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Closed-Form Design of FIR Frequency Selective

Filter Using Discrete Sine Transform

Chien-Cheng Tseng Su-Ling Lee


Depart. of Computer and Communication Engineering Depart. of Computer Sci. and Information Engi.
National Kaohsiung First University of Sci. and Tech. Chang-Jung Christian University
Kaohsiung, Taiwan Tainan, Taiwan
tcc@nkfust.edu.tw lilee@mail.cjcu.edu.tw

Abstract—In this paper, the closed-form design of FIR they need to solve matrix inverse or require an optimization
frequency selective filter (FSF) using discrete sine tool box. Therefore, it is meaningful to study the closed-form
transform (DST) is studied. First, the DST-based design of frequency selective filter such that the filter
frequency selective method is used to obtain the filtered coefficients can be easily computed without performing any
signal from the given digital signal. Then, the transfer optimization and matrix inversion. The closed-form method is
function of FSF is derived from the filtered signal by using very suitable for the design of long-length FIR filter because it
index mapping approach. Because the closed-form design does not take much time.
is obtained, the filter coefficients are easily computed On the other hand, A.K. Jain has shown that for a first-order
without performing any optimization. Finally, the long- Markov sequence, with given boundary conditions, the
length low-pass, band-pass and high-pass filter design Karhunen-Loeve transform, which is known to be statistically
examples are used to show the effectiveness of the optimal, reduces to the discrete sine transform (DST) [4].
proposed DST method. Until now, DST has been successfully applied to digital image
sharpening [5], design of fixed fractional Hilbert transformer
Keywords—frequency selective filter; discrete sine transform; [6], transform-domain adaptive filtering [7], signal
digital filter; closed-form; FIR filter interpolation [8], image coding and compression [9], speech
enhancement [10], underwater target recognition [11], EEG
I. INTRODUCTION signal classification [12] and tridiagonal commuting matrices
In the digital signal processing applications, it is desirable [13]. Moreover, some fast computation algorithms have been
to design a digital filter that passes certain frequency presented to compute the DST including sparse-matrix
components and totally rejects all others. This type of filter is factorization, fast recursive algorithm and prime factor
called as the frequency selective filter (FSF) which has been decomposition etc [14][15]. Because the DST is highly
widely used in communication system, image processing, computation-intensive, several efficient architectures have
speech processing, audio processing and biomedical been also suggested for its implementation in very large-scale
engineering etc. The ideal frequency response of the frequency integration (VLSI) systems [16].
selective filter is given by In this paper, the DST method is used to design frequency
­1 ω1 ≤ ω ≤ ω 2 selective filter. This paper is organized as follows. In section
Fd (ω ) = ® (1) II, the type I DST is applied to design digital frequency
¯0 others selective filter by using index mapping. Because the closed-
where [ω1 , ω 2 ] is the frequency band in which frequency form design is obtained, the filter coefficients are easily
components are selected to pass the system. If ω 1 is equal to computed without performing any optimization. This leads
that the proposed method can be used to design very long-
zero and ω 2 is the band-edge frequency, the FSF is a low- length FIR filter without taking much time. In section III, the
pass filter. Moreover, if ω 1 is the band-edge frequency and long-length low-pass, band-pass and high-pass filter design
ω 2 is equal to π , the FSF is a high-pass filter. In this paper, examples are used to show the effectiveness of the proposed
the frequency selective filter design problem is how to find a DST method. Finally, a conclusion is made.
digital filter such that its actual frequency response fits the II. DESIGN METHOD
ideal response F d (ω ) as well as possible. The existing
In the literature, there are four types of discrete sine
design techniques of FIR frequency selective filters are transforms, namely DST-I, DST-II, DST-III and DST-IV [13].
window method, least-squares (LS) method and mini-max In this section, the DST-I design method is presented. Given
method [1]-[3]. The window method is simple, but it not the sequence x(0), x(1), ..., x(N-1) which are sampled from
optimal. Although the LS and mini-max methods are optimal, continuous-time signal x (t ) , let us first study how to select

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the frequency components of the signal x (t ) from the When a signal s (n ) passes through this FIR filter, its output
discrete-time signal x (n ) by using DST-I. The DST-I is is given by the difference equation:
N −1
defined as
2 N −1
y (n) = ¦ g (A ) s (n − A ) (10)

¦
A=0
X (k ) = x ( n ) sin( ( n +1N)( +k 1+1)π ) (2a)
Now, the problem is how to use the formula in (6) to
N + 1 n=0
determine filter coefficients g (A ) such that the filter output
2 N −1
x(n) = ¦ X ( k ) sin( ( n +1N)(+k1+1)π )
N + 1 k =0
(2b) y (n ) is almost equal to the delayed frequency selected signal
FS [ s ( n − τ )] ; that is,
If two band-edge frequencies ω1 and ω 2 satisfy the y ( n ) ≈ FS [ s ( n − τ )] (11)
following expressions: In what follows, an index mapping method is used to solve
( K 1 + 1)π this problem. If we choose
ω1 = (3a) s ( n ) = x ( N − 1)
N +1
( K + 1)π s ( n − 1) = x ( N − 2 ) (12)
ω2 = 2 (3b) #
N +1
then output signal FS [ x ( n )] that x (n ) passes through the s ( n − N + 1) = x ( 0 )
frequency selective filter with the ideal frequency response in then equations (6) and (10) can be linked together. Equation
(1) can be approximated by (12) can be re-expressed as the form
2 K2 x(m) = s (n − ( N − 1) + m) 0 ≤ m ≤ N − 1 (13)
FS [ x ( n )] = ¦ X ( k ) sin( ( n +1N)( +k1+1)π ) (4)
N + 1 k = K1 Substituting the equation (13) and x(t ) = s (n − ( N − 1) + t )
into (6), we get
That is, the frequency components in [ω1 , ω 2 ] are selected N −1
and the others are all rejected. Substituting (2a) into (4), it FS [ s ( n − ( N − 1) + t )] = ¦ s ( n − ( N − 1) − m )φ ( m, t ) (14)
yields the result m =0

FS[ x(n)] Let A = ( N − 1) − m , the expression becomes


N −1
2 K2 § 2 N −1 · (5) FS[s(n − ( N − 1) + t )] = ¦ s(n − A)φ ( N − 1 − A, t )
= ¦ ¨
¨ ¦
N + 1 k =K1© N + 1 m=0
x(m) sin(( m+1N)(+k1+1)π ) ¸¸ sin(( n+1N)(+k1+1)π )
A=0
(15)
¹
Moreover, let τ = ( N − 1) − t , equation (15) reduces to
N −1 ­ 2 K2 ½
= ¦ x(m)® ¦ sin ( ( m+1)( k +1)π
N +1 )sin( ( n+1)( k +1)π
N +1 )¾
N −1
FS [ s (n − τ )] = ¦ s (n − A)φ ( N − 1 − A, N − 1 − τ ) (16)
m=0 ¯ N + 1 k =K1 ¿ A =0
Replacing the discrete-time variable n by the continuous-time Comparing (10) with (16), it can be seen that if we choose
variable t , the filtered signal of x (t ) is given by g (A ) = φ ( N − 1 − A, N − 1 − τ ) (17)
N −1
FS [ x ( t )] = ¦ x ( m )φ ( m , t ) (6) then it yields the result
N −1
m=0
FS[s(n − τ )] = ¦ g (A)s(n − A) (18)
where the function φ ( m , t ) is given by A=0

φ (m, t ) =
2 K2
¦ sin
N + 1 k = K1
( ( m + 1)( k + 1) π
N +1
)sin ( ( t + 1)( k + 1) π
N +1
) (7) that is, Eq.(11) is valid. Substituting (7) into (17), the filter
coefficients are given by
g (A )
In the following, let us use the result in (7) to obtain the 2 K2 § ( N − A)( k + 1)π · § ( N − τ )( k + 1)π · (19)
transfer function of the delayed frequency selective filter = ¦ sin ¨
N + 1 k = K1 © N +1
¸ sin ¨
¹ © N +1
¸
¹
whose frequency response approximates the following ideal
response well: Finally, two remarks are described below. Firstly, filter
Gd (ω ) = Fd (ω )e − jωτ (8) coefficients can be modified by the Hamming window
where τ is a prescribed delay. Here, the filter delay is
§ 2π A ·
w ( A ) = 0 . 54 − 0 . 46 cos ¨ ¸ (20)
incorporated into the ideal frequency response in order to get a © N −1¹
causal FIR filter design. The causal transfer function of FIR to get the window-based design; that is, the modified filter
filter is given by coefficients are given by
N −1
g w (A ) = g (A ) w (A ) (21)
G (z) = ¦ g (A ) z
A=0
−A (9)
Secondly, the closed-form design in (19) is obtained, so the
filter coefficients are easily computed without performing any

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optimization. Thus, the proposed method can be used to The filter coefficients of this conventional window design are
design very long-length FIR filter without taking much time. g w ( A ) = g i ( A ) w ( A ) . When the design parameters are
III. DESIGN EXAMPLES chosen as N = 999 , τ = 500 and ω 2 = 0 . 5π , Fig.2(a)(b)
In this section, several numerical examples are used to show the designed results (solid line) of the this conventional
demonstrate the effectiveness of the proposed DST method. window method. The error Err of conventional window
To evaluate the performance, the integral squares error of method is equal to 0 . 00121 which is larger than the error
frequency response is defined by 0 . 00114 of the DST design method in example 1. Thus, the
π 2 proposed DST method has a smaller design error than the
Err = ³0
G ( e j ω ) − G d (ω ) d ω (22) conventional window method.
Obviously, the smaller the error Err is, the better
performance of DST method has.
Example 1: In this example, let us study the performance of
long-length low-pass filter design. The design parameters for
low-pass filter design are chosen as N = 999 ,
τ = ( N + 1) / 2 = 500 , ω1 = 0 and ω 2 = 0 .5π . Thus, the
number K 1 = 0 and K 2 = 499 . The filter length is equal to
N + 1 = 1000 . Fig.1(a) shows the magnitude responses
(solid line) of the long-length low-pass filter G ( z ) . The
dashed line is the unity magnitude response. So, the
specification is fitted well. Fig.1(b) depicts the group delay
response. The dashed line is the ideal response τ . It can be
seen that the specification is approximated well in pass-band.
The group delay error in stop-band can be ignored because the
magnitude response in stop-band is almost equal to zero. Fig.2 The designed results of the conventional window
Moreover, the error Err in this design is equal to 0 . 00114 . method in low-pass filter case with N = 999 and τ = 500 .
(a) Magnitude response. (b) Group delay response.

Example 3: In this example, let us study the performance of


long-length band-pass filter design. The design parameters for
band-pass filter design are chosen as N = 999 ,
τ = ( N + 1) / 2 = 500 , ω 1 = 0 .2π and ω 2 = 0 .8π . Thus,
the number K 1 = 199 and K 2 = 799 . The filter length is
equal to N + 1 = 1000 . Fig.3(a) shows the magnitude
responses (solid line) of the long-length band-pass filter
G ( z ) . The dashed line is the unity magnitude response. So,
the specification is fitted well. Fig.3(b) depicts the group delay
response in band-pass filter design. The dashed line is the
ideal response τ . It can be seen that the specification is
Fig.1 The designed results of the proposed DST based low- approximated well in pass-band. Moreover, the error Err in
pass filter with N = 999 and τ = 500 . (a) Magnitude this design is equal to 0 . 00229 .
response. (b) Group delay response. Example 4: In this example, let us study the performance of
long-length high-pass filter design. The design parameters for
Example 2: In this example, we will compare the proposed high-pass filter design are chosen as N = 999 ,
DST-based method with the conventional window method in τ = ( N + 1) / 2 = 500 , ω 1 = 0 .5π and ω 2 = π . Thus,
low-pass filter design, that is, ω1 = 0 . Taking the inverse the number K 1 = 499 and K 2 = N = 999 . The filter
discrete-time Fourier transform of G d (ω ) in (8), the ideal length is equal to N + 1 = 1000 . Fig.4(a) shows the
impulse response of low-pass filter is given by magnitude responses (solid line) of the long-length high-pass
1 ω2 filter G ( z ) . The dashed line is the unity magnitude response.
g i (A) =
2π ³− ω 2
G d (ω )e jωA dω
(23)
So, the specification is fitted well. Fig.4(b) depicts the group
sin(ω 2 (A − τ )) delay response in high-pass filter design. The dashed line is
= the ideal response τ . It can be observed that the specification
π (A − τ )

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is approximated well in pass-band. Moreover, the error Err REFERENCES
in this design is equal to 0 . 00114 . [1] A.V. Oppenheim and R.W. Schafer, Discrete-Time Signal Processing,
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IV. CONCLUSIONS Circuits and Systems-II: Express Briefs, pp.262-266, Mar. 2007.
In this paper, a DST method has been presented to design
FIR frequency selective filter. Because the closed-form design
is obtained, the filter coefficients are easily computed without
performing any optimization. The proposed method is suitable
for us to design very long-length FIR filter without taking
much time. Numerical examples have been shown to
demonstrate the effectiveness of this design method. However,
only one-dimensional design is studied here, so it is interesting
to extend this DST method to design two dimensional FIR
frequency selective filters in the future. Moreover, except DST,
other unitary transforms can be also used to design FIR
frequency selective filters.

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