminated. Notable among these are [88] and [89]. cancellation in this case. It turns out that the synthesis fil-
These are based on the design of an appropriate optimized ters typically have much higher orders than the analysis fil-
prototype filter, from which the other filters can be derived ters, resulting in a much more expensive synthesis bank.
using a modified form of cosine modulation. The approx- In contrast to this, the perfect reconstruction FIRQMF bank
imate designs are sometimes less expensive to implement based on general lossless FIR (z) (rather than on uniform
than the perfect ones. DFT analysis bank) i s such that each synthesis filter has the
Price Paid for Perfect Reconstruction: It i s sometimes same length as the corresponding analysis filter (see (74)).
assumed that one must pay a high price in terms of design The overall cost of the complete analysislsynthesis system
complexity, to achieve perfect reconstruction in FIR QMF is therefore lower.
banks. This, however, i s not true. The PR property often Alias-component matrix, polyphase component matrix,
admits the representation of the analysis bank in terms of and interrelations: It can be shown [31] that the M x M
lattice structures, which are surprisingly efficient. If a PR matrix in (64) is lossless if and only if the polyphase com-
system i s implemented in lattice form, it is often not more ponent matrix F(z)is lossless. So by forcing (z) to be loss-
expensive than an approximate design with comparable less, we make the inversion of (64) simple; in fact, closed-
attenuations for the analysis filters. See section VI-E and form expressions for fk(n) as in (74) are obtained.
Tablesl-3of this paper;also seePO, pp. 701-703lforfurther For any M-band maximally decimated FIRQMF bank, con-
quantitative remarks. sider the following three properties: a) losslessness of (z);
As a further demonstration of this claim, consider the FIR b) the relation (74); and c) perfect reconstruction property
perfect reconstruction example in Fig. 34(c). The analysis (32). It can be shown [31] that any two of these imply the
filters here have linear phase. A design example i s pre- third. For example, if a QMF bank satisfies (74) and has per-
sented in [87l for the M = 3 case, where the filters Ho(z), fect reconstruction property, then &) must be lossless.
Hl(z),and H2(z)have lengths 56,53, and 56, respectively. The Similarly, if F(z) is lossless, then we have perfect recon-
entire analysis bank of three filters, if implemented using struction if and only if (74) holds. Note, however, that nei-
the lattice structure in [87, requires a total of only 8 MPUs ther losslessness of E(z)nor the relation (74) are, by them-
and 11 APUs! selves, necessary or sufficient conditions for perfect
Let us now count the total number of multiplications in reconstruction.
IX. UNCONVENTIONAL
APPLICATIONS
In this section we outline some unconventional appli-
cations of multirate filter banks in signal processing. These
include new sampling theorems, new techniques for effi-
cient quantization of filter coefficients, and adaptive fil- Fig. 36. Demonstration of nonuniform sampling.
tering in subbands.
A. New Sampling Theorems Based on the QMF Principle a sufficiently long period exceeds 8. More precise state-
and Polyphase Decompositions ment of this result can be found in [96] and references
therein.This result i s called the nonuniform sampling theo-
The sampling theorem for continuous-time signals has rem, and less formally, the folk theorem.The knowledge
been well known for several decades [94], [95]. The earliest of this result seems to date back [97] t o the days of Cauchy
known version says that if a continuous-time signal x,(t) i s (1841!). An extreme case of the folk theorem is the assertion
bandlimited to -u < Q < u, it can be reconstructed from that a bandlimited signal i s completely determined every-
equallyspaced samplesx(n) = x,(nT) provided thesampling where if all i t s past samples, taken at twice the Nyquist rate,
frequency 2dTexceeeds the so-called Nyquist rate 8 20. are available! The reconstruction of x,(t) from the nonun-
Under this condition, the transform X(e1) is a periodic rep- iformly spaced samples is, in general, very involved [98],
etition of X,(jQ) properly scaled, as shown in Fig. 35. Quan- Wl.
However, if we take a second look at these sampling theo-
rems from a discrete-time viewpoint, the insight obtainable
i s rewarding. For example, let x(n) be a bandlimited
*n sequence, i.e., a sequence whose transform X(e1) vanishes
-0 a
for w, 5 IwI 5 K, with wC < K. An example with wC = 2 d 3
(a) was seen in Fig. 8(b). It i s clear that we can construct a new
lowpass filler lor
recoveringx&
sequence y(n) with a lower rate, without losing information.
from samples, tx(ell Fig. 8(a) shows how to do this in terms of the fractional deci-
... ... 0
mator circuit (L = 2, M = 3). We know that if H(z) i s ideal
low-pass with cutoff 2d3, Y(e) i s a stretched version of
(b) X(e/) by the factor 3/2.
This, however, i s not theonlywayto reduce thesampling
Fig. 35. Transform of x,(t) and transform of sampled ver-
sion x,(nT). (a) Signal recovery. (b) Low-pass filtering. rate. A simpler way to compress x(n) in Fig. 8(b) would be
to divide the time index into segments of length three and
retain two out of three samples in each segment. This is
titatively, equivalent to nonuniform sampling of a hypothetical
underlying continuous-time signal. From this nonuni-
formly decimated version y(n), i s it possible to recoverx(n)?
The answer i s yes, as we would expect because of our
The signal x,(t) can be recovered from the samples x(n) knowledge about the folk theorems.
merely by low-pass filtering the impulse train Er= - m x(n) The main point we wish to make [30] i s that this problem
6(t - nT), which results in elimination of the unwanted cop- can be stated in the form of a QMF bank design problem.
ies of the basic spectrum X,(jQ) in Fig. 35(b). Notice that the With such a restatement, the algorithm for reconstruction
low-pass filter required i s an ideal filter which i s IIR, non- ofx(n)fromy(n)i s seen to beasimplesynthesis-bankdesign
causal, and unstable 17, 181. problem, which actually can be carried out in practice with
Several generalizations of this theorem are known [96]. FIR filters. To demonstrate these ideas, consider the exam-
Perhaps the earliest one, called the derivative sampling ple of Fig. 8(b) again. We wish a data rate reduction of 312.
theorem, says that if x,(t) and dx,(t)/dt are available (for Thiswill beaccomplished by retainingtwoout of everycon-
example, the position and velocity of an aircraft), then uni- secutive segment of three samples. Fig. 37 shows a way to
formly spaced samples of these two waveforms, taken at mechanize this subsampling based on multirate building
half the Nyquist rate, are sufficient to retain all the infor- blocks.The samples retained in the process are those num-
mation. In other words, the signal x,(t) can be recovered bered . . . -4, -3, -1,O, 2,3,5,6,8,9, . . . Consider now
from these samples. Once again, the use of ideal filters i s the M-channel maximally decimated QMF bank of Fig. 29(a)
d
1 1 1
r= o -I -2 . (84)
and that [o 0
[E,(z) E,MI r = [E,(z) E,(z) - E,(Z)I (81) Note again that T - l = There. We can then simplify Fig. 40
to Fig. 41. In this example the difference E0(z)- E&) again
we can redraw Fig. 39(a) as in Fig. 39(b). The coefficients of
Eo(z) - El(z) are composed of the first difference of h(n),
which are expected to be very small.
H(z) = c
k=O
U,(Z) Gk(ZM) (85)
n)y(-n)x(
G(z) = c PfJ-RzWk)
k =O
(86)
I ..
-20.000
m
0
z
l-4 -10.000
W
v)
z
0
n -60 008
I
v)
0
W
-80.000
-100.800
e. 0.200 0.400 0.600 8.800 1.000
NORWALIZEO FREOUENCY
(a)
1.000
8.800
W
z
yr
0
P
In 0.600
w
p:
W
a
0
I.- 8.100
U
z
(I
a
E
0.200
e.
e. 0.200 0.400 0.600 0.800 1.000
NORWALIZED fREQUENCY
(b)
Fig. 47. Design example for adjustable multilevel FIR filters. (a) Response of prototype
fifth band filter of length 59. (b) Two examples of multilevel design, obtained using poly-
phase network of Fig. 46.
X. MULTIRATE OF MULTIDIMENSIONAL
PROCESSING SIGNALS form Vn, where Vis a fixed nonsingular matrix of real num-
bers and n i s a vector of integers. The matrix V is said to
Multidimensional signals [I091arefunctionsof more than
generate the lattice. A discrete-parameter M D signal x(n)
one variable. These are denoted by notations such as x(n,,
can be obtained from a continuous-parameter M D signal
n2, * , nM) and abbreviated as x(n). (We also abbreviate
x,(f) by defining x(n) = x,(Vn). In other words, x,(f) has been
two-dimensional as 2D and multidimensional as MD.)
sampled at the locations of the lattice generated by V. For
It is necessary to manipulate and process such signals in
this reason, Vis called the sampling matrix. The number of
applications such as video, and in radar and sonar signal
samples per unit area which results from this sampling is
processing. In this section we shall be content with point-
equal to the reciprocal of the determinant of V. Notice that
ing out the existing literature on multirate multidimen-
a diagonal V corresponds to rectangular sampling.
sional signal processing.
The reciprocal lattice i s a lattice generated by the matrix
The concept of sampling a two- (or higher) dimensional
V-and plays a key role in sampling theory. Basically, if
signal proves to be more trickythan in the one-dimensional
X,(Q) and X(o) are the multidimensional Fourier transforms
case. This is because the sampling geometry can be rect-
of x,(f) and x(n), respectively, then they are related by
angular or nonrectangular. One of the earliest references
on the generalized sampling of multidimensional signals
i s the work by Petersen and Middleton [IIO], where the L
1
X(W) = - c Xa[V-T(o - 27rk)l
k
number theoretic concepts of lattices and reciprocal lat-
tices were used to describe the sampling and aliasing pro- where L i s equal to the determinant of V. This summation
cesses. A lattice is basically a collection of vectors of the extends over all integer vectors k. If a bandlimited M D sig-
.~
analysis filters as done in section VIII. Even though this tion. There are two cases considered at length in [67. These
problem has not been formulated or solved to this degree belong to subsets #4 and #3, respectively.
of generality, special cases of this can be found in [67] and Another open area in multirate DSP is the design of QMF
[87l (to obtain linear-phase FIR PR QMF banks) and in [55] banks with nonuniform decimation ratio. This means that
(for obtaining special nonconstant passband magnitude the decimators in Fig. 29(a)are not all equal to M but depend
responses for analysis filters in FIR PR QMF banks). One on the subband. For example, let nk represent the deci-
approach towards a generalized solution i s the following mation ratio for the k t h subband with 0 Ik 5 M - 1 (the
observation: an M x M polynomial matrix E(z) in the vari- passband widths of the filters are, of course, accordingly
able z-l has determinant equal to a delay if and only if it adjusted to be nonuniform). These integers should clearly
can be expressed in the form satisfy Clink = 1, to preserve the average number of sam-
ples per unittime. Examplesof such systems have beengen-
F(z) = U(z)A(z)V(z) (92)
erated inthe past bycombiningsmallerQMF bankmodules
where A(z) i s a diagonal matrix of delays (with nondecreas- in some way (such as in a tree structure [4]). Some of the
ing powers as we go down the diagonal) and U(z)and V(z) unresolved issues in the more general case are how to
are polynomial matrices in z-l with determinants equal to obtain alias cancellation (and, for the more ambitious
nonzero constants (in other words, U(z)and V(z)are "uni- designer, perfect reconstruction) with FIR filters. What are
modular" matrices [120])). This observation follows from the necessary and sufficient conditions for these, and how
the Smith-McMillan form of decomposition for polynomial can these be forced structurally?Does losslessness still have
matrices [120, Ch. 61, [121, Ch. 61. The preceding problem a role? Some progress in this direction has been reported
then reducesessentiallytooneof characterizing causal uni- in [123].
modular matrices: one seeks to develop a structure whose Finally, there has been some progress in the design of
multiplier parameters span all unimodular matrices; and two-dimensional filter banks, as mentioned in section X.
converselyevery transfer matrix generated by the structure However, systematic procedures for obtaining completely
i s required to be causal and unimodular. Certain general general lattice structures for lossless polyphase matrices
factorization theorems such as those in [I221 may give a are yet to be developed, due to lack of neat factorization
starting point for this problem. theorems in the general case. Factorization theorems for
Fig. 48 helps to visualize the FIR perfect reconstruction restricted special cases can be found in [82]. Finally, the 2D
family. The largest set in this Venn diagram i s the set of all versions of practically all the I D open problems are open
as well.
ALL FIR E (1)
NOMENCLATURE
AC All-pass complementary.
A/D Analog to digital.
ADPCM Adaptive delta pulse code modulation.
ALD Aliasing distortion.
AMD Amplitude distortion.
A PCM Adaptive pulse code modulation.
A PU Additions per unit time.
Fig. 48.Venn diagram, summarizing families of FIR QMF D/A Digital to analog.
banks for perfect reconstruction.
DC Doubly complementary.
DFT Discrete Fourier transform.
FIR analysis banks. We then have the subset for which the DS P Digital signal processing.
determinant of E(z)i s a minimum-phase polynomial so that EC Euclidean complementary.
a stable synthesis bank (not necessarily FIR) for perfect FDM Frequency division multiplexing.
reconstruction exists. The next subset is such that det E(z) FIR Finite-length impulse response.
is actually a delay so that an FIR synthesis bank for perfect GPP Generalized polyphase.
reconstruction exists. The synthesis filters can, however, be IDFT Inverse discrete Fourier transform.
longer than the analysis filters. The fourth subset i s such /FIR Interpolated finite-duration impulse response.
that F(z) can be expressed as in (91) (which obviously guar- IIR Infinite-duration impulse response.
antees that the determinant i s a delay). The synthesis filters LBR Lossless bounded real.
for perfect reconstruction, i n principle, can again be longer LOT Lapped orthogonal transform.
than the analysis filters. The last and the smallest subset is L PTV Linear periodically time-varying.
such that E(z) is lossless. In this case the synthesis filters L TI Linear time-invariant.
have the same length as the analysis filters, and moreover LTV Linear t i me-varying.
are given by the closed form formula (74). The advantage MD Multidimensional.
of subset #5 i s that it gives rise to lattice structures, which MOS Mean opinion score.
in turn lead to simple design procedures for the analysis MPU MuIt i pl icat ions per u n it-ti me.
filters, ensuring that the filter-responses are nicely PC Power complementary.
bounded. Thedisadvantage is that, in thetwo-channel case, PHD Phase distortion.
losslessness is not compatible with the linear-phase PR Perfect reconstruction.
requirement of analysis filters (unless they are trivial, as in QMF Quadrature mirror filter.
(37)). In 1671, losslessness has been relaxed to obtain linear- TDM Time-domain multiplexing.
phase two-channel QMF banks with perfect reconstruc- 20 Two-dimensional.
APPENDIX
A I would like to thank Dr. R. E. Crochiere of the AT&T Bell
CONDITIONFOR EQUIVALENCE OF FIGS. 6(a) AND 6(b) Labs for his encouragement in the preparation of this arti-
cle. It is a privilege to acknowledge the useful discussions
By applying the relations (3a), (3b) to Figs. 6(a) and 6(b) we I have had with Dr. M. Bellanger of TRT, France, Prof. M.
see that Vetterli of Columbia University, Profs. Smith and Barnwell
Y1(Z) =
1
kso
M-l
X(zLlMWkL) (A.4
of Georgia Institute of Technology, and Prof. T. Ramstad of
the Norwegian InstituteofTechnology.Also, theinput from
reviewers resulted in useful improvements of this paper.
and Finally, the valuable multirate interaction I have had with
my graduate students during the last few years is gratefully
1
Y 2 ( z )= -
M
c
M-l
k = ~
X(zL/MWk). (A.2) acknowledged. In particular, Zinnur Doganatas interac-
tions on lossless systems, Vincent Lius interactions on
The summations (A.l) and (A.2) are identical except for the nonuniform sampling and on 2D systems, and Truong
powers of W appearing in each. The right sides of (A.l) and Nguyens interactions o n symmetric and linear-phase QMF
(A.2) are identical if and only if the collection of M numbers banks have been particularly beneficial.
WkL,0 I k 5 M - 1 i s equal to the collection W k ,0 5 k
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[I171 R. Ansari and S. H. Lee, Two-dimensional multirate pro- was born in Calcutta, India, on October 16,
cessing on nonrectangular grids: Theory and filtering pro- 1954. He received the 6%. (Hons.) degree
cedures, preprint copy. in physics and the B. Tech and M. Tech
[I181 E. Viscito and J. Allebach, Design of perfect reconstruc- degrees in radiophysics and electronics, all
tion multi-dimensional filter banks using cascaded Smith from the University of Calcutta, India, in
form matrices, in Proc. /E Int. Symp. on Circuits and 1974, 1977, and 1979, respectively. He
Syst., Espoo, Finland, pp. 831-834, June 1988. received the Ph.D. degree in electrical and
[I191 J. H. McClellan and T. W. Parks, A unified approach to the computer engineering from the University
design of optimum FIR linear-phase digital filters, / E of California, Santa Barbara, in 1982.
Trans. Circuit Theory, vol. 20, pp. 697-701, Nov. 1973. He was a postdoctoral fellow at the Uni-
[120] F. R. Cantmacher, The Theory of Matrices, vols. 1 and 2. versity of California, Santa Barbara., from September 1982 to Feb-
New York: Chelsea Publishing, 1959. ruary1983. InMarch 1983 hejoined the California lnstituteof Tech-
[I211 T. Kailath, Linear Systems. Englewood Cliffs, NI: Prentice nology, Pasadena, as an Assistant Professor of electrical
Hall, 1980. engineering, and is currently as Associate Professor in the same
[I221 S. Tan and J.Vandewalle, Fundamental factorization theo- department. His main research interests are in digital signal pro-
rems for rational matrices over complex or real fields, in cessing, multirate filter bank systems, filter design, adaptive fil-
Proc. /E Int. Symp. on Circuits and Syst., Espoo, Finland, tering, and multivariable system theory.
June1988, pp. 1183-1186. Dr. Vaidyanathan served as Vice-chairman of the Technical Pro-
[I231 P.-Q. Hoang and P. P. Vaidyanathan, Non-uniform mul- gram Committeefor the 1983 IEEE International Symposium on Cir-
tirate filter banks: Theory and design, in Proc. /FEE Int. cuits and Systems and as an Associate Editor for the IEEE TRANS-
Symp. on Circuits and Syst., Portland, OR, May 1989, pp. ACTIONS ON CIRCUITS AND SYSTEMSfor the period 1985-1987. He was
371-374. a recipient of theAward for Excellence inTeachingattheCalifornia
[I241 A. C. Constantinides and R. A. Valenzuela, A class of effi- Institute of Technology for the year 1983-1984. He was also a recip-
cient interpolators and decimators with applications in ient of the NSFs Presidential Young Investigator Award in 1986.
transmultiplexers, in Proc. / E Int. Symp. on Circuits and In 1989 he received the IEEE ASSP Senior Award for his paper on
Syst, Rome, May 1982, pp. 260-263. multirate perfect-reconstruction filter banks.