Professional Documents
Culture Documents
POLYNOMIAL-BASED INTERPOLATION
FILTERS FOR DSP APPLICATIONS
DESIGN, IMPLEMENTATION, AND APPLICATIONS
POLYNOMIAL-BASED INTERPOLATION
FILTERS FOR DSP APPLICATIONS
DESIGN, IMPLEMENTATION, AND APPLICATIONS
Contents
1. Interpolation Filters
2. Fractional-Delay Filters
3. Lagrange Interpolation
4. Analog Model for Interpolation Filter
5. Polynomial-Based Interpolation Filters
6. Design
7. Applications
Tampere University of Technology 3
INTERPOLATION FILTERS
Terminology pitfalls:
Mathematical interpolation
vs. interpolation in DSP
vs. decimation
nl µl
Fig. 2. Simplified block diagram for interpolation filter.
FD FILTERS
Example: FD Filter
0.9
0.8
Amplitude Response
0.7
0.6
0.5
0.4
0.3
0.2
0.1
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Angular Frequency ω / π
3.5
3.4
Phase Delay Response
3.3
3.2
3.1
LAGRANGE INTERPOLATION
µlTin
nl (nl+µl)Tin nl+1
LINEAR INTERPOLATION
= x(n)
ya(t)
= y(l)
µlTin
nl (nl+µl)Tin nl+1
µlTin
nl (nl+µl)Tin nl+1
Sample at
tl = ( n l + µ l ) Tin
f
Fin/2
Fig. 7. The spectrum of the original continuous-time signal.
f
Fin/2 Fin 2Fin
Fig. 8. The spectrum of the input signal x(n) and the frequency
response of the reconstruction filter ha(t).
Tampere University of Technology 17
∑
m
ha (( k + µl ) Tin ) = cˆm ( k )µ l
m =0
(7)
∑ ∑ ∑
m m
y (l ) = x ( nl − k + N / 2 ) cˆm ( k − N / 2)µ l = v m ( nl ) µ l
(8)
k =0 m =0 m =0
where
N −1
vm ( nl ) = ∑ x(n − k + N / 2)cˆ
k =0
l m (k − N / 2)
(9)
x(nl)
^ ^ ^ ^
CM(z) C2(z) C1(z) C0(z)
y(l)
µl
x(nl + N/2)
cM(0) c1(0) c0(0)
Fin
−1 −1 −1
Z Z Z
cM(1) c1(1) c0(1)
−1 −1 −1
Z Z Z
POLYNOMIAL-BASED INTERPOLATION
FILTERS
(2t − nT ) m
− 1 for nT ≤ t < (n + 1)T
f m ( n, T , t ) = T (12)
0 otherwise,
Tampere University of Technology 23
cm ( n ) = ( −1) m cm ( −n − 1) (13)
1
Amplitude
−1
−2 −1 0 1 2
Time in Tin
0.7
0 (a)
−0.2
0.6
0
(b)
−0.6
0.17
0 (c)
−0.14
0.1
0 (d)
−0.1
1
Overall impulse response ha ( t )
(e)
0
−0.2
−8 −6 −4 −2 0 2 4 6 8
t/T
where
nl = lTout Tin and µ l = lTout Tin − nl . (19)
Tampere University of Technology 26
∑c
m
ha (( n + µl )Tin ) = µ − 1)
m ( n )( 2 l
(20)
m =0
1 1 1 1
Amplitude
−1 −1 −1 −1
0 1 0 1 0 1 0 1
Time in T
in
where
N −1
v m ( nl ) = ∑ x( nl − k + N / 2)c m (k ) (22)
k =0
Tampere University of Technology 27
x(nl)
y(l)
2µl-1
x(nl + N/2)
cM(0) c1(0) c0(0)
Fin
−1 −1 −1
Z Z Z
cM(1) c1(1) c0(1)
−1 −1 −1
Z Z Z
where
((2t − nT ) T − 1)m for n T ≤ t < (n + 1)T
f m (n, Tout , t ) = out out out out (25)
0 otherwise.
By substituting the resulting ha(t) into (3), the lth output sample can
be expressed, after some manipulations,
M N −1
y (l ) = ∑ ∑ c m (n)v m (n, l ), (26)
m=0 n=0
where
k up ( n , l )
∑ x(k )(2µ k − 1)
m
v m ( n, l ) = (27)
k = k low ( n , l )
with
k low (n, l ) = (l + N / 2 − n − 1) Tout Tin , (28)
s ( n, l ) − 1 if s (n, l ) is an integer
k up (n, l ) = (29)
s (n, l ) otherwise,
s (n, l ) = (l + N / 2 − n)Tout / Tin , (30)
and
µ k = kTin / Tout − kTin / Tout . (31)
2µ k − 1
x(k)
Fin
−1 −1 −1
I&D
Z Z Z
ov(l)
STRUCTURE CONSISTING OF A
CASCADE OF A LINEAR-PHASE FIR
INTERPOLATOR AND A MODIFIED
FARROW STRUCTURE
• The proposed structure consists of two basic building
blocks. The first block is a linear-phase FIR filter
transfer function of the following form:
KI
H I ( z ) = ∑ hI ( k ) z − k , (32)
k =0
DESIGN
Lagrange Interpolation
Example
1.2
1
Impulse response ha(t)
0.8
0.6
0.4
0.2
−0.2
−2 −1 0 1 2
Time in Tin
Fig. 19. Impulse response ha(t) for the cubic (solid line) and linear
(dashed line) Lagrange interpolation filters
−10
Magnitude in dB
−20
−30
−40
−50
−60
0 0.5 1 1.5 2 2.5 3 3.5 4
Frequency in F
in
Fig. 20. Magnitude responses for the cubic (solid line) and linear
(dashed line) Lagrange interpolation filters
Tampere University of Technology 40
−10
−20
M=1
Magnitude in dB
−30
−40
M=9
−50
−60
−70
−80
−90
0 0.5 1 1.5 2 2.5 3 3.5 4
Frequency in F
in
Fig. 21. The magnitude responses |Ha( f)| of the Lagrange
interpolation filters with degree M = 1, 3, 5, 7, and 9 (solid
lines) and the spectrum Xs( j2π f) of the input signal with
the bandwidth of 0.1Fin (dashed-line).
Tampere University of Technology 42
δ 2 = ∫ {W ( f )[H a ( f ) − D ( f )]} df
2
X (42)
Example I (minimax)
• Design parameters are:
− M = 7, N = 24, fp =0.4Fin, fs =0.6Fin, weights:
Wp=0.002, and Ws=1
−20
Magnitude in dB
−40
−60
−80
−100
0.8
Impulse response h (t)
a
0.6
0.4
0.2
−0.2
−12 −10 −8 −6 −4 −2 0 2 4 6 8 10 12
Time in T
in
Tampere University of Technology 46
Example II (L2-norm)
• Design parameters are:
− M = 5, N = 8, fp =0.35Fin, fs =0.65Fin, weights:
Wp=0.02, and Ws=1
−10
−20
Magnitude in dB
−30
−40
−50
−60
−70
−80
−90
0 0.5 1 1.5 2 2.5 3 3.5 4
Frequency in F
in
0.8
Impulse response ha(t)
0.6
0.4
0.2
−0.2
−4 −3 −2 −1 0 1 2 3 4
Time in Tin
Tampere University of Technology 47
APPLICATIONS
where
FC = min( Fin , Fout ). (45)
Tampere University of Technology 48
f
Fout/2 Fout 2Fout
Fig. 22. The spectrum of the output signal y(l) and the frequency
response of the filter ha(t).
Tampere University of Technology 49
Practical Criteria
[F / 2, ∞ ) for Type A
C
Ω s = [FC − f p , ∞ ) for Type B (48)
∞
∪ [kFC − f p , kFC + f p ] for Type C.
k =1
Direct design
0 TF
PTF
−20
Magnitude in dB
−40
−60
−80
−100
0 1 2 3 4 5 6 7 8
Frequency relative to F
out
0 FIR 1
FIR 2
Farrow
−20 Overall
Magnitude in dB
−40
−60
−80
−100
−120
0 1 2 3 4 5 6 7 8
Frequency in Fout
Fig. 24. The magnitude responses for both FIR filters in the two-
stage decimator, the Farrow structure, and the overall system.
Tampere University of Technology 54
∑ vm ( n )[2 µ − 1]
m
ya (t ) t = ( n + µ )Tin = p( n, µ ) =
(50)
m =0
d ya (t ) d p ( n, µ ) M
∑ vm ( n )2m[2 µ − 1] .
m −1
t = ( n + µ )Tin = =
dt dµ m=0
(51)
|Ha(ω)| |jωHa(ω)|
1 1
ω ω
Fig. 26. Ideal frequency responses for interpolator and
differentiator.
Tampere University of Technology 57
0.5
Amplitude
−0.5
−1
−1.5
−4 −3 −2 −1 0 1 2 3 4
Time / T
2.5
(a)
2
1.5
Amplitude
0.5
0
0 0.5 1 1.5 2 2.5 3
Frequency f / Fs
nl µl
∼ Fin=1/ Tin
Timing
estimation
DESIGN EXAMPLE
L2 design
−10
−20
Magnitude in dB
−30
−40
−50
−60
−70
−80
0 0.5 1 1.5 2 2.5 3 3.5 4
Frequency in Fin
Fig. 30. The magnitude response for the L2 interpolator (solid line)
and the amplitude spectrum of the reconstructed signal ya(t) (dark
area).
Tampere University of Technology 61
−10
−20
Magnitude in dB
−30
−40
−50
−60
−70
−80
0 0.5 1 1.5 2 2.5 3 3.5 4
Frequency in F
in
Fig. 31. The magnitude response for the linear interpolation filter
(solid line) and the amplitude spectrum of the reconstructed signal
ya(t) (dark area).
Tampere University of Technology 62
−10
−20
Magnitude in dB
−30
−40
−50
−60
−70
−80
0 0.5 1 1.5 2 2.5 3 3.5 4
Frequency in F
in
Some remarks
References
[1] T. I. Laakso, V. Välimäki, M. Karjalainen, and U. K. Laine, “Splitting the unit delay,” IEEE Signal
Processing Magazine, vol. 13, pp. 30-60, Jan. 1996.
[2] C. W. Farrow, “A continuously variable digital delay element,” in Proc. IEEE Int. Symp. Circuits
& Syst., Espoo, Finland, June 1988, pp. 2641-2645.
[3] F. M. Gardner, “Interpolation in digital modems - Part I: Fundamentals,” IEEE Trans. Commun.,
vol. 41, pp. 501-507, Mar. 1993.
[4] L. Erup, F. M. Gardner, and R. A. Harris, “Interpolation in digital modems - Part II: Implementa-
tion and performance,” IEEE Trans. Commun., vol. 41, pp. 998-1008, June 1993.
[5] D. Kincaid and W. Cheney, Numerical Analysis. Pacific Grove, 1991.
[6] J. Vesma and T. Saramäki, “Interpolation filters with arbitrary frequency response for all-digital
receivers,” in Proc. IEEE Int. Symp. Circuits & Syst., Atlanta, GA, May 1996, pp. 568-571.
[7] J. Vesma, M. Renfors, and J. Rinne, “Comparison of efficient interpolation techniques for symbol
timing recovery,” in Proc. IEEE Globecom 96, London, UK, Nov. 1996, pp. 953-957.
[8] J. Vesma and T. Saramäki, “Optimization and efficient implementation of FIR filters with
adjustable fractional delay,” in Proc. IEEE Int. Symp. Circuits & Syst., Hong Kong, June 1997, pp.
2256-2259.
[9] H. Ridha, J. Vesma, T. Saramäki, and M. Renfors, “Derivative approximations for sampled signals
based on polynomial interpolation,” in Proc. 13th Int. Conf. on Digital Signal Processing,
Santorini, Greece, July 1997, pp. 939-942.
[10] H. Ridha, J. Vesma, M. Renfors, and T. Saramäki, “Discrete-time simulation of continuous-time
systems using generalized interpolation techniques,” in Proc. 1997 Summer Computer Simulation
Conference, Arlington, Virginia, USA, July 1997, pp. 914-919.
[11] V. Tuukkanen, J. Vesma, and M. Renfors, “Combined interpolation and maximum likelihood
symbol timing recovery in digital receivers,” to be presented in 1997 IEEE Int. Conference on
Universal Personal Communications, San Diego, CA, USA, Oct. 1997.
[12] T. Saramäki and M. Ritoniemi, "An efficient approach for conversion between arbitrary sampling
frequencies," in Proc. IEEE Int. Symp. Circuits & Syst., Atlanta, GA, May 1996, pp. 285-288.
[13] J. Vesma, R. Hamila, T. Saramäki, and M. Renfors, “Design of polynomial interpolation filters
based on Taylor series,” in Proc. IX European Signal Processing Conf., Rhodes, Greece, Sep.
1998, pp. 283-286.
[14] J. Vesma, R. Hamila, M. Renfors, and T. Saramäki, “Continuous-time signal processing based on
polynomial approximation,” in Proc. IEEE Int. Symp. on Circuits and Systems, Monterey, CA,
USA, May 1998, vol. 5, pp. 61-65.
[15] D. Fu and A. N. Willson, Jr., “Interpolation in timing recovery using a trigonometric polynomial
and its implementation,” in IEEE Globecom 1998 Communications Mini Conference Record,
Sydney, Australia, Nov. 1998, pp. 173−178.
[16] f. harris, “Performance and design considerations of Farrow filter used for arbitrary resampling,” in
Proc. 13th Int. Conf. on Digital Signal Processing, Santorini, Greece, July 1997, pp. 595−599.
[17] G. Oetken, “A new approach for the design of digital interpolation filters,” IEEE Trans. Acoust.,
Speech, Signal Process., vol. ASSP−27, pp. 637−643, Dec. 1979.
[18] T. A. Ramstad, “Digital methods for conversion between arbitrary sampling frequencies,” IEEE
Trans. Acoust. Speech, Signal Processing, vol. ASSP−32, pp. 577−591, June 1984.
[19] T. A. Ramstad, “Fractional rate decimator and interpolator design,” in Proc. IX European Signal
Processing Conf., Rhodes, Greece, Sep. 1998, pp. 1949−1952.
Tampere University of Technology 65
[20] R. W. Schafer and L. R. Rabiner, “A digital signal processing approach to interpolation,” Proc.
IEEE, vol. 61, pp. 692−702, June 1973.
[21] M. Unser, A. Aldroubi, and M. Eden, “Fast B-spline transforms for continuous image
representation and interpolation,” Trans. Pat. Anal., Mach. Int., vol. 13, pp. 277−285, Mar. 1991.
[22] M. Unser, A. Aldroubi, and M. Eden, “Polynomial spline signal approximations: Filter design and
asymptotic equivalence with Shannon’s sampling theorem,” IEEE Trans. Information Theory, vol.
38, pp. 95−103, Jan. 1992.
[23] J. Vesma, Timing Adjustment in Digital Receivers Using Interpolation. M.Sc. Thesis, Tampere,
Finland: Tampere University of Tech., Department of Information Technology, Nov. 1995.
[24] V. Välimäki, Discrete-Time Modeling of Acoustic Tubes Using Fractional Delay Filters. Doctoral
thesis, Espoo, Finland: Helsinki University of Technology, Dec. 1995.
[25] S. R. Dooley and A. K. Nandi, “On explicit time delay estimation using the Farrow structure,”
Signal Processing, vol. 72, pp. 53−57, Jan. 1999.
[26] J. Vesma, “A frequency-domain approach to polynomial-based interpolation and the Farrow
structure,” to appear IEEE Trans. on Circuits and Systems II, March 2000.
[27] J. Vesma, Optimization and Applications of Polynomial-Based Interpolation Filters. Dr. Tech.
Thesis, Tampere, Finland: Tampere University of Tech., Department of Information Technology,
May 1999
[28] D. Babic, J. Vesma, T. Saramäki, M. Renfors, “Implementation of the transposed Farrow
structure,” in Proc. 2002 IEEE Int. Symp. Circuits and Systems, Scotsdale, Arizona, USA, 2002,
vol. 4, pp. 4−8.
[29] D. Babic, T. Saramäki and M. Renfors, “Conversion between arbitrary sampling frequencies using
polynomial-based interpolation filters,” in Proc. Int. Workshop on Spectral Methods and Multirate Signal Processing,
SMMSP’02, Toulouse, France, September 2002, pp. 57−64.
TLT-5806
Receiver Architectures
and Signal Processing
Vesa Lehtinen
Department of Communications Engineering
Tampere University of Technology
P.O.Box 553, 33101 Tampere, Finland
vesa.lehtinen@tut.fi
TLT-5806 Receiver Architectures and Signal Processing: Newton Interpolation for Fractional-Delay Filtering
–D+1 –D+N –1
------------------ -----------------------------
x[n] –D 2 N
1 – z –1 1 – z –1 1 – z –1
Disadvantages:
• Poor response (Lagrange)
– Requires an oversampled signal
• High order (due to poor response)
=> Longer delay in causal realisations
References
[1] L. Elden, L. Wittmeyer-Koch, and H.B. Nielsen, Introduction to Numerical Com-
putation. Studentlitteratur, Lund, 2004, pp. 107–113.
[2] http://mathworld.wolfram.com/UmbralCalculus.html
[3] S. Tassart and Ph. Depalle, “Fractional delays using Lagrange interpolators,“ in
Proc. Nordic Acoustic Meeting, Helsinki, Finland, 12–14 June, 1996.
[4] C. Candan, “An efficient filt ing structure for Lagrange interpolation,” in IEEE
Signal Processing Letters, Vol. 14, No. 1, Jan 2007, pp. 17–19.
[5] T.J. Goodman, M.F. Aburdene, “Interpolation Using the Discrete Pascal Trans-
form,” Proc. 40th Annual Conf. Information Sciences and Systems, 22–24
March 2006, pp. 1079–1083.
[6] Vesa Lehtinen, Markku Renfors, "Structures for Interpolation, Decimation, and
Nonuniform Sampling Based on Newton's Interpolation Formula," in Proc.
Sampling Theory and Applications (SAMPTA), Marseille, France, 18-22 May
2009. http://hal.archives-ouvertes.fr/hal-00451769/