Download A multirate, finite-width, bow-string interaction model In this paper we propose an efficient method to model the amount of bow hair in contact with the string in a physical model of a bowed string instrument.
Download 2-D digital waveguide mesh topologies in room acoustics modelling Digital waveguide mesh models have provided an accurate and efficient method of modelling the properties of many resonant structures, including acoustic spaces. 2-D rectilinear and triangular mesh structures have been used extensively in the past to model plates and membranes and are presented here as potential analogues to 2-D acoustic spaces. Impulse response measurements are taken and comparisons are made regarding the spectral content and the associated properties when compared with standard room acoustic parameters. Enhanced mesh structures are examined using frequency warping techniques and high-resolution sampling rates. The 2-D triangular mesh is shown to be considerably superior to the rectilinear mesh in terms of the measurements taken, with a further significant improvement being made by using the same mesh oversampled to a much higher resolution to improve the bandwidth of the measured impulse responses.
Download Full mesh warping techniques This paper discusses methods for the elimination of dispersion in a digital waveguide mesh. As in previous methods, a highly isotropic waveguide mesh is chosen as a starting point, reducing the problem to compensation of frequency-dependent dispersion. For this purpose, as an alternative to Savioja and Välimäki’s technique of frequency-warping the input/output signals, we propose (1) inhomogeneous allpass-warping of delay elements, which enables use of allpass filters without introducing delay-free loops, and (2) “mass loading” the mesh in such a way that high-frequency propagation speed is increased to partially compensate dispersion due to quantization over a grid.
Download Using the waveguide mesh in modelling 3D resonators Most of the results found by several researchers, during these years, in physical modelling of two dimensional (2D) resonators by means of waveguide meshes, extend without too much difficulty to the three dimensional (3D) case. Important parameters such as the dispersion error, the spatial bandwidth, and the sampling efficiency, which characterize the behavior and the performance of a waveguide mesh, can be reformulated in the 3D case, giving the possibility to design mesh geometries supported by a consistent theory. A comparison between different geometries can be carried out in a theoretical context. Here, we emphasize the use of the waveguide meshes as efficient tools for the analysis of resonances in 3D resonators of various shapes. For this purpose, several mesh geometries have been implemented into an application running on a PC, provided with a graphical interface that allows an easy input of the parameters and a simple observation of the consequent system evolution and the output data. This application is especially expected to give information on the modes resonating in generic 3D shapes, where a theoretical prediction of the modal frequencies is hard to do.
Download A method for spectrum separation and envelope estimation of the residual in spectrum modeling of musical sound We propose an original technique for separating the spectrum of the noisy residual component from that of the harmonic, quasideterministic one, and to estimate the envelope of the residual, for the spectrum modeling of musical sounds. The algorithm for spectrum separation relies on nonlinear transformations of the amplitude spectrum of the sampled signal (obtained via FFT), which allow to eliminate the dominant partials without the need for precisely tuned notch filters. The envelope estimation is performed by calculating the energy of the signal in the frequency domain, over a sliding time window. Eventually the residual can be obtained by combining its spectrum and envelope, so that separate processing can be performed on the two.
Download Transformation of instrumental sound related noise by means of adaptive filtering tecniques In this work we present an extension of the classic schema of a time-varying filter excited with white noise for the modeling of noise signals from musical instrument sounds. The framework used is that of statistical signal processing, and a structure that combines an Autoregressive (AR) model with an adaptive FIR filter is proposed. A combbased structure for the AR filter is used when tuned noise is to be modeled. The analysis/resynthesis schema proposed is used to perform some basic sound transformations such as time stretching, tuning and energy envelop control, and spectral processing.
Download Realization of Vold-Kalman tracking filter. A least squares problem The aim of this work was the implementation of the so-called VoldKalman filter. This filter was introduced by Vold and Leuridan in 1993 [1], it is a heterodyne filter for tracking the sinusoidal components of a noisy signal. The formulation of the Vold-Kalman filter leads to a least squares problem. The great advantage of this time-varying filter is that all sinusoids of a signal can be extracted simultaneously yielding a suppression of beating phenomena of close or crossing frequency trajectories. In this paper, we propose a realization for offline processing using the preconditioned conjugate gradient method. Furthermore, we present trivial expressions for the bandwidth and the transition time of first and second order filters.
Download A Java framework for FX development This paper describes the first version of a Java archive (a term basically equivalent to ‘library’ in other programming languages) that has been developed and made available as public domain software for the benefit of the DAFX community, and the COSTG6 web pages in particular. The library is available both as source code and ready to run bytecodes. The archive defines an easy to use set of classes that are modelled after an effects processor. Ready made classes like ‘Effect’, ‘Page of Parameters’, ‘Integer Range Parameter’, ‘Real Range Parameter’, etc. serve as a basis to implement effects and share them. Effects can run from web pages or as stand alone applications, sharing unified look and feel in a platform independent graphical user interface. The programmer only needs to specify the parameters the effect will use, and the method (function) that will apply the effect to each new sample. An automatic GUI interface is created, that enables the adjustment of parameters as well as the specification of input and output files to be used during processing. Developing Java audio effects according to the proposed scheme will allow transparent integration into more complex multiband and multieffect architectures that will be added on a second version of the archive.