Download Revisiting Implicit Finite Difference Schemes for Three-Dimensional Room Acoustics Simulations on GPU Implicit finite difference schemes for the 3-D wave equation using a 27-point stencil on the cubic grid are presented, for use in room acoustics modelling and artificial reverberation. The system of equations that arises from the implicit formulation is solved using the Jacobi iterative method. Numerical dispersion is analysed and computational efficiency is compared to second-order accurate 27-point explicit schemes. Timing results from GPU implementations demonstrate that the proposed algorithms scale over their explicit counterparts as expected: by a factor of M + 2, where M is a fixed number of Jacobi iterations (eight can be sufficient in single precision). Thus, the accuracy of the approximation can be improved over explicit counterparts with only a linear increase in computational costs, rather than the quartic (in operations) and cubic (in memory) increases incurred when oversampling the grid. These implicit schemes are advantageous in situations where less than 1% dispersion error is desired.
Download A Preliminary Model for the Synthesis of Source Spaciousness We present here a basic model for the synthesis of source spaciousness over loudspeaker arrays. This model is based on two experiments carried out to quantify the contribution of early reflections and reverberation to the perception of source spaciousness.
Download Low Frequency Group Delay Equalization of Vented Boxes using Digital Correction Filters In this paper methods to determine the group delay of vented boxes and techniques for the design of filters for group delay equalization are presented. First the transfer function and the related group delay are explained. Then it is shown how the group delay can be computed or approximated for a certain alignment of the box. Furthermore it is shown how to derive the required parameters of the transfer function from a simple electrical measurement of the box, which allows the determination of the group delay without knowledge of the box design parameters. Two strategies for the design and implementation of digital correction filters are shown where one approach allows for a real-time adjustability of the delay. Finally, the performance with a real speaker is evaluated.
Download Exploring the Vectored Time Variant Comb Filter This paper presents the time variant vectored comb filter. It is an extension of the feedback delay network to time variant and nonlinear domains. Effects such as chorus and flanger, tap delay and pitch shifter are examined in the context of the feedback scheme. Efficient implementation of a stateless vectorizable LFO for modulation purposes is presented, along with a recursive formulation of the Hadamard matrix multiplication. The time variant comb filter is examined in various effect settings, and presented with source code and sound examples.
Download Time-Varying Filters for Musical Applications A variety of methods are available for implementing time-varying digital filters for musical applications. The considerations for musical applications differ from those of other applications, such as speech coding. This domain requires realtime parametric control of a filter such as an equalizer, allowing parameters to vary each sample, e.g. by user interaction, a low-frequency oscillator (LFO), or an envelope. It is desirable to find a filter structure that is timevarying stable, artifact-free, computationally efficient, easily supports arbitrary filter shapes, and yields sensible intermediate filter shapes when interpolating coefficients. It is proposed to use the state variable filter (SVF) for this purpose. A novel proof of its stable time-varying behavior is presented. Equations are derived for matching common equalizer filter shapes, as well as any zdomain transfer function, making the SVF suitable for efficiently implementing any recursive filter. The SVF is compared to state of the art filter structures in an objective evaluation and a subjective listening test. The results confirm that the SVF has good audio quality, while supporting the aforementioned advantageous qualities in a time-varying digital filter for music. They also show that a class of time-varying filter techniques useful for speech coding are unsuitable for musical applications.
Download Perceptual Linear Filters: Low-Order ARMA Approximation for Sound Synthesis This paper deals with the approximation of a given frequency response by a low-order linear ARMA filter (Auto-Regressive Moving Average). The aim of this work is the audio synthesis, then to improve the perceptual quality, a criterion based on human listening is defined and minimized. Two complementary approaches are proposed here for solving this non-linear and non-convex problem: first, a weighted version of the Iterative Prefiltering, second, an adaptation of the Gauss-Newton method. This algorithm is adapted to guarantee the causality/stability of the obtained filter, and eventually its minimum phase property. The benefit of the new method is illustrated and evaluated.
Download Approximations for Online Computation of Redressed Frequency Warped Vocoders In recent work, the construction of non-uniform generalized Gabor frames for the time-frequency analysis of signals has been introduced. In particular, while preserving perfect reconstruction, these frames allow for tilings of the time-frequency plane with arbitrary allocation of partially overlapping frequency bands or time intervals. In a recent paper, the author demonstrated that the construction of such frames can be entirely based on warping operators, which are specified by the required frequency or time warping maps, which, in turn, interpolate the desired frequency or time intervals edges. However, while the online computation of Gabor expansions on non-uniform time intervals presents little or no problem, the computation of Gabor expansions on non-uniform frequency bands requires knowledge of the Fourier transform of the entire signal, which precludes online computation. In this paper we introduce approximations and ideas for the design of nearly perfect reconstruction analysis and synthesis atoms, which allow for the online computation of time-frequency representations on non-uniform frequency bands.
Download Hybrid Reverberation Processor with Perceptual Control This paper presents a hybrid reverberation processor, i.e. a realtime audio signal processing unit that combines a convolution reverb for recreating the early reflections of a measured impulse response (IR) with a feedback delay network (FDN) for synthesizing the reverberation tail. The FDN is automatically adjusted so as to match the energy decay profile of the measured IR. Particular attention is given to the transition between the convolution section and the FDN in order to avoid audible artifacts. The proposed reverberation processor offers both computational efficiency and flexible perceptual control over the reverberation effect.
Download Examining the Oscillator Waveform Animation Effect An enhancing effect that can be applied to analogue oscillators in subtractive synthesizers is termed Animation, which is an efficient way to create a sound of many closely detuned oscillators playing in unison. This is often referred to as a supersaw oscillator. This paper first explains the operating principle of this effect using a combination of additive and frequency modulation synthesis. The Fourier series will be derived and results will be presented to demonstrate its accuracy. This will then provide new insights into how other more general waveform animation processors can be designed.
Download Multi-Player Microtiming Humanisation using a Multivariate Markov Model In this paper, we present a model for the modulation of multiperformer microtiming variation in musical groups. This is done using a multivariate Markov model, in which the relationship between players is modelled using an interdependence matrix (α) and a multidimensional state transition matrix (S). This method allows us to generate more natural sounding musical sequences due to the reduction of out-of-phase errors that occur in Gaussian pseudorandom and player-independent probabilistic models. We verify this using subjective listening tests, where we demonstrate that our multivariate model is able to outperform commonly used univariate models at producing human-like microtiming variability. Whilst the participants in our study judged the real time sequences performed by humans to be more natural than the proposed model, we were still able to achieve a mean score of 63.39% naturalness, suggesting microtiming interdependence between players captured in our model significantly enhances the humanisation of group musical sequences.