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IEEE SIGNAL PROCESSING LETTERS, VOL. 22, NO.

8, AUGUST 2015

1021

Polyphase Decomposition of Digital


Fractional-Delay Filters
Hkan Johansson, Senior Member, IEEE, and fred harris, Fellow, IEEE

AbstractThis letter shows that, for arbitrary


-fold
polyphase decompositions, the polyphase components of digital
fractional-delay (FD) filters correspond to FD filters as well, but
with different delays and gains. For even values of
, the components also have additional and different phase offsets. The letter
also discusses the application of these results to the optimization
of high-order filters with few unknowns.
Index TermsFIR filters, fractional-delay filters, high-order filters, polyphase decomposition.

II. POLYPHASE DECOMPOSITION OF FD FILTERS


We consider linear filters represented with a real-valued impulse response
. The transfer function,
, is then
(1)
The

-fold polyphase decomposition of

is [1], [2]
(2)

I. INTRODUCTION

T IS well known that the polyphase components [1], [2]


of th-band lowpass digital filters (
), correspond to fractional-delay (FD) filters with distinct FD values
[3], [4]. This can, for example, be utilized when deriving efficient polyphase structures for -fold interpolation and decimation [5]. By implementing the polyphase components using the
Farrow structure [6], with a set of distinct FD values, they
share common parts which results in a low computational complexity. FD filters also play an important role in various other
applications like resampling with arbitrary conversion factors
(corresponding to a variable FD filter [7][9]), delay estimation,
and signal reconstruction [10][14]. It is thus of importance to
study the properties, design, and implementation of such filters.
This letter studies polyphase decomposition of FD filters
which has not been considered before. Section II will show
that, for arbitrary -fold polyphase decompositions ( being
an integer), the polyphase components correspond to FD filters
as well, but with different delays and gains. For even values of
, the components also have additional and different phase
offsets. In Section III, an application of these results will be
outlined, namely the optimization of high-order filters with
few unknowns, which are required in, e.g., filter banks and
transmultiplexers with many channels [4]. An example of a
high-order variable FD filter, and the corresponding th-band
filters, is also given. Finally, Section IV concludes the letter.

where the polyphase components,

, are given by
(3)

We are now interested in the polyphase components frequency responses,


, when the overall filter corresponds
to an FD filter with an FD of . That is, the overall frequency
response is ideally1
(4)
To derive an expression for
, given (4), we will make
use of the relation between the Fourier transforms of a continuous-time (CT) signal and its discrete-time (DT) counterpart.
Specifically, given a CT signal
and the DT signal (se, we
quence) obtained through the sampling
have [15], [16]
(5)
For simplicity, but without loss of generality, we will assume
hereafter that
. This means that
(6)
and

Manuscript received August 07, 2014; revised November 18, 2014; accepted
December 10, 2014. Date of publication December 18, 2014; date of current
version December 23, 2014. The associate editor coordinating the review of this
manuscript and approving it for publication was Prof. Mathew Magimai Doss.
H. Johansson is with the Divison of Communication Systems, Department
of Electrical Engineering, Linkping University, Linkping, Sweden (e-mail:
hakanj@isy.liu.se).
F. Harris is with the Department of Electrical and Computer Engineering, San
Diego State University, San Diego CA 92182 USA (e-mail: fharris@mail.sdsu.
edu).
Digital Object Identifier 10.1109/LSP.2014.2381603

(7)
corresponds to an
Now, since the forming of
-fold downsampling (due to
), and a multiplication of
by
(due to
), we have
1In a real-time implementation of a filter approximating (4), there will be an
additional (typically integer) delay to account for causality, but it is left out as
it does not affect the basic principles dealt with in this letter.

1070-9908 2014 IEEE. Personal use is permitted, but republication/redistribution requires IEEE permission.
See http://www.ieee.org/publications_standards/publications/rights/index.html for more information.

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IEEE SIGNAL PROCESSING LETTERS, VOL. 22, NO. 8, AUGUST 2015

VALUES OF

AND

B. Even Values of

TABLE I
IN EXAMPLE 1 WHERE

For even values of , the terms in (8) that fall into the band
are those for
, whereas the
are those for
terms that fall into the band
. In these two bands, respectively, we will now get
complex constants, say
, and their conjugates, instead of the
above. In the band
, we get
real constants

(8)
where
. Before proceeding, it is noted that,
since
represent real DT filters,
are con, with
jugate symmetric, implying that
denoting complex conjugate. We now distinguish between odd
and even values of .
A. Odd Values of
Due to the bandlimitation of (7), it is established that the
that fall into the band
only terms in (8) for odd values of
are those for
.
Utilizing
, and inserting (6) into (8), we then get

(13)
, the summation index ranges instead
In the band
from
to
. This corresponds to complex conjugation in (13). Thus, with
, we have shown that,
for even ,
(14)
where
(15)
with

(9)

and
being, again, given by (11) and (12), and with
denoting the sign of . Hence, the only difference beis that, for even values, we
tween even and odd values of
have the additional phase offsets
as given by (15).
C. Verification of the Results

which can be written as


(10)
with
(11)
and2

(12)
This shows that, for odd ,
are FD filters with the
FD values
given by (11), weighted with the real gain constants
given by (12). Note that
and
can take on both
positive and negative values depending on and .
2The last two equalities in (12) hold when
is not an integer
(thereby avoiding division by zero). This is ensured for all noninteger values
of which is the basic assumption as we deal with FD filters. If would be an
integer, it is obvious that one of the polyphase components would be unity and
the remaining ones zero.

Example 1: To verify the results derived above, we consider


and
using a 48th-order FD finite-length imand the
pulse response (FIR) filter, targeting an FD of
band
. The filter is here designed in the minimax sense, minimizing the modulus of the complex error, i.e.,
the difference between the actual filter frequency response and
the ideal response in (4). Table I gives the values of
and
whereas Figs. 1 and 2 plot the frequency responses for the
, as well as the ideal curves depolyphase components
termined by
and
, for
and
, respectively.
It is seen in Figs. 1 and 2 that the filter responses closely
follow the ideal curves, except in the neighborhood of
(
) for even (odd) values of . This is because
in (14) has phase offsets for even values of , which cannot
with real filters. On the other
be well approximated near
hand, for even
and
, the ideal
in (14) is
real-valued (due to the additional non-zero
) which means
with
that the ideal curves can be approximated well near
a real filter. In contrast, for odd values of , since the ideal
in (10) is complex-valued for
, as it lacks
(or, equivalently,
), a real filter cannot approximate the
ideal component well near
. However, since the ideal
does not have additional phase offsets for odd , they
.
can be approximated well near
Finally, it is noted that, despite of the features of the individual polyphase components, the overall real FD filter can always be designed to approximate the ideal response
as

JOHANSSON AND HARRIS: POLYPHASE DECOMPOSITION OF DIGITAL FRACTIONAL-DELAY FILTERS

1023

rameters) have to be determined in the design. For example, a


regular direct-form finite-length impulse response (FIR) filter
has
filter coefficient multipliers. To reduce
of order
the design complexity, it is therefore of interest to parametrize
the impulse response so that the number of free parameters is
lowered. For filters required in resampling, including regular
interpolation and decimation, as well as multirate filter banks,
one attractive way to accomplish this is to make use of the so
called Farrow structure [6]. This structure is generally used for
realizing a variable FD filter or, in the specific cases mentioned
above, a set of FD filters with fixed and distinct FD values,
which are then used as the polyphase components of th-band
filters [4], [5], [18]. Below, we will outline and demonstrate that
the results derived in Section II can be utilized to reduce the
number of free parameters even further.
Fig. 1. Gain (
ample 1 for

) and phase offset (


.

) of the polyphase components in Ex-

A. Variable FD Filters
The Farrow-structure transfer function is expressible as3
(16)

Fig. 2. Gain (
ample 1 for

) and phase offset (


.

) of the polyphase components in Ex-

, depends on the desired


where the number of terms,
approximation error and bandwidth [19]. When the subfilters
are designed appropriately,
approximate the
differentiators4
, and the overall filter frequency
response
approximates an FD filter with the ideal
,
,
. In the filter design,
response
the free parameters are the impulse response values
, and
they are determined in such a way that
meets a prescribed specification for all values of in a sampling interval,
i.e., for all
[19]. In the implementation,
after the filters have been designed,
are fixed whereas
is a variable parameter that determines the FD.
We also know from Section II that the polyphase components
of an -fold polyphase decomposed FD filter are also FD filters, but with different delays and gains (and phase offsets for
even ). For odd values of , this directly gives us a new FD
filter transfer function form as

(17)

Fig. 3. Upper: Magnitude response of the approximate 88th-band filter in Example 3. Lower: Approximation error modulus of the corresponding polyphase
components (FD filters) in Example 2.

well as desired for


[17].

, but not at

III. APPLICATION TO THE DESIGN OF HIGH-ORDER FILTERS


Filters with stringent requirements may require (very) high
orders which in turn means that many free filter coefficients (pa-

approximate the FD filter responses


,
where
and
are given by (11) and (12), respectively.
and where
Again, the only free design parameters are the impulse response
values
. However, since
correspond to sparse im), the order of
, say
, can be
pulse responses (due to
much lower in (17) than in (16), as will be demonstrated in Example 2 in the next subsection. Moreover, the overall filter order
is
. Similar to other filter design tech3The notation
is used to indicate that the transfer characteristics depend on the delay parameter .
4In an optimized regular variable FD filter, in the form of (16),
obtained via Taylor series expansion, but one can still observe that
approximate the terms
[20].

are not

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IEEE SIGNAL PROCESSING LETTERS, VOL. 22, NO. 8, AUGUST 2015

TABLE II
IMPULSE RESPONSE VALUES OF THE SUBFILTERS

niques that make use of periodic subfilters [9], [21][25],


approximately equals the filter order of the corresponding regshould
ular filter, here the filter of (16). This indicates how
be selected in (17), given
and the filter order required in the
regular filter of (16). Further, there is an excess filter order in
of (17), given by the term
. Hence, there is a trade-off
between a reduction of the number of parameters (reduction of
corresponding to an increased ) and an increased filter
.
order
The transfer function form in (17) is in fact also valid for
even values of , which can be explained as follows. Assume
in (17) approximate
(as above for
that
odd
). Then, replace
with
which corresponds
to replacing with
, which in turn means that the
approxicorresponding polyphase components
mate
. The terms
are precisely those additional terms present in the
polyphase components for even values of , as seen in (14).
Finally, since
and
are also functions
of
, we see that the form of (17) is valid for even values of
as well. The difference is that, for even ,
approximate
, as opposed to
in the case of odd 5.

B. Design Examples
Example 2: We consider the design of a variable FD filter
with a specification comparable to that of Example 1 of [9],
viz., a 90% bandwidth (
),
, an approxidB, and the FD covering one sammation error of some
pling interval. Here, we meet the specification using (17) with
and second-order FIR subfilters
with symmetric
(anti-symmetric) impulse responses for even (odd) values of
, as seen in Table II. The filter is designed in the minimax
sense, as in Example 1, but over all possible values of
, and essentially in the same way as outlined for Specification I in [19], but using (17) instead of (16)6.
Fig. 3 (lower) plots the approximation error modulus for the FD
values used in a 88th-band filter, to be considered in Example 3.
There are only 10 free impulse response values (parameters)
in our design, due to the impulse response symmetries and the
zero-valued taps of anti-symmetric even-order filters. In contrast, as seen in Table I in [9], the regular Farrow structure has
5When realizing the filter in the form of (17),
tend to depart from
approximating their ideal responses (obtained via Taylor series expansion), esin which case
can be of very low order
pecially for larger values of
as seen in Example 2. This is different from the regular form of (16), as noted
in Footnote 4.
6Since

the order of
even, we have here used

in (17) is

, and
.

is

IN

EXAMPLES 2 AND 3

191 free parameters. In [9], this number was reduced by combining the Farrow structure and a so called two-rate approach
which utilizes efficient half-band filters and low-order Farrow
subfilters. Together with the frequency response masking approach [21], [24], [25], the number was reduced to 71, which
still by far exceeds 10. As mentioned in Section III-A, a price to
pay using the form of (17) is an increased filter order and thus
an increase of the additional integer part of the overall delay for
the corresponding causal filter. Here, the integer-delay part is
53, which should be compared with 36 and 43, respectively, for
the two above mentioned methods.
Example 3: We now use
instances of the FD filter in
Example 2, with distinct FD values [3], as the polyphase
components of an approximate th-band filter7, to obtain
high-order frequency selective filters via few-parameter optimization. Here, we follow the polyphase decomposition in [26]
for even
to obtain a linear-phase lowpass filter, and with
the components being realized in accordance to [5] but using
(17) instead of (16). Fig. 3 (upper) shows the resulting lowpass
filter frequency response in the case of
. The overall
filter order is 9504 and the passband and stopband edges are
at
and
, respectively,
whereas the passband and stopband ripples are
and
, respectively.
The filter has the same stopband edge as the two filters compared in Example 5 of [4], but those filters meet a considerably
less stringent specification (passband edge at
, and passband and stopband ripples of 0.01 and 0.001, respectively). In
addition, those filters required more filter parameters to be determined in the design, namely 24 using the filters proposed in [4]
and 32 using the filters introduced in [27]. This should be compared with the 10 filter parameters that need to be determined
in our design. Again, this shows that we can design high-order
filters with very stringent requirements and yet with very few
free filter design parameters.
IV. CONCLUSION
This letter showed that the polyphase components of FD filters, for arbitrary -fold polyphase decompositions, are FD filters as well, but with different delays and gains. For even values
of , the components also have additional and different phase
offsets. The letter also discussed how these results can be used
in the optimization of high-order filters with few unknowns. In
particular, high-order variable FD filters and the corresponding
approximate th-band filters were discussed and exemplified
) through designs with only 10 unknown parame(for
ters in the optimization.
7The filter in Example 3 is an approximate th-band filter as the pure-delay
polyphase component of an th-band filter is approximated by an odd-order
linear-phase FIR filter.

JOHANSSON AND HARRIS: POLYPHASE DECOMPOSITION OF DIGITAL FRACTIONAL-DELAY FILTERS

REFERENCES
[1] P. P. Vaidyanathan, Multirate Systems and Filter Banks. Upper
Saddle River, NJ, USA: Prentice Hall, 1993.
[2] f. j. harris, Multirate Signal Processing for Communication Systems.
Upper Saddle River, NJ, USA: Prentice-Hall, 2004.
[3] M. G. Bellanger, G. Bonnerot, and M. Coudreuse, Digital filtering
by polyphase network: Application to sample-rate alteration and filter
banks, IEEE Trans. Acoust., Speech, Signal Process., vol. ASSP-24,
no. 2, pp. 109114, Apr. 1976.
[4] A. Eghbali and H. Johansson, On efficient design of high-order filters with applications to filter banks and transmultiplexers with large
number of channels, IEEE Trans. Signal Process., vol. 62, no. 6, pp.
11981209, Mar. 2014.
[5] H. Johansson and O. Gustafsson, Linear-phase FIR interpolation, decth-band filters utilizing the Farrow structure, IEEE
imation, and
Trans. Circuits Syst. I, vol. 52, no. 10, pp. 21972207, Oct. 2005.
[6] C. W. Farrow, A continuously variable delay element, in Proc. IEEE
Int. Symp., Circuits, Syst, Espoo, Finland, Jun. 79, 1988, vol. 3, pp.
26412645.
[7] D. Babic and M. Renfors, Power efficient structure for conversion
between arbitrary sampling rates, IEEE Signal Process. Lett., vol. 12,
no. 1, pp. 14, Jan. 2005.
[8] J. Selva, An efficient structure for the design of variable fractional
delay filters based on the windowing method, IEEE Trans. Signal
Process., vol. 56, no. 8, pp. 37703775, Aug. 2008.
[9] H. Johansson and E. Hermanowicz, Two-rate based low-complexity
variable fractional-delay FIR filter structures, IEEE Trans. Circuits
Syst. I, vol. 60, no. 1, pp. 136149, Jan. 2013.
[10] T. I. Laakso, V. Vlimki, M. Karjalainen, and U. K. Laine, Splitting
the unit delaytools for fractional delay filter design, Signal Process.
Mag., vol. 13, no. 1, pp. 3060, Jan. 1996.
[11] S. R. Dooley and A. K. Nandi, On explicit time delay estimation using
the farrow structure, Signal Process., vol. 72, pp. 5357, Jan. 1999.
[12] H. Johansson and P. Lwenborg, Reconstruction of nonuniformly
sampled bandlimited signals by means of digital fractional delay filters, IEEE Trans. Signal Processing, vol. 50, no. 11, pp. 27572767,
Nov. 2002.
[13] M. Olsson, H. Johansson, and P. Lwenborg, Time-delay estimation
using Farrow-based fractional-delay FIR filters: Filter approximation
vs. estimation errors, in Conf. Proc. XIV Eur. Signal Processing, Florence, Italy, Sep. 48, 2006.

1025

[14] S. Tertinek and C. Vogel, Reconstruction of nonuniformly sampled


bandlimited signals using a differentiator-multiplier cascade, IEEE
Trans. Circuits Syst. I, vol. 55, no. 8, pp. 22732286, Sep. 2008.
[15] A. V. Oppenheim and R. W. Schafer, Discrete-Time Signal Processing. Upper Saddle River, NJ, USA: Prentice Hall, 1989.
[16] H. Johansson, Sampling and quantization, Signal Process. Theory
Mach. Learn., vol. 1, pp. 169244, 2013.
[17] G. Cain, N. Murphy, and A. Tarczynski, Evaluation of several variable
FIR fractional-sample delay filters, in Proc. IEEE Int. Conf. Acoust.,
Speech, Signal Processing, Adelaide, Australia, Apr. 1922, 1994, vol.
3, pp. 621624.
[18] H. Johansson, Farrow-structure-based reconfigurable bandpass linearphase FIR filters for integer sampling rate conversion, IEEE Trans.
Circuits Syst. II, vol. 58, no. 1, pp. 4650, Jan. 2011.
[19] H. Johansson and P. Lwenborg, On the design of adjustable fractional delay FIR filters, IEEE Trans. Circuits Syst. II, vol. 50, no. 4,
pp. 164169, Apr. 2003.
[20] M. Abbas, O. Gustafsson, and H. Johansson, On the fixed-point implementation of fractional-delay filters based on the Farrow structure,
IEEE Trans. Circuits, Syst. I, vol. 60, no. 4, pp. 926937, Apr. 2013.
[21] Y. C. Lim, Frequency-response masking approach for the synthesis
of sharp linear phase digital filters, IEEE Trans. Circuits, Syst., vol.
CAS-33, no. 4, pp. 357364, Apr. 1986.
[22] T. Saramki, T. Neuvo, and S. K. Mitra, Design of computationally
efficient interpolated FIR filters, IEEE Trans. Circuits Syst., vol. 35,
no. 1, pp. 7088, Jan. 1988.
[23] T. Saramki, Finite impulse response filter design, in Handbook for
Digital Signal Processing, S. Mitra and J. Kaiser, Eds. New York,
NY, USA: Wiley, 1993, ch. 4, pp. 155277.
[24] T. Saramki, Y. C. Lim, and R. Yang, The synthesis of half-band filter
using frequency-response masking technique, IEEE Trans. Circuits
Syst. II, vol. 42, no. 1, pp. 5860, Jan. 1995.
[25] H. Johansson, Two classes of frequency-response masking
linear-phase FIR filters for interpolation and decimation, Circuits,
Syst., Signal Process., vol. 25, no. 2, pp. 175200, Apr. 2006.
[26] H. Johansson and A. Eghbali, Add-equalize structures for linear-phase
Nyquist FIR filter interpolators and decimators, IEEE Trans. Circuits
Syst. I, vol. 61, no. 6, pp. 17661777, Jun. 2014.
[27] R. Lehto, T. Saramki, and O. Vainio, Synthesis of narrowband
linear-phase FIR filters with a piecewise-polynomial impulse response, IEEE Trans. Circuits Syst. I, vol. 54, no. 10, pp. 22622276,
Oct. 2007.

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