Circuit for sample rate conversion

Information

  • Patent Grant
  • 7724162
  • Patent Number
    7,724,162
  • Date Filed
    Friday, July 25, 2008
    16 years ago
  • Date Issued
    Tuesday, May 25, 2010
    14 years ago
Abstract
The present invention is related to a circuit for converting the sample rate of a digital signal, comprising an input for applying the digital signal,a conversion filter having either a symmetrical or anti-symmetrical impulse response and implemented as a plurality of subfilters in parallel, each subfilter having a symmetrical or anti-symmetrical response derived from components of a polyphase decomposition of said impulse response,combining means for deriving from said applied digital signal input signals of said plurality of subfilters or for combining output signals of said plurality of subfilters into a digital signal with converted sample rate,an output for outputting said digital signal with converted sample rate.
Description

This application claims priority from Great Britain Patent Application, No. GB 0716942.8 filed on 31 Aug. 2007, which is incorporated by reference in its entirety.


FIELD

The present invention relates to the field of digital integer factor sample rate conversion (i.e. decimation and interpolation).


BACKGROUND

Digital sample rate conversion (SRC) is a technique to replace a system that includes a variety of master clock rates with a fixed clock rate system. Applications are abundant, e.g. acquisition (signal generator) systems with an oversampled ADC (DAC), followed (preceded) by a digital decimator (interpolator) to relieve the analog anti-aliasing (reconstruction) filter specifications. Integer factor SRC also appears in analysis (i.e. decimation) and synthesis (i.e. interpolation) filter banks.


To explain the problem tackled in the present invention a linear phase digital filter is assumed of order N, implying a finite impulse response h(k) of length N+1 (Eq. 1) with symmetric (Eq. 2) or anti-symmetric (Eq. 3) impulse response.










y


[
n
]


=




k
=
0

N








h


[
k
]




x


[

n
-
k

]








(

Eq
.




1

)







h


[
k
]


=



h


[

N
-
k

]







with





k



0











N






(

Eq
.




2

)







h


[
k
]


=



-

h


[

N
-
k

]








with





k



0











N






(

Eq
.




3

)








Tƒis defined as the sample period of the output signal (for an interpolator) or the input signal (for a decimator). Thus, Tƒis the sample period of the signal with the fastest sample rate. For a single-rate filter Tƒis equal to the sample period of both input and output signals.


#M is defined as the number of multiplications per output sample and per sample period Tƒ. A single rate realisation of Eq. 1 results in #M as given in Eq. 4. Also naive realisations of Eq. 1 in integer factor interpolators and decimators have that same #M as shown in Eq. 4.

#M=N+1  (Eq. 4)

In a single-rate system a linear phase FIR realisation can take advantage of its impulse response symmetry (Eq. 5) or anti-symmetry (Eq. 6); whereby └ ┘ denotes a truncation to the lower integer.











y


[
n
]


=




k
=
0




N
2











h


[
k
]




(


x


[

n
-
k

]


+

x


[

n
+
k
-
N

]



)






for





N





odd










y


[
n
]


=



h


[



N
2



]




x


[



N
2



]



+




k
=
0





N
2



-
1









h


[
k
]




(


x


[

n
-
k

]


+

x


[

n
+
k
-
N

]



)






for





N





even








(

Eq
.




5

)








y


[
n
]


=




k
=
0




N
2











h


[
k
]




(


x


[

n
-
k

]


-

x


[

n
+
k
-
N

]



)






for





N





odd










y


[
n
]


=



h


[



N
2



]




x


[



N
2



]



+




k
=
0





N
2



-
1









h


[
k
]




(


x


[

n
-
k

]


-

x


[

n
+
k
-
N

]



)






for





N





even








(

Eq
.




6

)








By means of Eq. 5 and Eq. 6 the number of multiplications per output sample #M is reduced from N+1 to










#





M

=




N
2



+
1





(

Eq
.




7

)







Another generally known principle is multi-phase decomposition for sample rate conversion. Equation 8 defines the polyphase components hλ(n·LTƒ) for an L-fold decomposition of impulse response h(k·Tƒ).

h′λ(n·LTƒ)=h((nL+λ)Tƒ)  (Eq. 8)

    • with λ=0 . . . L−1


      The filter h(k·Tƒ) has a frequency response to select the desired signal image (for an interpolator) or anti-aliasing band (for a decimator).


      The corresponding Z-transforms are combined in vector H(z) (Eq. 9).

      Hλ(z)=Z{(hλ(n·LTƒ)}
      H(z)=[H0(z) . . . H1(z) . . . HL−1(z)]T  (Eq. 9)

      The polyphase equation for an L-factor interpolator is shown in Eq. 10:

      Y(z)=H(zX(z)  (Eq. 10)

      whereby X(z) denotes the Z-transform of the input signal x(n·LT) and Y(z) the decomposition of the output signal y(k·Tƒ) as defined in Eq. 11.

      yλ(n·LTƒ)=y((nL+λ)Tƒ)
      Yλ(z)=Z{yλ(n·LTƒ)}
      Y(z)=[Y0(z) . . . Y1(z) . . . YL−1(z)]T  (Eq. 11)
    • with λ=0 . . . L−1


      For an L-factor decimator the polyphase equation is shown in equation 12.

      Y(z)=HT(zX(z)  (Eq. 12)

      y(n·LTƒ) hereby represents the output signal. The decomposition of the input signal x(k·Tƒ) is as defined in equation 13.

      xλ(n·LTƒ)=x((nL+λ)Tƒ)
      Xλ(z)=Z{xλ(n·LTƒ)}
      X(z)=[X0(z) . . . X1(z) . . . XL−1(z)]T  Eq. 13
    • with A=0 . . . L−1


      The efficiency of the polyphase realisations (as in Eqs. 10 and 12) lies in the fact that all filter operations are performed at the lowest rate, i.e. at 1/(LTƒ) instead of at 1/Tƒ. The average number of multiplications per output sample and per Tƒfor a polyphase interpolator (or decimator) is given in Eq. 14.










#





M

=


N
+
1

L





(

Eq
.




14

)







However, the h(kTƒ) (anti-)symmetry generally does not yield intra-phase (anti)-symmetry in most hλ(nLT) phases. The 2-fold decomposition for an even (N=8) order impulse response h(kTƒ) as shown in Eq. 15 is one example of an exception. Since h8=i=hi, both phases H0(z) and Hf(z) are symmetric.

H0(z)=h0+h2z−1+h4z−2+h6z−3+h8z−4
H1(z)=h1+h3z−1+h5z−2+h7z−3  (Eq. 15)

In Eq. 16 (where L=3 and N=11), H1(z) is symmetric for a symmetric H(z), but the other phases H0(z) and H2(z) are not symmetric.

H0(z)=h0+h3z−1+h6z−2+h9z−3
H1(z)=h1+h4z−1+h7z−2+h10z−3
H2(z)=h2+h5z−1+h8z−2+h11z−3  (Eq. 16)

However, in general, the multiplication reduction offered by filter response (anti-)symmetry, as in (Eq. 4) and (Eq. 5), is partially or completely lost with a polyphase realisation.


SUMMARY

The present invention aims to provide an integer factor sample rate conversion circuit, wherein the benefits of a symmetrical or asymmetrical impulse response on the one hand and those of a polyphase realisation on the other hand are combined.


The present invention relates to a circuit for converting the sample rate of a digital signal, comprising

    • an input for applying the digital signal,
    • a conversion filter having either a symmetrical or anti-symmetrical impulse response and implemented as a plurality of subfilters in parallel, each subfilter having a symmetrical or anti-symmetrical response derived from (a combination of) components of a polyphase decomposition of the impulse response,
    • combining means deriving from said applied digital signal input signals of said plurality of subfilters or for combining output signals of said plurality of subfilters into a digital signal with converted rate,
    • an output for outputting the digital signal with converted sample rate.


In a preferred embodiment the combining means are arranged for combining and interpolating output signals of the subfilters. The combining means further comprise means for adding and multiplying signals. The described circuit then acts as an L-factor interpolator.


In another preferred embodiment the combining means are arranged for decimating the input digital signal and for combining input signals to the subfilters. The combining means further comprise means for adding and multiplying signals. The circuit for sample rate conversion then advantageously further comprises adder means for summing output signals of the subfilters. The described circuit then acts as an L-factor decimator.


In another aspect the invention relates to an integrated circuit comprising a circuit for sample rate conversion as described.





SHORT DESCRIPTION OF THE DRAWINGS


FIG. 1 represents an L-factor interpolator circuit according to the invention.



FIG. 2 represents an L-factor decimator circuit according to the invention.





DETAILED DESCRIPTION

In the description of the invention below a Finite Impulse Response (FIR) with a symmetrical impulse response h(k) of length N+1 is assumed. Derivations for anti-symmetrical finite impulse response filters are similar and can readily be performed by a person skilled in the art. The filter is used as anti-imaging or anti-aliasing filter for L-factor interpolation or decimation, respectively.


The present invention aims at exploiting the h(k) symmetry and so to reduce the number of multiplications in a polyphase implementation, i.e. further than the factor L reduction in Equation 14.


The reduction is based on the fact that for each Hλ(z) that is not symmetrical by itself (“intra” symmetrical), a polyphase component with the “flipped” response exists. Define two transfer functions of order P, A(z) and B(z), as complementary if equation 17 is true. A(z) then is a “flipped” version of B(z), and vice versa.

A(z)=z−PB(z−1)  (Eq. 17)

First it is shown that each polyphase component Hλ(z) has a complement Hκ(z), according to equation 17, with P and κ given in equations 18 and 19, respectively. Note that P is always an integer number.









P
=


N
-
λ
-


(

N
-
λ

)






mod





L


L





(

Eq
.




18

)






κ
=


(

N
-
λ

)






mod





L





(

Eq
.




19

)








Hence, it has to be proven that equation 20 is an identity.

Hλ(z)=z−PHκ(z−1)  (Eq. 20)

Substituting (Eq. 19) yields (Eq. 21)

Hλ(z)=z−PH(N−λ)mod L(z31 1)  (Eq. 21)

Using the definition of Z-transform yields equation 22.













i
=

-




i
=










z

-
i





h
λ



(
i
)




=




j
=

-




j
=










z

-

(

P
-
j

)






h


(

N
-
λ

)


mod





L




(
j
)








(

Eq
.




22

)








Substituting equation 8 yields equation 23.













i
=

-




i
=










z

-
i




h


(

λ
+
Li

)




=




j
=

-




j
=










z

-

(

P
-
j

)





h


(



(

N
-
λ

)






mod





L

+
Lj

)








(

Eq
.




23

)








Polynomes are equal if, for each power of z, corresponding coefficients are equal. So, substituting j with P−i yields equation 24.

h(λ+Li)=h((N−λ)mod L+L(P−i))  (Eq. 24)

Using equation 2 yields equation 25.

h(λ+Li)=h(N−(N−λ)mod L−L(P−i))  (Eq. 25)

Substituting Equation 18 yields equation 26.











h


(

λ
+
Li

)


=

h


(

N
-


(

N
-
λ

)






mod





L

-

L


(



N
-
λ
-


(

N
-
λ

)






mod





L


L

-
i

)



)










h


(

λ
+
Li

)


=

h


(

N
-


(

N
-
λ

)






mod





L

-
N
+
λ
+


(

N
-
λ

)






mod





L

+
Li

)














h


(

λ
+
Li

)


=


h


(

λ
+
Li

)








(

Q
.
E
.
D
.

)







(

Eq
.




26

)








It is to be noted that if λ=(N−λ)mod L, the complement of Hλ(z) is Hλ(z) itself, in which case Hλ(z) is symmetrical. The fact that all polyphase components Hλ(z) are complementary, either with Hλ(z) itself or with another polyphase component, can be exploited to reduce the number of multiplications. This is elaborated next.


Define the L×L matrix A=[αi,j], constructed row by row αλT according to the following procedure (given in pseudocode):














for (λ = 0; λ L; ++λ)// start with row 0, end with row L-1


 κ = (N − λ)modL; //find complementary phase with Eq. 19








 if (κ == λ)
// symmetrical phase







  // type 1 row: all 0 except 1 on diagonal





  
aλT=0aλ,λ=1;(Eq.27)












 else if (κ > λ)
//2 phases with flipped response







  // row of type 2: all 0 except two 1's





  
aλT=0aλ,λ=aλ,Κ=1;//(Eq.28)






  // row of type 3: all 0 except single 1 and −1





  
aΚT=0;aΚ,λ=1;aΚ,Κ=-1;//(Eq.29)










Matrix A has the following properties:

    • 1. AT=A (A is symmetric)
    • 2. A is invertible.
    • 3. inner product <αiTj>=0 if i≠j (the rows in A are orthogonal)
    • 4. For the three types of rows aiT defined above
      • Type 1: <aiT,ai>=1
      • Type 2: <aiT,ai>=2
      • Type 3: <aiT,ai>=2


        Using properties 3 and 1, ATA is a diagonal matrix (Eq. 30).














A
T


A

=

[







a
0
T

,

a
0













a
0
T

,

a

L
-
1
























a

L
-
1

T

,

a
0













a

L
-
1

T

,

a

L
-
1








]







=

[







a
0
T

,

a
0









0















0









a

L
-
1

T

,

a

L
-
1








]








(

Eq
.




30

)








Equation 31 defines an L×L matrix D:









D
=

[




1




a
0
T

,

a
0










0















0






1




a

L
-
1

T

,

a

L
-
1









]





(

Eq
.




31

)








Then equation 32 is true, with IL×L the L×L identity matrix.

DATA=IL×L  (Eq. 32)

Substituting (Eq. 32) into (Eq. 10) yields equations 33 and 34.

G(z)=A H(z)  (Eq. 33)
Y(z))=DATG(z)X(z)  (Eq. 34)


Next it is pointed out why applying the expression of equation 34 is more efficient than that of Eq. 10 in terms of number of multiplications.

    • 1. G(z) can be precomputed (offline), so Eq. 33 does not take any real-time computing resources, and hence only Eq. 34 is considered.
    • 2. AT is very sparse and contains only 1 or −1 as non-zero values. Hence, multiplication with AT has no multiplication cost.
    • 3. D is a diagonal matrix. Using property 4, the values






1




a
i
T

,

a
i









on its diagonal are either 1 or ½. Hence, multiplicating with D involves virtually no multiplication cost.

    • 4. Three types of Gi(z) as elements of G(z) exist, depending on the row type aiT in AT.


      Type 1:


Since (N−i)mod L=i, Hi(z) is its own complement, i.e. is symmetrical.


Using (Eq. 27) and (Eq. 33), Gi(z)=Hi(z) and so Gi(z) inherits the symmetry present in Hi(z).


Type 2:


Using (Eq. 28)

Gλ(z)=Hλ(z)+Hκ(z)

Using (Eq. 19)

Gλ(z)=Hλ(z)+H(N−λ)mod L(z)

Replace z by z−1 and multiply both sides with z−P
z31 PGλ(z−1)=z−PHλ(z)+z−PH(N−λ)mod L(z−1)

Using equation 21 (which is true because Eq. 20 is proven to be true):

z−PGA(z−1)=H(N−λ)mod L(z)+Hλ(z)
z−PGλ(z−1)=Gλ(z)

Hence Gλ(z) is symmetric.


Type 3:


Using (Eq. 29):

Gλ(z)=Hλ(z)−Hκ(z)

Using (Eq. 19)

Gλ(z)=Hλ(z)−H(N−λ)mod L(z)

Replace z by z−1 and multiply both sides with z−P
z−PGλ(z−1)=z−PHλ(z−1)−z−PH(N−λ)mod L(z−1)

Using equation 21 (which is true because Eq. 20 is proven to be true):

z−PGλ(z−1)=H(N−λ)mod L(z)−Hλ(z)
z−PGλ(z−1)=−Gλ(z)

Hence Gλ(z) is anti-symmetric.


It has thus been shown that all elements Gi(z) are either symmetrical or anti-symmetrical functions of order P, with P given in (Eq. 18). Substituting P for N into (Eq. 7) and summing over all i=0 . . . L−1, averaged over L output samples yields the average number of multiplications per output sample #M (Eq. 35).










#





M

=


1
L






i
=
0


L
-
1








(





N
-
i
-


(

N
-
i

)






mod





L



2
·
L




+
1

)







(

Eq
.




35

)








For N>>L,








#





M






N

2
·
L




+
1


,





which is approximately a factor of two improvement versus (Eq. 14).


The original polyphase filtering operation H(z) in Eq. 10 is replaced by a sequence of 3 operations (Eq. 34) applied in the order G(z), AT, D.



FIG. 1 shows the corresponding schematic for an L-factor interpolator realisation. As represented by the arrows at the bottom in FIG. 1, the signal flows from left to right, starting with a scalar input signal X(z) and transformed consecutively by G(z), AT and D, into the polyphase vector Y(z).


First, X(z) is multiplied with G(z), a vector with only symmetrical and anti-symmetrical transfer functions as elements. This fact can be exploited for multiplication reduction, as explained before.


Then, the signal is multiplied with AT, a sparse vector with only 1 and −1 as non-zero values.


Then, the signal is multiplied with D, a diagonal matrix with either 1 or ½ on the diagonal.


Finally, the vector signal Y(z) is combined into the (interpolated) higher-rate, scalar output signal Y(z). This combination is the inverse operation of polyphase decomposition (Eq. 11). Note that this final stage has the behaviour of a parallel-to-serial conversion, or also of a fast commuting switch, picking in turn a value for Y(z) from each of the Yλ(z) element.


Depending on the actual value of (L,N), matrix AT consists of a number of Type 1 rows (Eq. 27) and a number of Type 2 (Eq. 28) and Type 3 (Eq. 29) row pairs, as explained before. The example in FIG. 1 shows one diaiTgi(z) row for αiT of Type 1, i.e. i=(N−i)modL, and one row pair for j≠(N−j)modL, i.e. with rows αiT and α(N−j)modLT of Types 2 and 3, respectively.


Equation 36 is the L-phase decimator equivalent for (Eq. 34).

Y(z)=GT(z)ADTX(z)  (Eq. 36)

The same expression for #M as given in equation 35 applies. FIG. 2 shows the corresponding schematic for an L-factor decimator realisation. As represented by the arrows at the bottom in FIG. 2, the signal flows from left to right. First, according to Eq. 13, a serial-to-parallel converter performs a polyphase decomposition from the scalar input signal X(z) into the output vector X(z).


Then, the vector X(z) is multiplied with diagonal matrix DT, with either 1 or ½ on the diagonal.


Next, the signal is multiplied with A, a sparse matrix with only 1 and −1 as non-zero values.


Finally, a dot product with GT(z) results in the scalar low rate (decimated) output signal Y(z). All elements of vector GT(z) are either symmetrical or anti-symmetrical transfer functions. This can be exploited to reduce the number of multiplications, as explained before.


Depending on the actual value of (L,N), matrix A consists of a number of Type 1 rows (Eq. 27) and a number of Type 2 (Eq. 28) and Type 3 (Eq. 29) row pairs.


The example in FIG. 2 shows one row for aiT of Type 1, i.e. i=N−i)modL, and one row pair for j≠(N−j)modL, i.e. with rows aiT and α(N−j)modLT of Types 2 and 3, respectively.

Claims
  • 1. Circuit for converting the sample rate of a digital signal, the circuit comprising an integer factor sample rate conversion circuit, wherein the benefits of a symmetrical or asymmetrical impulse response on the one hand and those of a polyphase realization on the other hand are combined, the circuit further comprising an input for applying said digital signal,a conversion filter having either a symmetrical or anti-symmetrical impulse response and a polyphase realization, the conversion filter implemented as a plurality of subfilters in parallel, each subfilter having a symmetrical or anti-symmetrical response derived from a combination of the conversion filter polyphase components, the combination including multiplying the digital signal only with factors of 1 and −1,combining means for combining said applied digital signal input signals of said plurality of subfilters or for combining output signals of said plurality of subfilters into a digital signal with converted sample rate, after multiplying the applied digital signal input signals of said plurality of subfilters or after multiplying the output signals of said plurality of subfilters, with factors 1 or ½,an output for outputting said digital signal with converted sample rate.
  • 2. A circuit for converting the sample rate of a digital signal, comprising: an input for applying said digital signal,a conversion filter having either a symmetrical or anti-symmetrical impulse response and implemented as a plurality of subfilters in parallel, each subfilter having a symmetrical or anti-symmetrical response derived from components of a polyphase decomposition of said impulse response,combining means for deriving from said applied digital signal input signals of said plurality of subfilters or for combining output signals of said plurality of subfilters into a digital signal with converted sample rate, andan output for outputting said digital signal with converted sample rate,wherein said combining means are arranged for combining and interpolating output signals of said subfilters, said combining means further comprising means for adding and multiplying signals.
  • 3. Circuit for sample rate conversion as in claim 2, whereby said combining means are arranged for decimating said input digital signal and for combining input signals to said subfilters, and combining means further comprising means for adding and multiplying signals.
  • 4. Circuit for sample rate conversion as in claim 3, further comprising adder means for summing output signals of said subfilters.
  • 5. An integrated circuit comprising a circuit for sample rate conversion as in claim 2.
Priority Claims (1)
Number Date Country Kind
0716942.8 Aug 2007 GB national
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Number Name Date Kind
5521946 Main May 1996 A
5594675 Peng Jan 1997 A
5717715 Claydon et al. Feb 1998 A
5892468 Wilson et al. Apr 1999 A
6362760 Kober et al. Mar 2002 B2
6531969 Chu Mar 2003 B2
7002940 Welborn Feb 2006 B2
20030067973 Lee et al. Apr 2003 A1
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Foreign Referenced Citations (1)
Number Date Country
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Related Publications (1)
Number Date Country
20090058692 A1 Mar 2009 US