SAMPLING RATE CONVERSION USING FRACTIONAL-SAMPLE DELAY

A. Tarczynski*, W. Kozinski** and G.D. Cain***

*SEMSE, University of Westminster, 115 New Cavendish Str., London, W1M SJS, England, on leave from Warsaw University of Technology. * *ISEP, Warsaw University of Technology, Koszykowa 75, 00-662 Warsaw Poland ***SEMSE, University of Westminster, 115 New Cavendish Str., London, W l M SJS, England.
Let y(kT) and x(kT) denote respectively the output and the input signal of the FSD. Its time-domain model can be described as follows:

ABSTMCT In this paper we demonstrate application of the Fractional Sample Delayor (FSD) to constructing sampling rate converters for incommensurate sampling rates. We use Lagrange polynomials for interpolating the values of the signal between sampling instants. It is shown that to deploy this technique effectively one has to use polynomials of high degree. Otherwise the interpolation error for high frequencies (above half Nyquist) is very big. Increasing the polynomial's degree leads inevitably to numerical problems. The interpolation process may be improved when additional samples of the signal are generated before interpolation. It is shown that FSD can be used for achieving this task. The final version of the sampling rate converter comprises two FSDs and a very simple four-point interpolator based on a Lagrange polynomial.

y(kT) = ji(kT-

Z)

(4)

where S (t) is a continuous-time signal obtained by using Shannon's reconstruction of x(kT). If x(kT) is a sampled version of some continuous time signal x,(t) whose spectrum is strictly bandlimited to the interval then obviously (4) becomes:

If1 5

gT
(5)

Y(kT)=x,(kT-t)

1. FRACTIONAL SAMPLJNG DELAYOR A Fractional sample D ~ (FSD) can be described as I ~ ~ ~ an allpass filtcr with arbitrary constant predcfined group delay. In the special case when the delay T is equal to a single sample interval T (or its niultiplc nT) the FSD is equivalent to an ordinary backward shift operator z'l (or z-'I). However. things get more interesting if the delay contains a fractional part of the sampling interval i.e.
': I

nT -k

r1

where f l

E (0,

T)

(1)

in the ideal reconstruction situation. Equations (2) and (5) reveal the enormous attractiveness of FSD utilization. First, (2) shows that FSD is somehow similar to an ordinary backward shift Operator but, due to a continuously variable parameter T > it ~ is much more flexible and versatile than its integer-delay counterpart. On the other hand, (5) shows that FSD gives the insight often desired at locations lying between values of a sampled signal. Both properties allow the designer to tackle many DSP problems at a different level of abstraction. A simple example showing the usefulness of an FSD is that of designing a discrete-time differentiator. In the time domain the differentiator may be described approximately by:

The frequency response of the FSD is:

x (kT) =
~ ( v= ) ,-2n\l'

x( kT) - x( kT - r )
C

(6)

(2)
where T is a sufficiently small time intenal. FSD gives an obvious solution of implementing such a differentiator. Tackling the same problem in 2-l leads to much more tedious work.

were 1' is normalised frequency related to the frequency f (in Hz) as follows:

v=fr

(3) III-285
0-7803-1775-OD4 $3.00 0 1994 IEEE

2. LAGRANGE INTERPOLATING

POLYNOMIALS In this paper we would like to demonstrate the appeal of FSDs for building up sampling rate convertcrs. Except for simple decimating, the change of sampling rate always involves determining the values of the discrcte-time signal between sampling instants. This can bc done using one of methods listed below: (i) effectivelly increasing the sampling rate of the signal by combined interpolation / decimation (e.g. [ 11) (ii) nonlinear interpolation (usually polynomial interpolation) techniques (e.g. 121) (iti) using FSD. Each of these methods has some advantages and disadvantages. Approaches (i) and (iii) cannot be easily used if the time instant at ivhich the value of the signal is to be interpolated varies its relative position for A E[O.T] to exactly fall onto the sampling instants. However. it is worth mentioning that such possibilities for FSD ivere reported [a). [j]. On the other hand. polynomial interpolation techniques beha1.e poorly [31 \\!hen used for reconstructing signals containing high-frequency components. namely those \\hose normalised frequency is

above half Nyquist frequency. Figure 1 presents an error surface for interpolating a signal using a Lagrange polynomial of 9th degree. The axis denoted as "Time" shows the relative position A of the time instant for which interpolation takes place. Figure 2 shows how the error vanishes when thc degree of the interpolating polynomial is increased. The plots are prepared for A=0.5T. Note that increasing the degree of the polynomial from n=9 to n=3 1 has not resultcd in a very significant diminishing of the error function.

0.04

0.03
L

E !
L

UJ

0.02

0.01 0

U
n=15
0.3

0.1

n.2

0.4

Frequency

0.8
0.6
L

Fig. 2. Interpolation error for different degrees of the polynomial

0.4

0.2
0 0.5

ti

Frequency

0 0

Ti r-ri e

The situation is entirely different if one uses for interpolation extra samples between sampling instants. The decrease of the error depends now on the number of extra samples as well as on their distribution within the basic sampling intend. To illustrate some aspects of this problem \\.e will shon once agai:i error surfaces obtained from interpolation using now only 3rd degree Lagrange polynomials when tu'o extra samples are placed at different positions ivithiii each sampling period. Figure 3 presents the interpolation error when a Lagrange polynomial of 3rd degree has been used for interpolating

Fig. 1. The interpolation error for a 9th degree Lagrange polynomial

In-286

the signal and the extra samples wcrc gcnerated at X T and at X T .

Frequency

Time

Fig. 4 . Interpolation error when estimated samples at 0.295T and 0.705T are used.
Frequency
0

Time
3. CONSTRUCTION OF THE SAMPLING RATE

Fig. 3 . Interpolation error when estimated samples at 1/3T and 2/3T are used.

Note that the "worst c;isc" errors \\hich occur for estimating the signal at time instants diffcring from the multiples of the sampling period at 0.12T and 0.88T can be further reduced if we gi\z up equidistant sampling and get additional samples at: 0.295T and 0.705T. The appropriate results are presented i n Figurc 4. Note that the easiest way of obtaining samplcs at 0.295T and 0.705T is ! to use FSDs. To achieve thc same result b increasing the sampling rate. one has to incrcase thc sampling rate 200 times (!) and then pick up c i c q 59th and l-ilst sample.

CONVERTER AND SIMULATION RESULTS Here belou, we present a construction of a simple sampling rate converter which uses four-point (3rd degree) polynomial interpolation carried out on two actual samples of the signal as well as two additional ones generated by means of FSDs. The additional samples are placed at time instants diffcring from the multiples of T by 0.295T and 0.705T. We haiz used IIR delayors designed using the method describcd in [6]. The overall structure of the sampling rate converter is presented i n Figure 5. The broken vertical line scparates the part of the system which works with the input sampling rate from that which works with the output sampling rate. Thc sampling rate converter has been simulated to increasc the sampling period of the following signal s(t) from 1 sec. to 1.0781 sec. :

s ( t ) =(cos(0.3xt)+O._i,sin(O.74xt)+3)u(t)
where u(t) denotes the unit step

(7)

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x(kT)
\

Z-n -n-0.295

taken every 1.0781 sec. while the broken line connects samples obtained at the same time instants by resampling.

~.-polydomial I I 1

I
I I

,
+-Z

interpolation
-11-0.705

z-n-l

bla/ck
I

4. CONCLUSIONS In this paper we have shown that FSDs may effectively be used for constructing sampling rate converters. Use of one or two FSDs significantly enhances the power of interpolation techniques such as polynomial interpolation, allowing reduction of the error introduced by modification of the sampling rate.

Fig. 5. The structure of the sampling rate converter.

5.0

4.0

3.0

REFERENCES [l] Crochiere, R.E. and L.R. Rabiner, Multirate Digital Englewood Cliffs, NJ: Signal Processing, Prentice-Hall. London, 1983. [2] Plybon, B.F. An Introduction to Applied Numerical Analysis, PWS-Kent Publishing Company, Boston, 1992. [3] Ging-Shing Liu and Che-Ho Wei. "A new variable fractional sample delay filter with nonlinear interpolation", IEEE Trans. Czrcuits and Systems II, vol. 39, pp. 123-126, 1992. [4] Hermanowicz, E. "Explicit formulas for weighting coefficients of maximally flat tunable FIR delayers", Electronic Letters. vol. 28, no. 20, 1992, pp. 1936- 1937. [5] Farrow, C.W. "A continuously variable digital delay element". Proc. IEEE Int. Sytiip. on Circuits and Systems. Espoo, vol. 3, pp. 2641-2645, 1988. [6] Tarczynski. A. and G.D. Cain "Design of IIR Filters for approximating perfect fractional-sample delayors", submitted for presentation at ISCAS'94.

2.0

10

20

30

Fig. 6. The samples of the signal and the signal after resampling.

Figure 6 presents both: the signal %(t) sampled with the new sampling period and the signal obtained from resampling. The stars denote the actual samples of %(t)