Paper Archive

Source-filter models are widely used in numerous audio processing fields, from speech processing to percussive/contact sound synthesis. The design of filters for these models—be it from scratch or from spectral analysis—usually involves tuning frequency and damping parameters and/or providing an all-pole model of the resonant part of the filter. In this context, and for the modelling of percussive (non-sustained) sounds, a source signal can be estimated from a filtered sound through a time-domain deconvolution process. The result can be plagued with artifacts when resonances exhibit very low bandwidth and lie very close in frequency. We propose in this paper a method that noticeably reduces the artifacts of the deconvolution process through an inter-resonance phase synchronization. Results show that the proposed method is able to design filters inducing fewer artifacts at the expense of a higher dynamic range.

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Modal synthesis is a practical and efficient way to model sounding structures with strong resonances. In order to create realistic sounds, one has to be able to extract the parameters of this model from recorded sounds produced by the physical system of interest. Many methods are available to achieve this goal, and most of them require a careful parametrization and a post-selection of the modes to guarantee a good quality/complexity trade-off. This paper introduces a two step analysis method aiming at an automatic and reliable identification of the modes. The first step is performed at a global level with few assumptions about the spectro/temporal content of the considered signal. From the knowledge gained with this global analysis, one can focus on specific frequency regions and perform a local analysis with strong assumptions. The gains of such a two step approach are a better estimation of the number of modal components as well as a better estimate of their parameters.

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