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Saturday, October 11 4:00 pm – 5:30 pm
Session Z5 Posters: Signal Processing, Part 2

Z5-1 Embedded Digital Filters for PWM GeneratorsAlberto Bellini, University of Parma, Parma, Italy
Thanks to their intrinsic high efficiency audio power switching amplifiers are becoming widespread in many applications where heating and costs are major concerns. Several topologies exist to reduce augmented distortion, and many of them rely on digital signal processors (DSPs). Nowadays the market offers several products with integrated peripherals for analog signal sampling and PWM generation. However, the latter operation is still a time consuming task, often performed with a number of different approaches. This paper presents a method for an efficient computation of the PWM signal corresponding to the input audio signal, suitable to feed a switching output power stage. The presented method exploits DSP capabilities and it is oriented to DSPs that integrate PWM generators. Its peculiar characteristic is the possibility to perform N taps FIR operation on the audio signal together with the computation of a suitable PWM signal with N MAC operations per sample.

Z5-2 Further Investigations of Inverse FilteringScott G. Norcross, Gilbert A. Soulodre, Michel C. Lavoie, Communications Research Centre, Ottawa, Ontario, Canada
Previous work has shown that inverse filtering can degrade the subjective quality of audio signals in certain conditions. Minimum phase inversion and regularization applied separately have also been studied and can be effective in some cases, but neither technique has proven to be robust. In this paper further methods involving various regularization methods applied to the full and minimum phase part of the impulse response (IR) are studied. Subjective tests were conducted in accordance with the MUSHRA method to evaluate the performance of the various inversion methods. The results of the subjective tests were also used to determine the effectiveness of the ITU-R PEAQ objective test model as a potential tool in the development and evaluation of inverse filtering techniques.

Z5-3 Pure Linear PredictionAlbertus den Brinker, Felip Riera-Palou, Philips Research Laboratories, Eindhoven, The Netherlands
Linear prediction (LP) has traditionally been used in speech coding. Recently, variants of LP have also shown to be appropriate for audio coding. In this paper we introduce a new prediction scheme, called pure linear prediction (PLP), which combines important features from previous approaches. We show that the modeling capability of the PLP can be tuned in a psychoacousticaly relevant way making it suitable for speech and audio coding. Moreover, under certain restrictions, this new scheme is directly realizable, stable, and retains the whitening property of conventional linear prediction. The processing of the prediction coefficients to perform operations such as quantization, interpolation, and spectral broadening is also addressed. As an example of the application of the PLP, we describe its use in the context of the sinusoidal coder proposed by Philips which is being standardized in MPEG-4 Extension 2.

Z5-4 Design of Low-Order Filters for Radiation SynthesisPeter Kassakian, David Wessel, University of California, Berkeley, Berkeley, CA, USA
A fundamental goal of sound synthesis is to reproduce, and to control, as many facets of the sound as possible. By numerically solving a carefully constructed optimization problem, we are able to design low-order filters for use with a dodecahedral loudspeaker array to synthesize low order spherical harmonics over specified frequency ranges. The method, a variant of least-squares, is general, allowing for the inclusion of side constraints, arbitrary array geometry, and incorporation of measured loudspeaker characteristics. We compare the predicted loudspeaker array performance with high-resolution measurements of the physical system.

Z5-5 A Numerical Method to Modify the NBR 10303 Filter Frequency ResponseAndré Luís Dalcastagnê, Sidnei Noceti Filho, Federal University of Santa Catarina, Florianópolis, Brazil; Homero Sette Silva, Selenium Loudspeakers, Nova Santa Rita, Brazil
This paper describes the design of three filters used in the filtering of pink noise which are described in the new Brazilian standard proposed to replace the current NBR 10303. The NBR 10303 filter does not permit the correct test of subwoofers because its low cut-off frequency is too high and for this reason we changed its component values in order to obtain three new filters with lower low cut-off frequencies. The design of these filters was carried out through a numerical method based on a modification of the NBR 10303 filter frequency response magnitude. The final component values were specified according to commercial values.

Z5-6 Time Delay Spectrometry Processing Using Standard Hardware PlatformsWolfgang Ahnert, ADA Acoustic Design Ahnert, Berlin, Germany; Stefan Feistel, SDA Software Design Ahnert GmbH, Berlin, Germany; Steven McManus, Gold Line Connector Inc., New Bedford, MA, USA; Waldemar Richert, SDA Software Design Ahnert GmbH, Berlin, Germany
The processing power today available on portable computer platforms is now so far advanced that it is no longer necessary to use dedicated digital signal processing platforms for the intensive analysis required in time delay spectrometry (TDS). Moving the processing from a dedicated platform onto a standard personal computer becomes possible with the TDS technology realized in a measurement and post-processing software instead. It also opens the way for more information to be extracted from a measurement after it has been made. Care must still be taken in the choice of data gathering systems, as timing between input and output data samples is critical.

Z5-7 Lossless Signal Processing with Complex Mersenne TransformsJames Angus, Tim Jackson, University of Salford, Salford, UK
The design and implementation of lossless audio signal processing using a Finite Field Transform is shown. In particular Complex Mersenne Transforms are developed. Finite field signal processing techniques are described. The effects of filter length and coefficient accuracy are also discussed. Finite field transform algorithms which would be suitable for lossless signal processing are presented. The paper concludes by presenting an example of lossless processing.

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