Digital Signal Processing 22 (2012) 847–858

Contents lists available at SciVerse ScienceDirect

Digital Signal Processing

www.elsevier.com/locate/dsp

Design of 2-D interpolation/decimation filters using a general 2-D digital allpass

filter ✩
Ju-Hong Lee a,b,∗ , Yuan-Hau Yang c
a
Department of Electrical Engineering, Graduate Institute of Communication Engineering, National Taiwan University, No. 1, Sec. 4, Roosevelt Rd., Taipei, 10617, Taiwan
b
Graduate Institute of Biomedical Electronics and Bioinformatics, National Taiwan University, No. 1, Sec. 4, Roosevelt Rd., Taipei, 10617, Taiwan
c
Graduate Institute of Communication Engineering, National Taiwan University, No. 1, Sec. 4, Roosevelt Rd., Taipei, 10617, Taiwan

a r t i c l e i n f o a b s t r a c t

Article history: Two-dimensional (2-D) interpolation/decimation digital filters are widely used for sampling rate
Available online 28 March 2012 conversion. A general structure consisting of a 2-D recursive digital allpass filter (DAF) and a 2-D pure
delay block is presented for designing 2-D recursive interpolation/decimation filters. We utilize a 2-D DAF
Keywords:
with symmetric half-plane (SHP) support for its filter coefficients to comply with the symmetry possessed
Doubly complementary
Allpass filter
by 2-D interpolation/decimation filters. The structure also possesses a preferable doubly complementary
Diamond-shaped filter half-band (DC-HB) property that reduces the number of required independent coefficients for designing
Symmetric half-plane filter 2-D interpolation/decimation filters. We appropriately formulate the design problem to obtain a simple
Sampling rate conversion linear optimization problem that minimizes the phase error of the 2-D recursive SHP DAF in the pth
norm (L p ) sense. Simulation results are provided for illustration and comparison.
© 2012 Elsevier Inc. All rights reserved.

1. Introduction terpolation/decimation digital filter with a diamond-shape as well

Many techniques have been presented for the design of 1-D re-
cursive digital filters using the parallel combination of 1-D allpass
sections [1–5]. The designed 1-D recursive digital filters structures
possess very low passband sensitivity and doubly complementary
(DC) properties. The design of 1-D DC digital filters composed of
two complex allpass sections was considered in [2,3] and two
real allpass sections were discussed in [4,5]. Recently, 1-D all-
pass filter structures were successfully applied to the design of the
analysis/synthesis filters for two-channel nonuniform-division fil-
ter (NDF) banks [6]. For the 2-D case, the design of 2-D recursive
circularly symmetric lowpass filters (CS-LPF) based on the parallel
combination of allpass subfilters (PCAS) with coefficients restricted
in the 2-D quarter-plane (QP) was investigated by [7] and [8]. Two
PCAS structures are cascaded to remove the unwanted passband
for designing a CS-LPF.
2-D digital filters have been widely applied to video and image
signal processing [9,10]. For example, 2-D interpolation/decimation
digital filters play an important role for sampling rate conversion,
such as the conversion between the rectangular and hexagonal
sampling structures used in television systems. Moreover, a 2-D in-
✩
This work was supported by the National Science Council under Grants NSC97-
2221-E002-116-MY3 and NSC100-2221-E002-200-MY3.
*Corresponding author at: Department of Electrical Engineering, Graduate In-
stitute of Communication Engineering, National Taiwan University, No. 1, Sec. 4,
Roosevelt Rd., Taipei, 10617, Taiwan. Fax: +886 2 23671909.
E-mail address: juhong@cc.ee.ntu.edu.tw (J.-H. Lee).
as quadrant-symmetry in frequency response can provide a maxi-
mum resolution in the horizontal and vertical directions.
In this paper, we deal with the design problem of 2-D inter-
polation/decimation digital filters. To comply with the symmetric
characteristics possessed by 2-D interpolation/decimation digital
filters, a 2-D recursive digital filter structure consisting of a 2-D re-
cursive digital allpass filter (DAF) with symmetric half-plane (SHP)
support for its filter coefficients and a 2-D pure delay block is
presented. We show that the proposed 2-D recursive digital filter
possesses two preferable frequency characteristics in terms of the
doubly complementary (DC) and half-band (HB) properties. The
combination of the DC and HB properties (termed as the DC-HB
property) provides two significant advantages that we only have to
find almost half of the independent filter coefficients and consider
the passband response during design process. Several important
frequency properties of the proposed 2-D recursive SHP DAF are
investigated. The design problem of a 2-D interpolation/decimation
digital filter is then equivalent to the design problem of a 2-D
recursive SHP DAF. By constraining the phase characteristics to en-
sure the filter stability [11], the design problem in the L p optimal
sense is formulated based on the phase response error of the 2-D
recursive SHP DAF. After some algebraic manipulations, we obtain
a linear optimization problem for finding the filter coefficients. As
a result, a closed-form solution can be easily found for the L 2
design, while the primal affine-scaling variant of Karmarkar’s al-
gorithm [12] (PAS algorithm) can be applied to solve the linear
programming problems related to the L 1 and L ∞ designs, respec-
tively.

1051-2004/$ – see front matter © 2012 Elsevier Inc. All rights reserved.
http://dx.doi.org/10.1016/j.dsp.2012.03.007
848 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858

Table 1
Frequency response of a 2-D recursive digital allpass filter at the CPs.

Crucial points Order of A (e j ω1 , e j ω2 ) A (e j ω1 , e j ω2 )

Ωcp0 For all M and N 1

Ωcp1 M odd −1
M even 1

Ωcp2 M + N odd −1
M + N even 1

Ωcp3 N odd −1
N even 1

Fig. 1. Crucial points (shown by open circles) in the (ω1 , ω2 ) plane.

This paper is organized as follows. Section 2 reviews the fre-

quency characteristics of 2-D recursive DAFs. The stability con-
straints on the phase responses of 2-D recursive filters are also
briefly described. Section 3 presents a general filter structure com-
posed of a 2-D recursive SHP DAF and a 2-D pure delay. We formu-
late the design problem of a 2-D interpolation/decimation digital Fig. 2. The proposed structure for a 2-D recursive digital filter.

filter using the proposed 2-D recursive filter structure in Section 4.

In Section 5, we present efficient design techniques to solve the We note from Property 2 that some unwanted passbands or
resulting optimization problems. In Section 6, several design ex- stopbands may be induced when A (e j ω1 , e j ω2 ) is utilized for the
amples are provided for illustration and comparison. We conclude construction of a 2-D QP recursive digital filter. The orders M and
the paper in Section 7. N of this 2-D QP recursive DAF must be appropriately specified to
avoid the possible unwanted passbands or stopbands [8].
2. 2-D recursive digital allpass filters
2.2. Stability of 2-D recursive allpass filter
2.1. Conventional 2-D recursive digital allpass filters
It has been shown in [9] that a 2-D recursive digital filter is
stable if the unwrapped phase of its denominator is continuous
The transfer function of a conventional 2-D recursive digital all-
and doubly periodic. Consequently, the phase φ(ω1 , ω2 ) must be
pass filter (DAF) can be expressed as
continuous and satisfy the following periodic conditions:
D ( z1−1 , z2−1 )
A ( z1 , z2 ) = z1− M z2− N , (1) φ(ω1 , ω2 ) = φ(ω1 + 2π , ω2 ) (5)
D ( z1 , z2 )
and
where the denominator D ( z1 , z2 ) is given by
φ(ω1 , ω2 ) = φ(ω1 , ω2 + 2π ). (6)

M 
N
d(m, n) z1−m z2−n .
According to the results shown by [11], a necessary and sufficient
D ( z1 , z2 ) = (2)
condition that guarantees the stability of the 2-D recursive DAF
m =0 n =0
A ( z1 , z2 ) of (1) is given as follows:
The support region for the coefficients d(m, n) of D ( z1 , z2 ) is re-
π  
stricted to the first quarter-plane (QP) of the 2-D (m, n) plane. Let d
the phase responses of A ( z1 , z2 ) and D ( z1 , z2 ) be θ(ω1 , ω2 ) and − θ(ω1 , ω2 ) dω1 = M π , for −π  ω2  π (7)
dω1
φ(ω1 , ω2 ), respectively. We have from (1) that 0
 
φ(ω1 , ω2 ) = − M ω1 + N ω2 + θ(ω1 , ω2 ) /2. (3) and
π  
Two important properties regarding the frequency characteristics d
of the 2-D recursive DAF A ( z1 , z2 ) are as follows:
− θ(ω1 , ω2 ) dω2 = N π , for −π  ω1  π . (8)
dω2
0

Property 1. The phase response φ(ω1 , ω2 ) is equal to zero at some fre-
quency points termed as the crucial points (CPs) listed in Fig. 1 [8]. Hence,
the phase response θ(ω1 , ω2 ) of A ( z1 , z2 ) is given by
θ(ω1 , ω2 )|(ω1 ,ω2 )∈Ωcp = − M ω1 − N ω2 ,
where Ωcp denotes the set of the CPs.
Property 2. The frequency responses A (e j ω1 , e j ω2 ) are restricted to 1
or −1 only for (ω1 , ω2 ) ∈ Ωcp . The details of this property are listed in
Table 1.
This condition reveals that θ(ω1 , ω2 ) exhibits monotone decreas-
ing behavior and spans a range of M π ( N π ) radians as ω1 (ω2 )
increases from 0 to π for −π  ω2  π (−π  ω1  π ). There-
fore, by specifying a desired phase response θd (ω1 , ω2 ) to satisfy
(4) the above constraints, we can neglect the stability problem during
the design process.
3. A general 2-D recursive digital filter structure
Fig. 2 depicts the proposed structure for constructing a 2-D re-
cursive digital filter G ( z1 , z2 ). This structure consists of a parallel
 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858 849


interconnection of a 2-D recursive DAF and a 2-D pure delay ele-  
M2 N2

ment. According to the results presented in [1–5], we would expect + 2d(m, n) cos(mω1 ) cos(nω2 ) . (12)
that G ( z1 , z2 ) provides two advantages of low passband sensitivity m =1 n =1
and good frequency selectivity over conventional 2-D direct-form
Next, we have from (9) that the shifted version of G (e j ω1 , e j ω2 )
structures. The frequency response of G ( z1 , z2 ) is as follows:
given by
 e − jM 1 ω1 e − jN 1 ω2 + A (e j ω1 , e j ω2 ) 
G e j ω1 , e j ω2 = , (9) G e j (ω1 −π ) , e j (ω2 −π )
2
1  − jM 1 (ω1 −π ) − jN 1 (ω2 −π )  
where A (e j ω1 , e j ω2 ) is a 2-D recursive SHP DAF with transfer = e e + A e j (ω1 −π ) , e j (ω2 −π )
function like that of (1) but the denominator D ( z1 , z2 ) of order 2
M 2 × N 2 given by 1  
= e − jM 1 (ω1 −π ) e − jN 1 (ω2 −π ) + A −e j ω1 , −e j ω2 (13)
2

M2

N2
has the following property:
D ( z1 , z2 ) = d(0, 0) + d(m, n) z1−m z2−n
m=− M 2 n=− N 2 
G e j (ω1 −π ) , e j (ω2 −π )

N2
 
M2 N2 ⎧ − jM ω − jN ω
= d(0, 0) + d(0, n) z2−n + d(m, n) z1−m z2−n ⎪
⎪
e 1 1e 1 2 − A (e j ω1 ,e j ω2 )
,
⎪
⎪ 2
⎨
n =1 m =1 n =1 for M 1 + N 1 = even and M 2 + N 2 = odd,
= (14)
 
M2 N2 ⎪
⎪ −e − jM 1 ω1 e − jN 1 ω2 + A (e j ω1 ,e j ω2 )
−n ⎪
⎪ ,
+ d(−m, n) zm
1 z2 . (10) ⎩ 2

m =1 n =1
for M 1 + N 1 = odd and M 2 + N 2 = even,

By setting the coefficients d(m, n) = d(−m, n) for 1  m  M 2 with

and 1  n  N 2 , we note from (10) that the corresponding fre- d(m, n) = 0, for m + n = odd. (15)
quency response of D ( z1 , z2 ) can be rewritten as
j ω1 j ω2 j (ω1 −π ) j (ω2 −π )
Accordingly, G (e , e ) and G (e ,e ) satisfy the 2-D
 
N2
doubly complementary (DC) property, i.e.,
D e j ω1 , e j ω2 = d(0, 0) + d(0, n)e − jnω2 (I) The allpass complementary property:
n =1
  jω jω  
 
M2 N2 G e 1 , e 2 + G e j (ω1 −π ) , e j (ω2 −π )  = 1, for all (ω1 , ω2 ).

+ d(m, n) e − jmω1 + e jmω1 e − jnω2
(16)
m =1 n =1


N2 (II) The power complementary property:
= d(0, 0) + d(0, n)e − jnω2   j ω j ω 2   j (ω −π ) j (ω −π ) 2
G e 1 , e 2  + G e 1 ,e 2  = 1,
n =1

 
M2 N2 for all (ω1 , ω2 ). (17)
+ 2d(m, n) cos(mω1 )e − jnω2
Moreover, this 2-D DC property implies that the proposed 2-D re-
m =1 n =1
 cursive digital filter G ( z1 , z2 ) possesses the 2-D DC symmetry with
= D e − j ω1 , e j ω2 . (11) respect to the half-band (HB) frequency (ω1 , ω2 ) = (π /2, π /2) in
the first quarter of the (ω1 , ω2 ) plane, i.e., the so-called 2-D DC-HB
Eq. (11) reveals that the frequency response of 2-D recursive
property. In contrast, a conventional 2-D finite impulse response
SHP DAF possesses the preferable quadrantally symmetric prop-
(FIR) HB filter H ( z1 , z2 ) only has the power complementary prop-
erty. This property makes the proposed 2-D recursive digital filter
erty.
G ( z1 , z2 ) more suitable to comply with the symmetry possessed
The 2-D DC-HB property of the proposed 2-D recursive digital
by 2-D interpolation/decimation filters. In contrast, the authors
filter G ( z1 , z2 ) demonstrates an important frequency characteristic,
of [13] adopted the nonsymmetric half-plane (NSHP) support re-
i.e., |G (e j ω1 , e j ω2 )| = 0 in the stopband if |G (e j ω1 , e j ω2 )| = 1 in the
gions as the basic sections for building the design structure of
passband. Hence, we only have to consider either the passband
the 2-D recursive DAFs. Nevertheless, the resulting 2-D recursive
or stopband response of G (e j ω1 , e j ω2 ) during the design process.
NSHP DAF does not possess the preferable quadrantally symmetric
Moreover, (15) indicates that about half of A ( z1 , z2 )’s coefficients
property, hence it is not suitable for the design of 2-D interpola-
can be set to zero when designing G ( z1 , z2 ).
tion/decimation filters. Moreover, the phase associated with (11) is
given by
4. Design of 2-D interpolation/decimation digital filters
φ(ω1 , ω2 )

  4.1. Sampling structure conversion
= arg D e j ω1 , e j ω2
 N The sampling rate conversion between different periodic sam-
 2
−1
= − tan d(0, n) sin(nω2 ) pling structures is important for multidimensional signal process-
n =1 ing. There are numerous choices for periodicity matrix P and the
 sampling matrix S [9,10]. Here, we consider two widely used sam-
 
M2 N2
pling structures, i.e., rectangular sampling structure and hexagonal
+ 2d(m, n) cos(mω1 ) sin(nω2 )
sampling structure. For a rectangular sampling, the sampling ma-
m =1 n =1
trix S R and periodicity matrix P R are given by
 
N2  
d(0, 0) + d(0, n) cos(nω2 ) T1 0
SR = (18)
n =1 0 T2
850 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858

Table 2
The relationship between |G (e j ω1 , e j ω2 )| for (ω1 , ω2 ) ∈ Ωcp and the filter orders.

(ω1 , ω2 ) Magnitude response

|G (e j ω1 , e j ω2 )| = 1 |G (e j ω1 , e j ω2 )| = 0
Ωcp0 for all M 1 − M 2 Impracticable
and N 1 − N 2

Ωcp1 M 1 − M 2 even M 1 − M 2 odd

Ωcp2 M 1 − M 2 even, N 1 − N 2 even M 1 − M 2 even, N 1 − N 2 odd

M 1 − M 2 odd, N 1 − N 2 odd M 1 − M 2 odd, N 1 − N 2 even

Ωcp3 N 1 − N 2 even N 1 − N 2 odd

Substituting (3) into (24) yields


G e j ω1 , e j ω2
 
M1 − M2 N1 − N2
Fig. 3. The passband Ω p and stopband Ωs of G d (ω1 , ω2 ). = cos − ω1 − ω2 + φ(ω1 , ω2 )
2 2
  
M1 + M2 N1 + N2
and
  × exp j − ω1 − ω2 − φ(ω1 , ω2 ) .
1 2 2
 −1 T T1
0
PR = SR = , (19) (25)
1
0 T2
According to Property 1, φ(ω1 , ω2 ) = 0 for (ω1 , ω2 ) ∈ Ωcp .
where T 1 and T 2 represent the horizontal and vertical sampling
Hence, the frequency response G (e j ω1 , e j ω2 ) becomes
periods for a spatial sampling pattern, respectively, whereas for
spatiotemporal conversions, T 1 and T 2 represent the vertical and  
G e j ω1 , e j ω2 (ω ,ω )∈Ω
the frame periods, respectively. The matrices associated with the 1 2 cp
(L , K ) hexagonal sampling are given by  
M1 − M2 N1 − N2
  = cos − ω1 − ω2
LT 1 LT 1 2 2
SL, K = (20)   
K T2 −K T 2 M1 + M2 N1 + N2
and
× exp j − ω1 − ω2 (26)
2 2
 1 1

 T
P L , K = S− 1
=
2LT 1 2LT 1
, (21) for (ω1 , ω2 ) ∈ Ωcp . We note from (26) that G (e j ω1 , e j ω2 )|(ω1 ,ω2 )∈Ωcp
L,K 1 −1 is only a function of M 1 , M 2 , N 1 , and N 2 . Therefore, M 1 , M 2 , N 1 ,
2K T 2 2K T 2
and N 2 should be appropriately chosen to approximate a speci-
where L and K are positive integer parameters of the hexagonal
fied filter response. The details regarding the relationship between
sampling structure.
|G (e j ω1 , e j ω2 )| for (ω1 , ω2 ) ∈ Ωcp and the orders M 1 × N 1 , M 2 × N 2
To achieve the conversion processing between rectangular and
are listed in Table 2. Moreover, by letting
hexagonal sampling structures, 2-D decimation/interpolation filters
with a diamond-shaped frequency response are good candidates M1 − M2 N1 − N2
because they allow a maximum resolution in the horizontal and θm (ω1 , ω2 ) = − ω1 − ω2 + φ(ω1 , ω2 ) (27)
2 2
vertical directions. If T 1 and T 2 are set to 1, the ideal frequency
response of a 2-D decimation/interpolation filter in the first QP and
(0  ω1  π and 0  ω2  π ) is specified by M1 + M2 N1 + N2
 θ p (ω1 , ω2 ) = − ω1 − ω2 − φ(ω1 , ω2 ), (28)
− j ( gd1 ω1 + gd2 ω2 ) |ω1 | |ω2 | 2 2
Ge , for ω1p + ω2p  1,
G d (ω1 , ω2 ) = (22)
we obtain
0, otherwise,



where ω1p = π / L and ω2p = π / K . The filter gain G is set to 1 for G e j ω1 , e j ω2 = cos θm (ω1 , ω2 ) exp j θ p (ω1 , ω2 ) (29)
a decimation filter and set to 2L K for an interpolation filter. Fig. 3
and
depicts the magnitude characteristic of G d (ω1 , ω2 ).
M2 N2
4.2. Design problem formulation using the proposed 2-D digital filter φ(ω1 , ω2 ) = − ω1 − ω2
2 2
θ p (ω1 , ω2 ) − θm (ω1 , ω2 )
Let the frequency response of the 2-D recursive SHP DAF of (9) − . (30)
be expressed by 2
 Consequently, both the magnitude and phase responses of
A e j ω1 , e j ω2 = e j θ (ω1 ,ω2 ) . (23)
G (e j ω1 , e j ω2 ) are simultaneously determined by φ(ω1 , ω2 ). There-
Then, (9) becomes fore, the design problem of G (e j ω1 , e j ω2 ) is equivalent to finding
 e − jM 1 ω1 e − jN 1 ω2 + e j θ (ω1 ,ω2 ) φ(ω1 , ω2 ) to approximate the desired response φd (ω1 , ω2 ). From
G e j ω1 , e j ω2 = (30), we can find φd (ω1 , ω2 ) as follows:
2

  
= cos −( M 1 ω1 + N 1 ω2 ) − θ(ω1 , ω2 ) /2 M2 N2

   φd (ω1 , ω2 ) = − ω1 − ω2
× exp j −( M 1 ω1 + N 1 ω2 ) + θ(ω1 , ω2 ) /2 . 2 2
θ p ,d (ω1 , ω2 ) − θm,d (ω1 , ω2 )
(24) − , (31)
2
 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858 851

where θm,d (ω1 , ω2 ) and θ p ,d (ω1 , ω2 ) should be decided based on and a vector d containing the unknown filter coefficients as fol-
(29) according to the given design specification of G (e j ω1 , e j ω2 ). lows:
Therefore, the design problem can be formulated as follows: 
   p d = d( M 2 , N 2 ), d( M 2 − 1, N 2 ), . . . , d(0, N 2 ),
Minimize  W (ω1 , ω2 ) φ(ω1 , ω2 ) − φd (ω1 , ω2 )  , (32)
d( M 2 , N 2 − 1), d( M 2 − 1, N 2 − 1), . . . , d(0, N 2 − 1), . . . ,
where x denotes the pth norm of x and W (ω1 , ω2 ) the fre-
p T
d( M 2 , 1), d( M 2 − 1, 1), . . . , d(0, 1) , (37)
quency weighting function.
where the superscript T denotes the matrix-transpose operation.
5. Design techniques To incorporate the constraint related to the 2-D DC-HB property
into the design process, we set the filter coefficients d(m, n) = 0
5.1. Frequency sampling and approximation scheme for m + n equal to odd in (37). Using the results of (35)–(37), we
can express the minimization problem of (34) in matrix form as
From the design problem shown by (32), we have to find the follows:
real coefficients d(m, n) of the 2-D recursive SHP DAF A ( z1 , z2 )  
such that the resulting phase response φ(ω1 , ω2 ) given by (12) Minimize W(Ud − s) p , (38)
approximates the desired phase response φd (ω1 , ω2 ) in the L p op-
timal sense. Substituting (12) into (32) yields where W represents a diagonal weighting matrix whose diagonal
values are the preset weights on the phase error at the correspond-
 ⎡   N 2 ⎤ p
 − tan−1 d(0, n) sin(nω2 )  ing frequency points. For the problem of minimizing the squared
 n =1 
 ⎢   ⎥ phase error, i.e., p = 2, the solution of (38) is given by
 ⎢ + M 2 N 2 ⎥
 ⎢ m =1
2d (m , n ) cos (m ω1 ⎥
)
 ⎢ n=1 ⎥  − 1
 ⎢ × sin(nω2 ) ⎥
 ⎢ ⎥ d = UT U U T s. (39)
 ⎢   ⎥
Minimize  W (ω1 , ω2 ) ⎢ / d(0, 0) + N 2
d ( 0, n ) cos ( n ω ) ⎥ .
 ⎢ n =1 2 ⎥
 ⎢ ⎥
 ⎢ +  M 2  N 2 2d(m, n) cos(mω ) ⎥ 5.2. Linear programming formulations for the L 1 and L ∞ design
 ⎢ m = 1 n = 1 1 ⎥ problems
 ⎢ ⎥
 ⎣ × cos(nω2 ) ⎦
 
 
−φd (ω1 , ω2 ) Here, we present the formulations for solving two other widely
(33) considered designs in the L p optimal sense, i.e., p = 1 and p = ∞.
L 1 Design Problem: From (38), the L 1 solution can be obtained
Without loss of generality, we set d(0, 0) = 1. After performing by directly solving the following minimization problem:
some algebraic manipulations as shown in Appendix A, we can
rewrite the minimization problem (33) as follows: Minimize 1T e
 ⎡  ⎤
 N2 p W|Ud − s|  e,
 d(0, n) sin nω2 + φd (ω1 , ω2 )  Subject to (40)
 ⎢ n= 1 ⎥
 ⎢ M2 N2 ⎥
 ⎢ + m=1 n=1 2d(m, n) cos(mω1 ) ⎥ where e is the R×1 error bound vector, 1 is an R×1 column vector
Minimize  W (ω1 , ω2 ) ⎢ ⎥ .
 ⎢ × sinnω + φ (ω , ω ) ⎥ whose elements are one, and W is the preset weighting matrix.
 ⎣ 2 d 1 2 ⎦
   (40) can be rewritten as follows:
 + sin φ (ω1 , ω2 ) 
d
(34) Maximize −1T e
 T    
Through a frequency sampling approximation approach, we let the (WU)T −(WU)T d Ws
frequency pair (ω1r , ω2r ) represent the rth uniformly sampled fre- Subject to  . (41)
quency grid point on Ω = {(ω1 , ω2 ) | 0  ω1  π , 0  ω2  π }. The
−1T −1T e −Ws
design process is then performed on the frequency grid points. If From (41), we obtain the standard dual form of a linear program-
the number of grid points is sufficiently large, the obtained best ming (LP) problem as follows:
approximation solution of the objective function based on the fre-
quency sampling approach will be close to the best solution found Maximize bT w
based on Ω . This conclusion can be justified by the theorem due to
Cheney [14, Chapter 3]. Assume that the frequency pair (ω1r , ω2r ) Subject to A T w  c, (42)
represents the rth uniformly sampled frequency grid point in the where the vectors b = [0
−1 ] , T
w =[ d
T T
e T ]T , T
considered frequency bands, where 1  r  R, R is the total num-  (WU)T −(WU)T 
c = [ (Ws) T
−(Ws) ] , and the matrix A =
T T
, where
ber of the uniformly sampled frequency grid points. Next, we de- T T
−1 −1
fine a matrix U with its rth row Ur given by the R×1 column vector 0 has all entries equal to zero.
L ∞ Design Problem: From (38), the weighted L ∞ solution can be
 
Ur = 2 cos( M 2 ω1r ) sin N 2 ω2r + φd (ω1r , ω2r ) , obtained by directly solving the following minimization problem:
 
2 cos ( M 2 − 1)ω1r sin N 2 ω2r + φd (ω1r , ω2r ) , . . . , Minimize δ

sin N 2 ω2r + φd (ω1r , ω2r ) , . . . , Subject to W|Ud − s|  δ 1. (43)

2 cos( M 2 ω1r ) sin ω2r + φd (ω1r , ω2r ) , (43) can be rewritten as follows:
 
2 cos ( M 2 − 1)ω1r sin ω2r + φd (ω1r , ω2r ) , . . . ,
  Maximize −δ
sin ω2r + φd (ω1r , ω2r ) , (35)  T    
(WU)T −(WU)T d Ws
a vector s with its rth element given by Subject to  . (44)
−1 −1 δ −Ws

sr = − sin φd (ω1r , ω2r ) , (36) Then, we can obtain the standard dual form of (44) as follows:
852 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858

Maximize bT w Consider the projection operator given by (52). Substituting (50)

T into (52) yields
Subject to A w  c, (45)
 − 1
where the vectors b = [ 0T
−1 ], w = [ dT δ ]T ,
T P = I − Dx A T AD2x A T ADx . (54)
 WU )T −(WU)T

c = [ (Ws) T −(Ws)T ] T , and the matrix A = (
. Accordingly, we have from (50), (51), and (54) that
−1 −1
  − 1 
5.3. The algorithm for solving the optimization problem c̃ p = Dx I − A T AD2x A T AD2x c. (55)

Using Definition 3.1 of [15], we can obtain the standard primal Next, we define the following two vectors:
form as follows:  − 1
w = AD2x A T AD2x c and r = c − A T w, (56)
T
Minimize c x
where w represents the dual variable vector associated with the
Subject to Ax = b, x  0, (46) primal variable vector x. Hence, the primal variable xk+1 at the
(k + 1)th iteration is found by the following equation:
where x denotes the corresponding primal variable vector. Given
the vectors b, c, and matrix A, which define the L 1 problem (42), μ
xk+1 = xk − dx , (57)
the corresponding standard primal form is given by η
Minimize cT x where

Subject to Ax=b, x  0, (47) dx = D2x r and

  
where the vectors b = [ 0 T
−1 ] , c = [ (Ws) −(Ws) ] , the
T T T T T
η = max eiT c̃ p = max eiT Dx r = max eiT D2x r/xi (58)
 (WU)T −(WU)T  i i i
matrix A = , and x = [x1 , . . . , x R , x R +1 , . . . , x2R ] T
−1T −1T
corresponds to the dual variable vector w =[ d T e T ] T . with xi being the ith entry of x.
Similarly, the corresponding standard primal form of the L ∞
problem (45) is given by 5.4. Iterative procedures

Minimize cT x In this section, we apply the above PAS algorithm to solve the
L 1 and L ∞ design problems. The resulting iterative procedures are
Subject to Ax = b, x  0, (48) summarized step by step as follows:
where the vectors b = [ 0 T
−1 ] , c = [ (Ws) −(Ws)T ] T ,
T T
 (WU)T −(WU)T
 5.4.1. L 1 design
A = , and x = [x1 , . . . , x R , x R +1 , . . . , x2R ] T corre-
−1 −1 Step 1: Choose an initial guess x0 which satisfies the equality con-
sponds to the dual variable vector w = [ d T δ ] T . straints Ax0 = b and x0 > 0 shown by (47). We can simply choose
Next, we adopt the primal-form affine-scaling variant of Kar- x0 = [ (x01 ) T (x02 ) T ] T , where x01 = x02 = 1/2. And set the iteration
markar’s algorithm (the PAS algorithm) of [12] for solving (47) and number k = 0.
(48). Assume that an initial solution x which satisfies the con-  k
Step 2: Compute w = dk = (AD2x A T )−1 AD2x c according to the fol-
straints is given. Then, x is mapped into a vector y0 with all entries e
lowing process:
equal to one as follows:
(2.1): Construct the diagonal matrices D1 = diag(xk1 ) and D2 =
y = D− 1
x x, (49) diag(xk2 ),
where Dx denotes a diagonal matrix containing the entries of x. (2.2): Construct the diagonal matrices M1 = D21 + D22 and M2 =
Based on the mapping, we create D1 − D22 ,
2

(2.3): Compute the matrix K = (WU) T (M1 − M2 M−1

1 M2 )W,
à = ADx and c̃ = Dx c. (50) −
(2.4): Compute the vectors d = (KU) (Ks) and ek =
k 1
−1
To satisfy the equality constraints, we project c̃ onto the null space M1 M2 W(Udk − s).
of à to obtain Step 3: Terminate the design process if
 k −1 T 
c̃ p = Pc̃, (51)  |er | 1 − |ekr |T 1 
   ε,
 
where |ekr −1 |T 1
 − 1 where ε is a preset small number and ekr = W(Udk − s). Otherwise,
P = I − Ã T ÃÃ T Ã (52)
go to Step 4.
denotes the projection operator. Next, we move from the initial Step 4: Update the reduced cost vector r = [ r1T r2T ] T = c − A T w
y to y1 in the direction −c̃ p to reduce the transformed objective according to the following process:
function in the maximum rate according to (4.1): r1 = ek − ekr ,
c̃ p (4.2): r2 = ek + ekr .
y1 = 1 − μ , (53)  d1 
maxi (eiT c̃ p ) Step 5: Compute the feasible direction dx = d2
= D2x r as follows:
(5.1): d1 = D21 r1 ,
where the required step size μ ∈ (0, 1) is chosen so that y1 > 0,
1 represents a vector with appropriate size and all entries equal (5.2): d2 = D22 r2 .
μ
to one, and ei is a vector with appropriate size and the ith entry Step 6: Compute the step size η from (58) and update the primal
equal to one and the others equal to zero. After obtaining y1 , we  xk1+1 
variable vector according to xk+1 = = xk − μ
η dx . Then, set
then find a new feasible solution x1 for (47) and (48) by perform- xk2+1
ing the inverse mapping. k = k + 1 and go to Step 2.
 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858 853

5.4.2. L ∞ design Stopband Magnitude Mean-Squared Errors (SMSE):

Step 1: Choose an initial guess x0 which satisfies the equality con-      
straints Ax0 = b and x0 > 0 shown by (48). We can simply choose SMSE = G e j ω1 , e j ω2  − G d (ω1 , ω2 ) 2
1
x0 = [ (x01 ) T (x02 ) T ] T , where x01 = x02 = 2R . And set the iteration (ω1 ,ω2 ) ∈Ωs
number k = 0.  k /number of grid points in the stopband.
Step 2: Compute w = dk = (AD2x A T )−1 AD2x c according to the fol-
δ
lowing process: Peak Stopband Attenuation (PSA) in dB:
(2.1): Construct the diagonal matrices D1 = diag(xk1 ) and D2 =   
PSA = − max 20 log10 G e j ω1 , e j ω2  .
diag(xk2 ), (ω1 ,ω2 )∈Ωs
(2.2): Construct the diagonal matrices M1 = D21 + D22 and M2 = Passband Phase Mean-Squared Error (PPMSE):
D1 − D22 ,
2
 
  2
(2.3): Compute the scalar g = 1 T (D21 + D22 )1 = 1 T M1 1, PPMSE = arg D e j ω1 , e j ω2 − φd (ω1 , ω2 )
(2.4): Compute the matrix K = (WU) T (M1 − 1
M 11 T M2 )W, (ω1 ,ω2 ) ∈Ω p
g 2
(2.5): Compute the vectors d = k
(KU)−1 (Ks) and δ k
= /number of grid points in the passband,
1 M2 W(Udk
1 T
− s) .
g where arg{x} denotes the phase response of x.
Step 3: Terminate the design process if Stopband Phase Mean-Squared Errors (SPMSE):
δk  
  2
 ρ, SPMSE = arg D e j ω1 , e j ω2 − φd (ω1 , ω2 )
max{|ekr |}
(ω1 ,ω2 ) ∈Ωs
where ρ is a preset real number and ekr = W(Udk − s). Otherwise, /number of grid points in the stopband.
go to Step 4.
Step 4: Update the reduced cost vector r = [ r1T r2T ] T = c − A T w Peak Passband Phase Error (PPPE):
according to the following process: 
 j ω j ω  
PPPE = max arg D e 1 , e 2 − φd (ω1 , ω2 ).
(4.1): r1 = δ 1 − ekr , (ω1 ,ω2 )∈Ω p
(4.2): r2 = δ 1 + ekr .
 d1  Peak Stopband Phase Error (PSPE):
Step 5: Compute the feasible direction dx = = D2x r as follows:
d2 
 j ω j ω  
(5.1): d1 = D21 r1 , PSPE = max arg D e 1 , e 2 − φd (ω1 , ω2 ).
(ω1 ,ω2 )∈Ωs
(5.2): d2 = D22 r2 .
μ
Step 6: Compute the step size η from (58) and update the primal Example 1 (2-D diamond-shaped DC-HB filter). This example is the
k+1
 xk1+1  same as that given by [17]. We use the same design specifications:
variable vector according to x = =x − μk
η dx . Then, set
xk2+1 (L , K ) = (1, 1), the passband edge frequencies ω1p = ω2p = 0.88π
k = k + 1 and go to Step 2. and the stopband edge frequencies ω1s = ω2s = 0.12π . The desired
According to our design experience, we summarize several ob- magnitude response is given by
servations similar to those presented in [16] on the computational  |ω | |ω |
efficiency of the PAS algorithm as follows:   j ω j ω  1 for 0.881π + 0.882π  1,
G d e 1 , e 2  = (59)
|ω | |ω |
0 for 1.121π + 1.122π  1.
(1) The PAS algorithm is less complicated and more natural as
compared to other linear programming algorithms like those Thus, the magnitude characteristics possess the 2-D HB property.
used by [17] and [18]. At each iteration, the main computa-
tional complexity is due to the computation of the dual vari- Due to the 2-D DC-HB property of the proposed 2-D recur-
able vector w. sive digital filter structure, we set about half of the coefficients to
(2) The value of the objective function decreases significantly zero and consider only the frequency grid points uniformly sam-
in the early iterations. However, the decreasing speed slows pled in the passband during the design processes. A 2-D diamond-
down considerably when the current solution becomes closer shaped filter G ( z1 , z2 ) is obtained by selecting M 1 = M 2 = 10 and
to the optimal solution. N 1 + 1 = N 2 = 11. Hence, the number of independent coefficients
(3) The PAS algorithm is somewhat sensitive to primal degener- is [( M 2 + 1) N 2 /2] = 60, where [x] denotes the integer less than or
acy, especially when the iteration proceeds near the optimal equal to x. We design such a 2-D SHP DAF A ( z1 , z2 ) that the phase
solution. However, the PAS algorithm still performs quite well φ(ω1 , ω2 ) of its denominator D (z1 , z2 ) approximates the desired
even with the presence of primal degeneracy. passband phase response:
   
6. Simulation examples M1 − M2 N1 − N2
φd (ω1 , ω2 ) = ω1 + ω2 (60)
2 2
Here, we present simulation results of designing the 2-D
diamond-shaped filters illustrated in Fig. 3 for evaluation and com- which is obtained by substituting the following desired responses:
parison. For comparison, the proposed techniques and the minimax
techniques presented by [17] and [18] are also utilized for the θm,d (ω1 , ω2 ) = 0 and θ p ,d (ω1 , ω2 ) = − M ω1 − N ω2 (61)
design examples. The significant performance parameters for com- into (31). Then, the proposed design techniques discussed in Sec-
parison are defined as follows: tion 5 are applied to find the coefficients of A ( z1 , z2 ) in the L 2 , L 1 ,
Passband Magnitude Mean-Squared Errors (PMSE): and L ∞ optimal senses, respectively. Since φd (ω1 , ω2 ) satisfies the
       stability conditions discussed in Section 2.2, the stability of the
PMSE = G e j ω1 , e j ω2  − G d (ω1 , ω2 ) 2
designed A ( z1 , z2 ) is guaranteed. Moreover, each of the designed
(ω1 ,ω2 ) ∈Ω p filter possesses approximately linear phase response with the ideal
/number of grid points in the passband. phase specification given by θ p ,d (ω1 , ω2 ) = − M ω1 − N ω2 .
854 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858

Table 3
Significant design results for Example 1.

L 2 design (38) L 1 design L ∞ design

PMSE 1.6850 × 10−14 5.6947 × 10−14 1.8656 × 10−14
SMSE 1.2388 × 10−7 1.5881 × 10−7 1.4044 × 10−7
PSA (dB) 55.6789 53.3541 55.9953
PPMSE 1.2388 × 10−7 1.5881 × 10−7 1.4045 × 10−7
PPPE 1.6446 × 10−3 2.1493 × 10−3 1.5858 × 10−3
Number of 0 17 19
iterations

Table 4
The filter coefficients for Example 1 (the L ∞ design).

d(1, 1) 0.49449151390392 d(1, 7) −0.00102248600442

d(3, 1) 0.00151258074499 d(3, 7) −0.00159569452773
d(5, 1) 0.00075517612506 d(5, 7) −0.00499069288388
d(7, 1) 0.00020307777406 d(7, 7) 0.00599275812235
d(9, 1) 0.00004857145616 d(9, 7) 0.00008128170356
d(0, 2) 0.24569637139828 d(0, 8) 0.00040310859226
d(2, 2) −0.11832229358874 d(2, 8) 0.00051949413194
d(4, 2) −0.00108241088400 d(4, 8) 0.00090064226082
d(6, 2) −0.00041271349518 d(6, 8) 0.00293006593842
d(8, 2) −0.00008366874060 d(8, 8) −0.00332448561335
d(10, 2) −0.00005080282590 d(10, 8) −0.00003052357585
d(1, 3) −0.05801425239827 d(1, 9) −0.00020429593981
d(3, 3) 0.05425962816312 d(3, 9) −0.00024963824267
d(5, 3) 0.00070222441114 d(5, 9) −0.00046985556069
d(7, 3) 0.00028049732806 d(7, 9) −0.00166564324378
d(9, 3) 0.00004580441205 d(9, 9) 0.00170710628000
d(0, 4) 0.01302730953143 d(0, 10) 0.00007217401500
d(2, 4) 0.02631262295759 d(2, 10) 0.00008685130487
d(4, 4) −0.02973257940197 d(4, 10) 0.00015386543503
d(6, 4) −0.00042859706083 d(6, 10) 0.00023275511953
d(8, 4) −0.00013377927381 d(8, 10) 0.00086248740049
d(10, 4) −0.00004482202825 d(10, 10) −0.00076011137019
d(1, 5) −0.00551723888018 d(1, 11) −0.00002790655197
d(3, 5) −0.01437281055496 d(3, 11) 0.00000012501785
d(5, 5) 0.01731669452760 d(5, 11) −0.00005110755951
d(7, 5) 0.00028690016132 d(7, 11) −0.00012903172565
d(9, 5) 0.00009300996151 d(9, 11) −0.00038666388377
d(0, 6) 0.00218945141807
d(2, 6) 0.00285010936100
d(4, 6) 0.00837920066971
d(6, 6) −0.01026171913062
d(8, 6) −0.00015392826716
d(10, 6) −0.00005054466046

The parameters used for this design are listed as follows: μ =

0.97, ε = 1 × 10−8 , and ρ = 0.9999999.
# We set the frequency
weighting function to W (ω1 , ω2 ) = ω12 + ω22 in the passband.
The significant design results are listed in Table 3. Table 4 lists
the filter coefficients d(m, n) for the L ∞ design. The magnitude
and phase responses obtained by using the L ∞ design technique
presented in Section 5.4.2 are depicted in Fig. 4 and Fig. 5, re-
spectively. As compared with the PSA = 49.3 dB obtained by the
L ∞ design technique of [17] with 64 independent coefficients, the Fig. 4. Magnitude response of the L ∞ design for Example 1. (a) Perspective plot.
proposed techniques provide more satisfactory performance. (b) Contour plot.

Example 2 (2-D diamond-shaped filter). This example is similar to M 2 = 4 and N 1 + 1 = N 2 = 7. Thus, the number of independent
that considered by [18]. The desired magnitude response is given coefficients is (M 2 + 1) N 2 = 35. Using the techniques proposed in
by Section 5, we design an SHP DAF A ( z1 , z2 ) with φ(ω1 , ω2 ) of its
 |ω | |ω | denominator D ( z1 , z2 ) approximating the desired phase response
  j ω j ω  1 for π1 + 0.82π  1, φd (ω1 , ω2 ). In this case, φd (ω1 , ω2 ) is given by
G d e 1 , e 2  = (62)
0
|ω | |ω |
for 1.51π + 1.22π  1.
   
M1 − M2 N1 − N2
φd (ω1 , ω2 ) = ω1 + ω2 (63)
Clearly, the magnitude characteristics do not possess the 2-D HB 2 2
property. in the passband due to

Based on the proposed 2-D digital filter structure, a 2-D

θm,d (ω1 , ω2 ) = 0 and θ p ,d (ω1 , ω2 ) = − M ω1 − N ω2 . (64)
diamond-shaped filter G ( z1 , z2 ) is obtained by setting M 1 = In contrast, φd (ω1 , ω2 ) is given by
 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858 855

Table 5
Significant design results for Example 2.

L 2 design (38) L 1 design L ∞ design

PMSE 5.1715 × 10−10 2.9892 × 10−8 3.8802 × 10−11
SMSE 1.3422 × 10−6 2.1052 × 10−6 1.8850 × 10−6
PSA (dB) 34.9551 29.5745 46.8967
PPMSE 4.2567 × 10−6 1.6560 × 10−5 6.7191 × 10−6
SPMSE 1.3422 × 10−6 2.1054 × 10−6 1.8850 × 10−6
PPPE 4.9311 × 10−2 1.3421 × 10−1 1.3113 × 10−2
PSPE 1.7876 × 10−2 3.3217 × 10−2 4.5204 × 10−3
Number of 0 23 14
iterations

The parameters used for this design are listed as follows: μ =

0.97, ε = 1 × 10−8 , and ρ = 0.99. We set the frequency weight-
ing function W (ω1 , ω2 ) = 1 and 2 in the passband and stopband,
respectively. The significant design results are shown in Table 5.
Table 6 lists the filter coefficients d(m, n) for the L ∞ design. The
resulting magnitude and phase responses are depicted in Fig. 6
and Fig. 7, respectively. As compared with the PSA = 33.85 dB
obtained by the L ∞ design technique of [18, Fig. 4(b)], with 36
independent coefficients, the proposed techniques provide more
satisfactory performance.

7. Conclusion

This paper has presented a general 2-D recursive digital filter

structure for designing 2-D recursive interpolation/decimation fil-
ters. This structure consists of a 2-D recursive digital allpass filter
(DAF) with symmetric half-plane (SHP) support for its filter co-
efficients and a 2-D pure delay block. Therefore, it possesses a
preferable doubly complementary half-band (DC-HB) property that
reduces the number of required independent filter coefficients for
design. The design problem has been appropriately formulated to
obtain a linear optimization problem that minimizes the phase er-
ror of the 2-D recursive SHP DAF in the pth norm (L p ) sense.
Design techniques have been presented for the designs optimal in
the L 1 , L 2 , and L ∞ sense. The effectiveness of the research work
has been confirmed by simulation results.

Appendix A

Here, we show the equivalence between the minimization prob-

lems (33) and (34). Ideally, we assume that the residual between
φ(ω1 , ω2 ) and φd (ω1 , ω2 ) in the bracket of (33) is zero, i.e.,


N2
−1
− tan d(0, n) sin(nω2 )
n =1

Fig. 5. (a) Phase response φ(ω1 , ω2 ) and (b) phase error φ(ω1 , ω2 ) − φd (ω1 , ω2 ) of  
M2 N2

the L ∞ design in the passband and stopband for Example 1. + 2d(m, n) cos(mω1 ) sin(nω2 )
m =1 n =1
   
M1 − M2 N1 − N2 π  
N2
φd (ω1 , ω2 ) = ω1 + ω2 + (65)
2 2 2 d(0, 0) + d(0, n) cos(nω2 )
in the stopband due to n =1

 
M2 N2
θm,d (ω1 , ω2 ) = 0 and + 2d(m, n) cos(mω1 ) cos(nω2 )
θ p ,d (ω1 , ω2 ) = − M ω1 − N ω2 − π . (66) m =1 n =1

Again, φd (ω1 , ω2 ) satisfies the stability conditions discussed in − φd (ω1 , ω2 ) = 0. (A.1)

Section 2.2, the stability of the designed A ( z1 , z2 ) is guaranteed.
Hence,
Moreover, each of the designed filter possesses approximately lin-
ear phase response with the ideal phase specification given by

N2
θ p ,d (ω1 , ω2 ) = − M ω1 − N ω2 and θ p ,d (ω1 , ω2 ) = − M ω1 − N ω2 − π − tan −1
d(0, n) sin(nω2 )
in the passband and stopband, respectively.
n =1
856 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858

Table 6
The filter coefficients d(m, n) for Example 2 (the L ∞ design).

m=0 m=1 m=2 m=3 m=4

n=1 0.20748248666476 0.39405324429593 −0.02961316340509 0.00894638743755 −0.00163787323093
n=2 0.29418753852512 −0.06418104149017 −0.06676711106463 0.00823570465844 −0.00256273771504
n=3 −0.01098701932075 −0.06583873828102 0.03750534576446 0.01549780166167 −0.00248279202680
n=4 0.00809584993651 0.01046225684140 0.02227004227280 −0.01806088159965 −0.00346353620140
n=5 −0.00560613798252 −0.00072070415640 −0.00895915265167 −0.00604509206370 0.00646863158591
n=6 −0.00078875877845 0.00312059771101 −0.00059956480434 0.00501101156450 0.00143787295782
n=7 −0.00229663832682 0.00009140882254 −0.00196080329515 −0.00021532295926 −0.00220542349238

Fig. 6. Magnitude response of the L ∞ design for Example 2. (a) Perspective plot. Fig. 7. (a) Phase response φ(ω1 , ω2 ) and (b) phase error φ(ω1 , ω2 ) − φd (ω1 , ω2 ) of
(b) Contour plot. the L ∞ design in the passband and stopband for Example 2.
 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858 857


 
M2 N2
As a result, the assumption of (A.1) is equivalent to that shown
+ 2d(m, n) cos(mω1 ) sin(nω2 ) by (A.6). This reveals that the residual in the bracket of (33) is
m =1 n =1 equivalent to that of (34).
 
N2
d(0, 0) + d(0, n) cos(nω2 ) References
n =1

 
M2 N2 [1] P.P. Vaidyanathan, S.K. Mitra, Y. Neuvo, A new approach to the realization
+ 2d(m, n) cos(mω1 ) cos(nω2 ) of low sensitivity IIR digital filters, IEEE Trans. Acoust., Speech, Signal Pro-
cess. 34 (2) (Apr. 1986) 350–361.
m =1 n =1
[2] P.P. Vaidyanathan, P. Regalia, S.K. Mitra, Design of doubly complementary IIR
= φd (ω1 , ω2 ). (A.2) digital filters using a single complex allpass filter, with multirate applications,
IEEE Trans. Circuits Syst. 34 (4) (Apr. 1987) 378–389.
Taking the tangent function for both side of (A.2), we obtain [3] M. Ikehara, M. Funaishi, H. Kuroda, Design of complex all-pass networks us-
ing Remez algorithm, IEEE Trans. Circuits Syst. II: Analog Digital Signal Pro-

N2
cess. 39 (8) (Aug. 1992) 549–556.
− d(0, n) sin(nω2 ) [4] S.S. Lawson, A. Klouche-Djedid, Technique for design of two-channel approxi-
n =1 mately linear phase QMF bank and its application to image compression, IEE
 Proc. – Vision, Image Signal Process. 148 (2) (Apr. 2001) 85–92.
 
M2 N2
[5] J.-H. Lee, Y.-H. Yang, Design of two-channel linear-phase QMF banks based on
+ 2d(m, n) cos(mω1 ) sin(nω2 ) real IIR allpass filters, IEE Proc. – Vision, Image Signal Process. 150 (5) (Oct.
m =1 n =1 2003) 331–338.
 
N2
[6] J.-H. Lee, Y.-H. Yang, Minimax design of two-channel nonuniform-division filter
banks using IIR allpass filters, IEEE Trans. Signal Process. 52 (11) (Nov. 2004)
d(0, 0) + d(0, n) cos(nω2 ) 3227–3240.
n =1 [7] H. Toyoshima, M. Ikehara, S. Takahashi, A new class of 2-D digital filters com-
 posed of all-pass subfilters, Electron. Commun. Jpn., Part 3 73 (4) (1990) 91–98.
 
M2 N2
[8] H. Safiri, M. Ahmadi, V. Ramachandran, Design of stable, causal 2-D digital
+ 2d(m, n) cos(mω1 ) cos(nω2 ) filters using real coefficient 2-D allpass building blocks, IEEE Trans. Circuits
m =1 n =1 Syst. II: Analog Digital Signal Process. 44 (5) (May 1997) 409–412.

 [9] D.E. Dudgeon, R.M. Mersereau, Multidimensional Digital Signal Processing,
= tan φd (ω1 , ω2 ) . (A.3) Prentice Hall, Englewood Cliffs, NJ, 1984.
[10] E. Dubois, The sampling and reconstruction of time-varying imagery with ap-
It follows from (A.3) that
plication in video systems, Proc. IEEE 73 (4) (Apr. 1985) 502–522.
[11] S.C. Pei, J.J. Shyu, Eigenfilter design of 1-D and 2-D IIR digital all-pass filters,

N2
 IEEE Trans. Signal Process. 42 (4) (Apr. 1994) 966–968.
d(0, n) sin(nω2 ) cos φd (ω1 , ω2 ) [12] I. Adler, N. Karmarkar, M.G.C. Resende, G. Veiga, An implementation of Kar-
n =1 markar’s algorithm for linear programming, Math. Programming 44 (1989)
297–335.
 
M2 N2
 [13] J.-H. Lee, Y.-H. Yang, Two-channel parallelogram QMF banks using 2-D NSHP
+ 2d(m, n) cos(mω1 ) sin(nω2 ) cos φd (ω1 , ω2 ) digital all-pass filters, IEEE Trans. Circuits Syst. I: Regular Papers 57 (9) (Sept.
m =1 n =1 2010) 2498–2508.

+ d(0, 0) sin φd (ω1 , ω2 ) [14] E.W. Cheney, Introduction to Approximation Theory, McGraw–Hill, New York,
1966.

N2
 [15] C.H. Papadimitriou, K. Steiglitz, Combinatorial Optimization: Algorithms and
+ d(0, n) cos(nω2 ) sin φd (ω1 , ω2 ) Complexity, Prentice Hall, New Jersey, 1982.
[16] S.-C. Fang, S. Puthenpura, Linear Optimization and Extension, Prentice Hall, En-
n =1
glewood Cliffs, NJ, 1993.
 
M2 N2
 [17] P. Siohan, V. Ouvrard, Design of optimal nonseparable 2-D half-band filters,
+ 2d(m, n) cos(mω1 ) cos(nω2 ) sin φd (ω1 , ω2 ) = 0. Proc. IEEE Int. Symp. Circuits Syst. 1 (1993) 914–917.
[18] P. Carrai, G.M. Cortelazzo, G.A. Mian, Characteristics of minimax FIR filters for
m =1 n =1
video interpolation/decimation, IEEE Trans. Circuits Syst. Video Technol. 4 (5)
(A.4) (Oct. 1994) 453–467.

(A.4) can be rewritten as follows:


N2
 Ju-Hong Lee received his B.S. degree in electrical engineering from
d(0, n) sin nω2 + φd (ω1 , ω2 ) the National Cheng-Kung University, Tainan, Taiwan, in 1975, the M.S. de-
n =1 gree in electrical engineering from the National Taiwan University (NTU),
 
M2 N2 Taipei, Taiwan, in 1977, and the Ph.D. degree in electrical engineering from

+ 2d(m, n) cos(mω1 ) sin nω2 + φd (ω1 , ω2 ) Rensselaer Polytechnic Institute, Troy, NY, in 1984. From September 1980
to July 1984, he was a Research Assistant with the Department of Electri-
m =1 n =1
 cal, Computer, and Systems Engineering, Rensselaer Polytechnic Institute,
+ d(0, 0) sin φd (ω1 , ω2 ) = 0. (A.5) where he was involved in research on multidimensional recursive digi-
tal filtering. From August 1984 to July 1986, he was a Visiting Associate
Without loss of generality, we set d(0, 0) = 1 to obtain Professor with the Department of Electrical Engineering, NTU, where he
became an Associate Professor in August 1986 and has been a Professor

N2
 since August 1989. He has been appointed NTU’s Tenured Distinguished
d(0, n) sin nω2 + φd (ω1 , ω2 ) Professor since August 2006. He was appointed Visiting Professor with
n =1 the Department of Computer Science and Electrical Engineering, Univer-
sity of Maryland, Baltimore, during a sabbatical leave in 1996. His current
 
M2 N2

+ 2d(m, n) cos(mω1 ) sin nω2 + φd (ω1 , ω2 ) research interests include multidimensional digital signal processing, im-
age processing, detection and estimation theory, analysis and processing
m =1 n =1
 of joint vibration signals for the diagnosis of cartilage pathology, statisti-
+ sin φd (ω1 , ω2 ) = 0. (A.6) cal signal processing, and adaptive signal processing for smart antennas
858 J.-H. Lee, Y.-H. Yang / Digital Signal Processing 22 (2012) 847–858

with applications in mobile wireless communication systems. He has been
a Member of the Editorial Board of the International Journal of Antennas and
Propagation since January 2009.
Dr. Lee received Outstanding Research Awards from the National Sci-
ence Council (NSC) of Taiwan in the academic years of 1988, 1989, and
1991–1994; Distinguished Research Awards from the NSC in the academic
years of 1998–2004; and the NSC Research Fellowship for the academic
years of 2005–2008 and 2011–2014, respectively.
Yuan-Hau Yang received his B.S. degree from the Department of Com-
munication Engineering, National Chiao-Tung University, Hsinchu, Taiwan,
in 2000 and the M.S. degree and the Ph.D. degree in communication engi-
neering from the National Taiwan University, Taipei, Taiwan, in 2003 and
2008, respectively. Since 2008, he has been with Novatek Microelectronics
Corporation, Hsinchu, Taiwan. His current research interests include VLSI
signal processing, digital filter design, and audio effects for TV applica-
tions.