2011 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics October 16-19, 2011, New Paltz, NY ARBITRARY SAMPLE RATE CONVERSION WITH RESAMPLING FILTERS OPTIMIZED FOR COMBINATION WITH OVERSAMPLING Andreas Franck Fraunhofer IDMT Ehrenbergstraße 31 98693 Ilmenau, Germany [email protected] ABSTRACT Arbitrary sample rate conversion (ASRC) is used in many applications of DSP. ASRC algorithms based on integer-ratio oversampling and continuous-time resampling filters enable good resampling quality for wideband signals. A previous publication introduced an overall optimization scheme for structures based on oversampling and Lagrange interpolation to design the oversampling component such that the design error of the overall frequency response is minimized with respect to a selectable error norm. However, the achievable quality strongly depends on the continuous-time resampling filters. Lagrange interpolators show severe deficiencies when used in this role. The present paper proposes a design objective for continuoustime resampling filters that are specifically adapted for use with oversampling and the overall optimization scheme. ASRC systems utilizing these so-called optimized image band attenuation (OIB) resampling functions achieve significant quality improvements over existing approaches. Performance analyses show that this class of filters enables implementations with reduced complexity for a wide range of design specifications. Index Terms— Sample rate conversion, Delay filters, Acoustic signal processing 1. INTRODUCTION Arbitrary sample rate conversion (ASRC) is useful in many applications of audio signal processing, for instance for converting audio signals between different standard sample rates [1], digital audio effects [2], or sound reproduction systems such as wave field synthesis [3]. ASRC can be considered as a generalization of conventional, rational-ratio sample rate conversion (SRC) techniques (e.g. [4]) that exhibits several advantages. First, it supports arbitrary conversion ratios R = Ti /To , where Ti and To denote the input and output sampling periods. Moreover, ASRC algorithms allow multiple conversion ratios without requiring separate filter designs, and they enable continuously time-varying ratios. For this reason, ASRC techniques can be used to interface asynchronous systems with different sampling clocks [1]. A convenient way to describe ASRC algorithms is the hybrid analog/digital model [4, 5]. It models the conversion process by a discrete-to-continuous conversion (D/C) followed by resampling at This work has been supported by the DIOMEDES project, funded under the European Commission ICT 7th Framework Programme. c 978-1-4577-0693-6/11/$26.00 2011 IEEE x[n] L Hdig (e jω ) Integer-ratio SRC Hint (jω) y[m] Continuous-time resampling filter Figure 1: Signal flow of an ASRC structure based on integer oversampling and a continuous-time resampling filter the output sampling period To . In this way, the ASRC algorithm is completely determined by the frequency response of the continuous-time anti-imaging filter of the C/D process, denoted Hc (jω). The frequency response of the ideal anti-imaging filter is given by ( Ti , |ω| ∈ Xp b . (1) Hc (jω) = 0 , |ω| ∈ Xs Here, Xp = [0, ωc ] and Xs = [2π − ωc , ∞) represent the passband and stopband intervals of the design specification, respectively. ωc denotes the cutoff frequency of the input signal. For notational convenience, the angular frequency ω = 2πf Ti is normalized to the input period Ti . Two convenient measures for the resampling quality, the maximum passband error δp and the minimum stopband attenuation As , are directly obtained from the frequency response Hc (jω) b c (jω)| δp = max |Hc (jω) − H (2) b c (jω)| . As = −20 log10 max |Hc (jω) − H (3) ω∈Xp ω∈Xs Algorithms for ASRC fall into three general classes: Methods based on numerical interpolation techniques such as Lagrange or spline interpolation, piecewise polynomial resampling filters such as the modified Farrow structure [5], and methods based on oversampling and a continuous-time resampling function (e.g. [1, 6, 7]). 2. STRUCTURES BASED ON OVERSAMPLING AND CONTINUOUS-TIME RESAMPLING FILTERS The widespread use of structures based on integer-ratio oversampling and fixed continuous-time resampling filters is justified by several reasons. First, it enables efficient conversion of wideband signals. Second, it allows the multitude of well-established design techniques and implementations for rational-factor SRC to be used. 149 2011 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics Hc (jω) 1.001 1 0.999 ω 0.998 0π 0.2π 0.4π 0.6π 0.8π 1π |Hc (jω)| φ 0 dB φ0 Xp0 Hc (jω) Lagrange Hc (jω) Lagrange (opt) ω Hint (j L ) Lagrange −40 dB |Hc (jω)| 0 φ0 Xp φ0 −20 dB (a) Passband detail −60 dB −65 dB −70 dB October 16-19, 2011, New Paltz, NY −60 dB −80 dB 5π 5.5π 6π 6.5π 7π ω −100 dB 0π 2π (b) Stopband detail 4π 6π 8π ω 12π 10π (c) Frequency response Figure 2: Frequency response of an Oversampling+Lagrange structure, overall optimization scheme compared to conventional design. Parameters L = 3, Nint = 5, Ndig = 159, ωc = 0.85π, L∞ design. Xp0 and φ0 denote the passband and transition band images of Hdig (e jω ). The signal flow of this structure is depicted in figure 1. The oversampling component consists of a sample rate expander followed by a discrete-time anti-imaging filter, or prefilter, Hdig (e jω ). Although the continuous-time resampling filter Hint (jω) conceptually operates on continuous-time signals, it is generally implemented as a discrete-time filtering operation which evaluates the signal value only at the requested output instants. In conventional approaches, these two components are designed independently. In most cases. Hdig (e jω ) is designed as a lowpass filter according to design specifications for integer-ratio SRC systems, e.g. [4]. Likewise, Hint (jω) typically forms a simple resampling filter, such as a low-order Lagrange interpolator. However, this independent design has two major drawbacks. First, the resulting ASRC structures are not optimal with respect to a given error norm such as the (weighted) least squares L2 or the minimax L∞ norm. Moreover, it prohibits the inclusion of additional time- or frequency-domain conditions. 2.1. Overall Optimization of the Discrete-Time Prefilter To overcome these deficiencies, an overall optimization scheme for the discrete-time prefilter Hdig (e jω ) has been proposed in [8]. Although this approach uses Lagrange interpolators for Hint (jω), the general idea is readily applicable to other resampling filters. The overall continuous-time frequency response of the system can be expressed as Hc (jω) = jω 1 Hdig e L Hint L jω L . (4) As the prefilter Hdig (e jω ) is advantageously implemented as a linear-phase FIR filter, e.g. [4], its frequency response is given by 0 Ndig jω Hdig (e ) = X 0 b[n] trig(n, ω) with Ndig = j Ndig −1 2 k , (5) n=0 Ndig even, n = 0 1 , trig(n, ω) = 2 cos(nω) , Ndig even, n > 0 . 2 cos n + 1 ω , N odd dig 2 (6) Thus, the overall frequency response can be stated as a linear combination of basis functions Go (n, ω) 0 Ndig Hc (jω) = X b[n]Go (n, ω) with (7) n=0 ω jω 1 Hint . Go (n, ω) = trig n, L L L (8) In this way, the design of the filter coefficients b[n] can be stated as an optimization problem with respect to a norm Lp 0 PNdig b c (jω) . minimize b[n]G (n, ω) − H (9) o n=0 {b[n]} p For widely-used error norms such as the L2 or the L∞ norm, (9) can be efficiently solved as a convex optimization problem. For example, a method operating on a discretized representation of the approximation region X = Xp ∪ Xs is described in [8]. A design example for this overall optimization scheme is shown in figure 2. In this example, the passband error δp is reduced to 48.3 % compared to a conventionally designed prefilter, while the minimum stopband attenuation is increased by 4.0 dB. This performance improvement follows from two causes: In the passband region, the magnitude roll-off of the Lagrange interpolator is compensated by the design method, yielding an approximately equiripple behavior. In the stopband region, the improvement results from the shaping of the frequency response in the image regions φ0 of the transition band φ = ( ωLc , 2π − ωLc ) of the discrete-time prefilter. 2.2. Alternative Continuous-Time Resampling Filters However, the above design example also reveals shortcomings of Lagrange interpolation when used in combination with oversampling. On the one hand, the limited attenuation of Hint (jω) in the passband images of Hdig (e jω ) (denoted Xp0 in figure 2) effectively limits the achievable stopband attenuation. On the other hand, the relatively flat passband response of Lagrange interpolators is possibly not required in this application, since most passband errors are readily compensated in the design of the prefilter. For these reasons, the use of alternative resampling filters, such as B-spline basis functions and O-MOMS functions (Optimal maximal-order interpolation of minimal support) [9], appears promising. 150 2011 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics Hc (jω) 1 |Hc (jω)| 0 dB 0.9 ω 0.8 0π 0.2π 0.4π 0.6π 0.8π 1π (a) Passband detail Hc (jω) O-MOMS Hc (jω) OIB ω Hint (j L ) O-MOMS ω Hint (j L ) Oib −20 dB −40 dB −60 dB −80 dB |Hc (jω)| −100 dB −100 dB −120 dB −125 dB −150 dB October 16-19, 2011, New Paltz, NY 5π 5.5π 6π 6.5π 7π ω −140 dB 0π 2π (b) Stopband detail 4π 6π 8π 10π ω 12π (c) Frequency response Figure 3: Frequency response of an Oversampling+OIB structure, comparison to Oversampling+O-MOMS design. Parameters L = 3, Nint = 5, Ndig = 159, ωc = 0.85π, L∞ design . class of functions to model such resampling filters. They include Lagrange interpolators, B-splines, and O-MOMS functions as subsets and are efficiently implemented by the modified Farrow structure. Their frequency response takes the form ω |Hint (j L )| 0 dB −40 dB −80 dB −120 dB −160 dB Lagrange B-spline O-MOMS OIB 0 Hint (jω) = M N int X X bmn G(m, n, ω) , (10) m=0 n=0 ω 2πL−ωc 2πL+ωc Figure 4: Comparison of different resampling functions Hint (jω). Parameters Nint = 5, L = 3, ωc = 0.85π. Hatched area represents the first passband image of Hdig (e jω ). Spline basis functions are widely used in DSP, for instance in image processing [10]. Compared to Lagrange interpolation, they provide superior image attenuation with an asymptotic rate of decay proportional to ω −Nint −1 at the expense of a more severe passband roll-off. O-MOMS functions aim at minimizing the L2 approximation error for a given length of the resampling filter. In resampling applications, they effectively decrease the magnitude of the first signal image while reducing the asymptotic rate of decay. For more information, the reader is referred to [9]. The use of these resampling functions in combination with the proposed overall optimization scheme is investigated in [11]. For the design example of figure 2, the use of B-spline and O-MOMS resampling filters increases the minimum stopband attenuation by 27.6 dB and 38.8 dB, respectively. The passband errors are decreased by the same ratios. 3. OPTIMIZED IMAGE BAND ATTENUATION DESIGN Notwithstanding the improvements gained by the resampling filters considered in the preceding section, none of these are specifically adapted to ASRC systems incorporating oversampling. It is therefore worthwhile to consider design specifications for Hint (jω) that take the characteristics of this structure into account. Symmetric piecewise polynomials, e.g. [5], form a suitable where bmn form the elements of a coefficient matrix and G(m, n, ω) are basis functions as defined, for instance, in [5]. As argued above, the attenuation of Hint reS (jω) in theωimage c gions of the passband of Hdig (e jω ), Xs0 = ∞ k=1 [2πk − L , 2πk + ωc ], is of paramount importance for the performance of the comL plete system. In contrast, passband deviations of Hint (jω) can be rectified within certain limits by an appropriately designed prefilter. However, design specifications purely based on the stopband behavior typically lead to degenerate solutions, or require a large passband gain of Hdig (e jω ), which in turn deteriorates the overall performance. A relatively loose specification for the maximum passband error, e.g. δp = 0.5, proves to be a sensible choice. Thus, a suitable design specification reads 0 X Nint M X (11) minimize max0 bmn G(m, n, ω) {bmn } ω∈Xs m=0 n=0 0 X Nint M X bmn G(m, n, ω) − 1 ≤ δp , 0 ≤ ω ≤ ωLc . subject to m=0 n=0 In figure 4, a function designed according to an OIB specification is compared to the other resampling filters considered in this paper. It is observed that the minimum stopband attenuation in the passband image region is considerably higher than for Lagrange, Bspline, and O-MOMS functions. On the other hand, the passband roll-off is comparable than that of B-spline and O-MOMS filters. The design example of section 2 is repeated with an OIB resampling filter and a discrete-time prefilter designed with the overall optimization scheme. The resulting frequency response is shown in figure 3. Compared to the O-MOMS design, the minimum stopband attenuation is further increased by about 17.6 dB. Again, the 151 2011 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics Method δp /δp∗ δp ∗ Lagrange (conventional) Lagrange (optimized) B-spline basis function O-MOMS OIB As −3 As − A∗s 1.37 · 10 − 59.6 dB − 6.61 · 10−4 48.29 % 63.6 dB 4.0 dB 2.75 · 10−5 2.01 % 91.2 dB 31.6 dB 7.64 · 10−6 0.56 % 102.4 dB 42.8 dB 1.01 · 10−6 0.07 % 120.0 dB 60.4 dB Table 1: Performance comparison for ASRC designs based on oversampling. Parameters L = 3, Nint = 5, Ndig = 159, ωc = 0.85π. Improvements δp /δp∗ and As − A∗s with respect to conventional Oversampling+Lagrange design 700 600 500 Farrow structure Oversampling+Lagrange Oversampling+B-spline Oversampling+O-MOMS Oversampling+OIB October 16-19, 2011, New Paltz, NY B-spline or O-MOMS basis function require significantly less instructions. The OIB design gains a further significant reduction of the complexity. For example, the modified Farrow structure and the Oversampling+Lagrange structure require 433 and 461 instructions, respectively, to compute one output sample for As = 100 dB. In contrast, the OIB design reduces the instruction count to 238. 5. CONCLUSIONS In this paper, ASRC structures based on integer-ratio oversampling and continuous-time resampling functions have been considered. It has been demonstrated that the performance is significantly improved compared to existing approaches by using resampling functions specifically adapted to this structure. The proposed optimized image band attenuation (OIB) filters are conveniently combined with an overall optimization scheme that incorporates the characteristics of the resampling filter into the design of the oversampling component. In this way, considerable complexity reductions are achieved for a wide range of design specifications. 400 6. REFERENCES 300 [1] R. Lagadec, D. Pelloni, and D. Weiss, “A 2-channel, 16-bit digital sampling frequency converter for professional digital audio,” in Proc. IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, vol. 7, May 1982, pp. 93–96. 200 100 40 dB [2] J. O. Smith, S. Serafin, J. Abel, and D. Berners, “Doppler simulation and the Leslie,” in Proc. 5th Int. Conf. on Digital Audio Effects (DAFx-02), Sept. 2002, pp. 13–20. As 60 dB 80 dB 100 dB 120 dB [3] A. Franck, K. Brandenburg, and U. Richter, “Efficient delay interpolation for wave field synthesis,” in AES 125th Convention, Oct. 2008. Figure 5: Minimum number of instructions to compute one output sample for minimum stopband attenuation levels As = 40, 45, . . . , 120. Parameters ωc = 0.85π, conversion ratio R ≈ 1. [4] R. E. Crochiere and L. R. Rabiner, Multirate Digital Signal Processing. Eaglewood Cliffs, NJ: Prentice Hall, Inc., 1983. [5] J. Vesma and T. Saramäki, “Polynomial-based interpolation filters—Part I: Filter synthesis,” Circuits Systems Signal Process., vol. 26, no. 2, pp. 115–146, Apr. 2007. passband error is reduced by the same amount due to the uniform error weighting utilized. The performance improvements gained by the resampling filters considered in this paper are summarized in table 1. [6] T. A. Ramstad, “Digital methods for conversion between arbitrary sampling frequencies,” IEEE Trans. Acoust., Speech, Signal Processing, vol. 32, no. 3, pp. 577–591, June 1984. 4. PERFORMANCE COMPARISON The above design example shows that the OIB design enables significant quality improvements for fixed, suitably chosen design parameters L, Ndig , and Nint . However, a criterion more relevant to practical application is the minimal computational complexity required to reach a prescribed quality requirement. In figure 5, the minimum number of instructions to compute one output sample is displayed as a function of the achieved stopband attenuation As . In this example, a cutoff frequency of ωc = 0.85π and a conversion ratio close to unity R ≈ 1 are assumed. For a large number of design parameter variations (M = 1, . . . , 7, N = 1, . . . , 72 for the modified Farrow structure and L = 1, . . . , 6, Ndig = 1, 5, . . . , 305, Nint = 1, . . . , 7 for structures comprising oversampling), the attained quality and the required computational effort are evaluated. The designs that meet a prescribed performance goal are selected for several stopband attenuation levels. It is observed that the modified Farrow structure and Oversampling+Lagrange designs exhibit similar complexity for all considered quality levels. In contrast, the designs using oversampling and [7] G. 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