486 IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMSII: EXPRESS BRIEFS, VOL. 62, NO.
5, MAY 2015
Two-Stage-Based Polyphase Structures for
Arbitrary-Integer Sampling Rate Conversion
Hkan Johansson, Senior Member, IEEE, and Heinz Gckler
AbstractThis brief derives efficient two-stage-based polyphase
structures for arbitrary-integer sampling rate conversion. For
even-integer conversions, the overall structures correspond to par-
allelized conventional two-stage structures, but the derivations in
this brief offer further insights when comparing the two cases of
odd- and even-integer conversions. For the class of linear-phase
finite-length impulse response M th-band filters, it is then demon-
strated through design examples that conversions by odd factors
are in fact more efficient than by even factors.
Index TermsDecimation, interpolation, M th-band filters,
multistage structures, Nyquist filters, polyphase structures, sam-
pling rate conversion.
I. I NTRODUCTION
I NTERPOLATION and decimation (sampling rate conver-
sions) are fundamental signal processing operations required
in numerous applications [1][7]. This brief considers sampling
rate conversion by an arbitrary integer factor, denoted by M .
Such conversions can be used either alone or as parts of
noninteger conversions. Since a decimator structure is obtained
by transposing the corresponding interpolator structure [1], this
brief will henceforth focus on interpolation.
From the circuit and systems point of view, the objective is
to carry out the interpolation in a computationally efficient way
to end up with efficient implementations [4][8]. When M can
be factored as a product of K integers, it is well known that the
K-stage interpolator is more computationally efficient than the
one-stage counterpart [8], [9]. This brief focuses on the two-
stage case, because the largest savings are typically obtained
when going from one to two stages [8]. Two-stage converters
can be efficiently used alone for small to medium values of M
or as efficient parts of multistage converters for larger values of
M . However, when M is a prime number, one can obviously
not make use of integer-conversion stages. Instead, at least one
of the stages must convert by a rational factor [9], [10]. The use
of multistage converters with rational-conversion stages is not
a new idea, but the focus of such converters has largely been
on the case where the overall conversion factor is a rational
number (for irrational numbers, one has to include other types
Manuscript received August 12, 2014; revised October 14, 2014; accepted
December 28, 2014. Date of publication January 9, 2015; date of current
version April 23, 2015. This brief was recommended by Associate Editor
G. J. Dolecek.
H. Johansson is with the Division of Communication Systems, Department
of Electrical Engineering, Linkping University, 581 83 Linkping, Sweden
(e-mail: hakanj@isy.liu.se).
H. Gckler is with Ruhr-Universitt Bochum, 44780 Bochum, Germany,
and also with Karlsruhe Institute of Technology, 76133 Karlsruhe, Germany
(e-mail: heinz.goeckler@ruhr-uni-bochum.de).
Digital Object Identifier 10.1109/TCSII.2015.2389291
1549-7747 2015 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.
Fig. 1. (a) Interpolator. (b) Polyphase structure.
of conversion structures [11]). Here, we are interested in using
a rational-conversion stage when the overall conversion factor
is an integer. Specifically, the contribution and outline of this
brief is as follows.
After recapitulating polyphase structures in Section II,
Section III derives efficient two-stage-based polyphase struc-
tures for arbitrary-integer values of M .1 The basic principle
is to first interpolate by two, then interpolate by M , and
finally downsample by two. It will be shown that the overall
interpolator corresponds to a regular interpolator, i.e., without
aliasing that seemingly appears due to the downsampling. The
basic principle was considered in [10], but that brief did not
consider the structures to be derived here, and it did not take
the aliasing-free property into account in the filter design, which
results in higher filter orders and complexities than necessary.
Section III also considers the particularly efficient class of
M th-band (Nyquist) filters [8], [12]. Then, Section IV con-
siders the design of linear-phase finite-length impulse response
(FIR)M th-band filters for interpolation by M . Through design
examples, it will be shown that interpolation by odd factors is
in fact more efficient than by even factors for this filter class.
Finally, Section V concludes this brief.
II. P OLYPHASE S TRUCTURES
As depicted in Fig. 1(a), interpolation by an integer factor
of M can be described as upsampling-by-M followed by an
M th-band filter (or approximate M th-band filter) represented
by the transfer function H(z),2 which is the z-transform of the
filters impulse response h(n). In practice, the interpolation is
carried out using the corresponding polyphase structure shown
in Fig. 1(b). This structure is derived from the polyphase
representation of H(z), given by

M 1
H(z) = z k Hk (z M ) (1)
k=0
1 For even values of M , the overall structures can alternatively be obtained
from the conventional two-stage integer-conversion structures by parallelizing
the polyphase branches in the second stage, but our derivations offer further
insights when comparing the two cases of odd and even values of M .
2 For simplicity, we will frequently write filter H(z) and understand that it
means a filter with the transfer function H(z).
JOHANSSON AND GCKLER: POLYPHASE STRUCTURES FOR ARBITRARY-INTEGER SAMPLING RATE CONVERSION 487

the so obtained result. Two cases are distinguished below,

depending on whether M is odd or even. Before proceeding
with those details, it is noted that the overall filter order is
NH = (M NE + NF )/2 because E(z M )F (z) is sandwiched
between upsampling and downsampling by two.

Fig. 2. Equivalent structures. A. Structures for Odd Values of M

For odd values of M , one first obtains
where Hk (z) denote the polyphase components transfer func-

M 1
tions. All filtering in the polyphase interpolator structure is
carried out at the input sampling rate, and the output sequence H(z) = E0 (z M ) z m F2m (z M )
is, in practice, formed via a commutator that interleaves the m=0
polyphase filters output samples. 
M 1
+ z (M +1)/2 E1 (z M ) z m F2m+1 (z M ). (6)
III. T WO -S TAGE -BASED P OLYPHASE S TRUCTURES m=0

This section derives the proposed structures3 utilizing the
equivalences shown in Fig. 2. The point of departure is that
interpolation by an integer factor of M is to be performed. Here,
this is principally carried out, as shown in Fig. 2(a). The first
step is to interpolate by two through the twofold upsampler and
the filter E(z). Next, interpolation by M follows via the M -fold
upsampler and the filter F (z). Finally, twofold downsampling
takes place, which means that the overall scheme interpolates
by M .
Using a noble identity [1], and the fact that cascaded upsam-
plers obviously can be interchanged, the equivalent structure in
Fig. 2(b) follows. Then, we utilize that a scheme comprising
a filter sandwiched between upsampling and downsampling
by two corresponds to a linear and time-invariant (LTI) sys-
tem with a transfer function that equals the zeroth polyphase
component of the filter [1]. Utilizing this fact, the equivalent
structure in Fig. 2(c) follows, where:
A(z 1/2 ) + A(z 1/2 )
H(z) = = A0 (z)
2 halved.5 To elaborate on this point in more detail, we assume
with
A(z) = A0 (z 2 ) + z 1 A1 (z 2 ) = E(z M )F (z).
The transfer function H(z) = A0 (z) is found by utilizing the
twofold and 2M -fold polyphase decompositions of E(z) and
F (z), respectively, i.e.,
Comparing this expression with the polyphase decomposition
in (1), one finds that the polyphase components are given by
Hk (z) = E0 (z)F2k (z) + z (1+sk )/2 E1 (z)F2k+sk M (z) (7)
where

1, k = 0, 1, . . . , k0
sk = (8)
1, k = k0 + 1, k0 + 2, . . . , M 1
with k0 = (M 1)/2. It is noted that both E0 (z) and E1 (z)
are symmetric subfilters4 when E(z) is a symmetric filter. As
to Fm (z), only F0 (z) and FM (z) are symmetric subfilters
when F (z) is a symmetric filter. However, it is known that
each unsymmetric polyphase component has a correspond-
ing polyphase component with a reversed impulse response
[14][16]. This means that one can realize these two compo-
nents as a sum and difference of symmetric and antisymmetric
(2) subfilters, which implies that the multiplication complexity is
hereafter that the overall filter is noncausal. The corresponding
causal filter is readily obtained through an appropriate time
(3) shift (delay), which merely corresponds to a renumbering of
the polyphase components.
For a noncausal filter, with a 2M -fold polyphase decompo-
sition according to (5), the polyphase components fm (n) and
f2M m (n), for m = 1, 2, . . . , M 1, satisfy

E(z) = E0 (z 2 ) + z 1 E1 (z 2 ) (4) fm (n) = f2M m (n), n =

NFm NFm
, + 1, . . . ,
NFm
2 2 2

2M 1 (9)
F (z) = z m Fm (z 2M ). (5)
where NFm denotes the order of Fm (z) and F2M m (z). There-
m=0
fore, these M 1 pairs of filters can be efficiently realized as
It then follows that H(z) is obtained by keeping the even- the sum and difference of M 1 pairs of linear-phase filters,
indexed terms (i.e., the terms corresponding to z m for even e.g., Bm (z) and Cm (z), given by
m) in the 2M -fold polyphase representation of E(z M )F (z)
when utilizing (4) and (5) and then replacing z 2 with z in Bm (z) = [Fm (z) + F2M m (z)] /2
Cm (z) = [Fm (z) F2M m (z)] /2. (10)
3 The basic structures, expressible in terms of the polyphase components
in (7), can be used for general FIR filters as well as infinite-length impulse
4 A symmetric filter H(z) [antisymmetric filter H(z)] means that the
response (IIR) filters when E(z) and F (z) are expressed in the polyphase
forms of (4) and (5), like in M th-band IIR filters [13]. This brief focuses filters impulse response is symmetric [antisymmetric], i.e., h(n) = h(n)
on linear-phase FIR filters, which is why we will take (7) a step further and [h(n) = h(n)] for a noncausal filter.
present structures expressed in terms of the polyphase components in (12). 5 Alternative realizations exist (derived via transposition) that achieve the
These expressions enable the use of symmetric and antisymmetric subfilters same multiplication savings, but they require substantially more delay elements
that reduce the multiplication complexity in the implementation. [9], [14], [17].
488 IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMSII: EXPRESS BRIEFS, VOL. 62, NO. 5, MAY 2015
Fig. 3. Polyphase interpolator structure for M = 5.
The filters Bm (z) are then symmetric filters, whereas Cm (z)
are antisymmetric filters. The filters Fm (z) and F2M m (z) are
then realized as
Fm (z) = Bm (z) + Cm (z)
F2M m (z) = Bm (z) Cm (z). (11)
Making use of the preceding equations, the polyphase com-
ponents Hk (z) in (7), for k = 1, 2, . . . , M 1, can be rewrit-
ten as in (12), shown at the bottom of the page with k1 =
k0 + 1. It is seen that the same even-indexed components of
Bm (z) and Cm (z) are used when forming E0 (z)[B2k (z) +
C2k (z)] and E0 (z)[B2M 2k (z) C2M 2k (z)]. Likewise, the
same odd-indexed components of Bm (z) and Cm (z) are
used when forming z 1 E1 (z)[BM 2k (z) CM 2k (z)] and
z 1 E1 (z)[B2kM (z) + C2kM (z)]. This means that only one
copy is needed for all subfilters in the final realization. To
illustrate this, the structure for M = 5 is shown in Fig. 3.
B. Structures for Even Values of M
For even values of M , one first obtains
 M
 1
M/2
M
H(z) = E0 (z ) + z M
E1 (z ) z m F2m (z M ).
m=0
(13)
Comparing this expression with the polyphase decomposition
in (1), one finds that the polyphase components Hk (z) can
again be written as in (7), but here with k0 = (M 2)/2 in
(8). Further, utilizing Bm (z) and Cm (z), as defined by (10),
one obtains (12) again, but now with k1 = k0 + 2. We thus
obtain similar polyphase component expressions for both odd
E0 (z)F0 (z) + z 1 E1 (z)FM (z),
E0 (z) [B2k (z) + C2k (z)] + z 1 E1 (z) [BM 2k (z) CM 2k (z)] ,
Hk (z) =
E0 (z)FM (z) + E1 (z)F0 (z),
E0 (z) [B2M 2k (z) C2M 2k (z)] + E1 (z) [B2kM (z) + C2kM (z)] ,
Fig. 4. Polyphase interpolator structure for M = 6.
and even values of M . However, a main difference between the
two cases is that, for even M , only the even-indexed polyphase
components F2k (z) are used. This is because, for even M ,
dowsampling by two follows an even-fold interpolation in
which case odd-indexed polyphase components are apparently
never used. Furthermore, in the final realization, two copies of
each subfilter emanating from F2k (z) are now required. This
is because only the even-indexed components F2k (z), and thus
also B2k (z), and C2k (z), are now used when forming all terms
in (12). As an illustration, the structure for M = 6 is shown in
Fig. 4. For even M , one may alternatively implement the two-
stage interpolator using a regular polyphase structure for each
stage. In that case, only one copy of the polyphase components
F2k (z) are needed, but then they operate at two times the input
rate, which implies the same number of arithmetic computa-
tions per time unit.
C. Structures for M th-Band Filters
A particularly efficient filter class for M -fold sampling rate
conversion is constituted by M th-band filters. For this filter
class, every M th impulse response value is zero, except for
the center tap, which is equal to unity in the interpolation case
(1/M in the decimation case). In the z-domain, this means that
H0 (z) = 1 in our noncausal-filter formulation. This is achieved
by letting E(z) be a half-band filter and F (z) be an M th-band
filter. In this case, E0 (z) = F0 (z) = 1 and FM (z) = 0 [due to
the 2M -fold polyphase decomposition of an M th-band filter,
given by (5)]. It then follows directly from (12) that H0 (z) =
1. Section IV will specifically consider the design of linear-
phase FIR M th-band filters. Alternatively, approximately
k =0
k = 1, 2, . . . , k0
(12)
k =M 2 (only for even M )
k = k1 , k1 + 1, . . . , M 1
JOHANSSON AND GCKLER: POLYPHASE STRUCTURES FOR ARBITRARY-INTEGER SAMPLING RATE CONVERSION 489

linear-phase FIR or IIR M th-band filters can be used to reduce TABLE I

R ESULTS OF THE D ESIGN E XAMPLES . T HE Q UANTITY J D ENOTES THE
the delay and/or complexity [7], [13]. N UMBER OF M ULTIPLICATIONS OF A LL S YMMETRIC AND
A NTISYMMETRIC S UBFILTERS , W HICH A RE S HOWN IN F IGS . 3 AND
4 FOR M = 5 AND M = 6, W HEREAS R D ENOTES THE M ULTIPLICATION
IV. L INEAR -P HASE FIR M TH -BAND F ILTERS R ATE . T HE N UMBERS OF A DDITIONS F OLLOW THE S AME T REND AND
A RE T HEREFORE N OT S HOWN . T HE C OLUMNS W ITH
This section considers the design of linear-phase FIR M th- S UBSCRIPT S C ORRESPOND TO S INGLE - STAGE D ESIGNS
band filters and provides design examples.

A. Design
In the design of M th-band filters, it normally suffices to
consider the stopband region [(1 + )/M, ], 0 < < 1.
This is because the passband and stopband ripples, e.g., c
and s , are related. For a linear-phase FIR M th-band filter,
with a unity passband gain,6 we have c (M 1)s [12]. To
find the lowest overall complexity for each M , we design the
corresponding overall filter in three steps as follows.
Step 1: Design a low-pass half-band filter E(z) with a
stopband edge at (1 + )/2 and a stopband ripple of s . This
gives an estimated subfilter order of E(z), e.g., N
E .
Step 2: Design a low-pass M th-band filter F (z) with a
stopband edge at (1 + )/M and a stopband ripple of s .7
This gives an estimated subfilter order of F (z), e.g., N
F .
Here, = 1 gives a rather pessimistic estimation N
F . We have
experimentally observed that 10 and 2.5 for odd and
even M , respectively, are appropriate.
Step 3: For all combinations of allowed even NE and NF
around N
E and N
F , design the overall M th-band filter H(z)
with a stopband edge at (1 + )/M and a stopband ripple of
s .8 Among all overall filters that meet the specification, select
the one with the lowest computational complexity (indicated by
J and R in Table I).
The design problems in Steps 1 and 2 are convex opti-
Fig. 5. Magnitude responses for M = 7. (Top) E(z M ) (solid) and F (z)
mization problems. The overall filter in Step 3 makes use of
(dashed). (Bottom) H(z) (solid), A(z 1/2 ) (dashed), and A(z 1/2 ) (dash-
cascaded subfilters, which gives rise to nonconvex optimization dotted).
problems.9 However, for linear-phase FIR filters, the objective
function is well behaved, and the optimization is rather insen-
sitive to the initial solution. As initial solutions, we use the
filters obtained via Steps 1 and 2 for each pair of values of NE
and NF . This guarantees a good final solution as the overall
optimization then only refines a solution that is close to the
final one. We design all filters in the minimax sense using the
function f minimax in MATLAB.10

B. Design Examples
We assume = 0.2 and s = 0.001. The results for M
between 3 and 20 are summarized in Table I, including single-

6 In the design, unity-gain passbands are assumed. In the interpolator im- Fig. 6. Magnitude responses for M = 8. (Top) E(z M ) (solid) and F (z)
plementations, the so obtained impulse responses e(n) and f (n) are then (dashed). (Bottom) H(z) = A(z 1/2 ) = A(z 1/2 ).
multiplied by two and M , respectively, to maintain the signal power.
7 As explained in Section III, and in the design example, odd-indexed impulse
stage designs to illustrate the efficiency of the two-stage de-
response values of F (z) are effectively not used for even M , and they are
therefore set to zero in the design. signs. For M = 7 and M = 8, Figs. 5 and 6, respectively,
8 In the examples, we use three and seven values of N
E and NF around
plot the magnitude responses of the filters (normalized by the
N
E and N
F , respectively, satisfying the following: NE = two times an even corresponding interpolation factors). Table I reveals that, as M
integer plus two (half-band filter), NF = even integer, and NH = 0.5(M NE + increases, the interpolation becomes relatively more compu-
NF ) = even integer. Consequently, around ten designs are required for each tationally efficient for odd values of M than for even values
M , and each design takes only a few seconds on a standard laptop.
9 Nonconvex optimization problems appear for all cascade-filter structures of M . This emanates from a higher order of F (z) for even
and is thus not specific for the proposed structures and design. M . This is because the odd-indexed polyphase components of
10 The function f minimax is an optimizer for nonconvex problems but F (z) are not used for even M , which implies that the effective
works well also for convex problems. filtering of F (z), at the higher sampling rate, corresponds
490 IEEE TRANSACTIONS ON CIRCUITS AND SYSTEMSII: EXPRESS BRIEFS, VOL. 62, NO. 5, MAY 2015
to a filter of the form P (z 2 ) and, thus, a -periodic filter.11
Consequently, A(ej/2 ) = A(ej/2 ), where A(z) is given
by (3). Then, since H(ej ) = (A(ej/2 ) + A(ej/2 ))/2, we
get H(ej ) = A(ej/2 ), which means that the requirements
on A(ej/2 ) coincide with those of H(ej ). This can also
be understood by recalling that the overall filter for even M
corresponds to two cascaded filters,12 which also explains the
overdesign in some frequency regions for even M , as shown
in Fig. 6. For odd M , on the other hand, the individual re-
quirements on A(ej/2 ) and A(ej/2 ) can be relaxed as these
terms are then different and can partly cancel each other out, as
shown in Fig. 5. This enables a reduction of the passband re-
quirements of F (z), which corresponds to a reduced passband
edge and a reduced filter order. The passband droop of F (z) is
compensated by E(z). Although this leads to a large stopband
ripple of E(z), due to the passband and stopband ripple relation
of M th-band filters, and thus a large stopband ripple of the
individual terms A(ej/2 ) and A(ej/2 ), the cancellation
between them enables the overall filter H(z) to still meet its
requirements. It is noted that the complexity difference between
even and odd M is obtained when E(z) is a half-band filter. The
difference disappears when E(z) is a general filter, which is
required if the overall filter cannot include /M in its transition
band. As the passband and stopband ripples are not related for
a general filter, one can then relax the requirements on F (z) for
all M .
To assess the coefficient sensitivity, we have rounded all
coefficients to 24 bits. Then, for both single- and two-stage
designs, the magnitude response deviations from the MATLAB
precision designs are below 1.1 103 dB for all M . This
indicates that the single- and two-stage structures have similar
sensitivity properties.
V. C ONCLUDING R EMARKS
This brief has derived efficient two-stage-based polyphase
structures for arbitrary-integer sampling rate conversions.13 For
the class of linear-phase FIR M th-band filters, it was demon-
strated that conversions by odd factors are then more efficient
than by even factors, in which case the structures can alterna-
tively be derived from regular two-stage structures. Thereby,
the proposed structures are also more efficient than the recently
introduced addequalize structures14 in [18], which can also
be used for arbitrary-integer conversion, but whose complexity
is comparable (but not lower) to that of regular multistage
converters (when such can be used) but not more efficient for
odd M than for even M . To exemplify, for M = 13, = 0.125,
11 The -periodicity of F (z) for even M explains the differences between
odd and even M regarding the passband gains of F (z) and A(z 1/2 ).
12 Instead of using two identical sets of subfilters at the output ports of E (z)
0
and E1 (z), emanating from F (z), one may consider using two different sets.
One could conjecture that one thereby could overcome the overhead for even
M , as the structure then no longer corresponds to two cascaded filters, which
gives more free filter parameters for the same implementation complexity.
However, we have experimentally observed that this generalized structure also
has similar properties and offers only modest (if any) subfilter order reductions.
In the design examples, at most two multiplications were saved.
13 As always when using two-stage-based linear-phase converters, a price to
pay for the complexity reductions is a somewhat increased delay as compared
with the single-stage counterparts [8] (4%29% increase in the examples).
14 Unlike the two-stage-based structures in this brief, the structures in [18]
are derived by iteratively adding and equalizing the output signals of fractional-
delay filters.
and s = 0.01 [18, Example 2]), we obtain NE = 34 and NF =
40 using the proposed structure. This gives a multiplication rate
of 2.15 compared to 2.38 for the structure in [18] and 4.92 for
the single-stage structure.
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