Paper Archive

The synthesis of guitar tones was one of the first uses of physical modeling synthesis, and many approaches (notably digital waveguides) have been employed. The dynamics of the string under playing conditions is complex, and includes nonlinearities, both inherent to the string itself, and due to various collisions with the fretboard, frets and a stopping finger. All lead to important perceptual effects, including pitch glides, rattling against frets, and the ability to play on the harmonics. Numerical simulation of these simultaneous strong nonlinearities is challenging, but recent advances in algorithm design due to invariant energy quadratisation and scalar auxiliary variable methods allow for very efficient and provably numerically stable simulation. A new design is presented here that does not employ costly iterative methods such as the Newton-Raphson method, and for which required linear system solutions are small. As such, this method is suitable for real-time implementation. Simulation and timing results are presented.

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The thin plate is a key structure in various musical instruments, including many percussion instruments and the soundboard of the piano, and also is the mechanism underlying electromechanical plate reverberation. As such, it is a suitable candidate for physical modelling approaches to audio effects and sound synthesis, such as finite difference methods—though great attention must be paid to the problem of numerical dispersion, in the interest of reducing perceptual artefacts. In this paper, we present two finite difference schemes on hexagonal grids for such a thin plate system. Numerical dispersion and computational costs are analysed and compared to the standard 13-point Cartesian scheme. An equivalent finite volume scheme can be related to the 13-point Cartesian scheme and a 19-point hexagonal scheme, allowing for fitted boundary conditions of the clamped type. Theoretical modes for a clamped circular plate are compared to simulations. It is shown that better agreement is obtained for the hexagonal scheme than the Cartesian scheme.

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This article addresses the Doppler effect of a planar vibrating piston in a duct, as a plane wave radiation approximation generated by a loudspeaker membrane. This physical model corresponds to a nonlinear problem, because the linear propagation is excited by a moving boundary condition at the piston face: this introduces a varying propagation time between the piston and a fixed receiver. The existence of a regular function that solves the problem (a socalled “strong” solution) is proven, under a well-posed condition that guarantees that no shock occurs. This function satisfies an implicit equation to be solved. An algorithm based on the perturbation method is proposed, from which an exact solution can be built using power series. The convergence of the power series is numerically checked on several examples. Simulations derived from a truncated power series provide sound examples with audible intermodulation and distortion effects for realistic loudspeaker excursion and speed ranges.

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In this paper a new technique is introduced that allows for the variable step-size simulation of wave digital filters. The technique is based on the preservation of the underlying network variables which prevents fluctuation in the stored energy in reactive network elements when the step-size is changed. This method allows for the step-size variation of wave digital filters discretized with any passive discretization technique and works with both linear and nonlinear reference circuits. The usefulness of the technique with regards to audio circuit simulation is demonstrated via the case study of a relaxation oscillator where it is shown how the variable step-size technique can be used to mitigate frequency error that would otherwise occur with a fixed step-size simulation. Additionally, an example of how aliasing suppression techniques can be combined with physical modeling is given with an example of the polyBLEP antialiasing technique being applied to the output voltage signal of the relaxation oscillator.

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Processing and recording multitrack audio from an electronic drum set is demanding of computational power and hardware resources. In this paper, we present a complete musical instrument system capable of up to 16-channel percussion sampling, processing, and recording, all in real time. The system leverages a field programmable gate array (FPGA) for parallel audio processing and includes audio effects such as pitch shift, delay, reverb, distortion, a virtual analog low-pass filter, and bit crush. The FPGA also provides interfaces for other system hardware, including an Ethernet audio interface and various audio effect control interfaces. The final design has a cost of under $500 and utilizes about half of the hardware resources on an entry-level FPGA, providing a future platform for more advanced percussion synthesis using real-time physical modeling.

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We present two Transformer-based methods for Task A of the 1st DAFx Parameter Estimation Challenge, which requires estimating six effective physical parameters of a synthetic plate-reverb model from its impulse response (IR). Method A1 combines an Audio Spectrogram Transformer encoder and Transformer regressor with differentiable IR refinement. Method A2 uses the same encoder to condition a continuous normalizing flow and refines sampled candidates using particle swarm optimisation (PSO) and gradient polishing. Both methods preserve the absolute IR scale to recover surface density. On a synthetic holdout set of 100 IRs, both refinement procedures reduce waveform and parameter errors by more than three orders of magnitude relative to the unrefined neural outputs. The PSO-based pipeline achieves the lowest errors, indicating near-perfect recovery in this matched synthetic setting.

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The modeling of room acoustics and simulation of sound wave propagation remain a difficult and computationally expensive task. Two main techniques have evolved, with one focusing on a real physical - wave-oriented - sound propagation, while the other approximates sound waves as rays using raytracing techniques. Due to many advances in computer science, and especially computer graphics over the last decade, interactive 3D sound simulations for complex and dynamic environments are within reach. In this paper we analyze sound propagation in terms of acoustic energy and explore the possibilities to map these concepts to radiometry and graphics rendering equations. Although we concentrate on ray-based techniques, we also partially consider wavebased sound propagation effects. The implemented system exploits modern graphics hardware and rendering techniques and is able to efficiently simulate 3D room acoustics, as well as to measure simplified personal HRTFs through acoustic raytracing.

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Frequency warping is a modifier that acts on sound signals by remapping the frequency axis. Thus, the spectral content of the original sound is displaced to other frequencies. At the same time, the phase relationship among the signal components is altered, nonlinearly with respect to frequency. While this effect is interesting and has several applications, including in the synthesis by physical models, its use has been so far limited by the lack of an accurate and flexible real-time algorithm. In this paper we present methods for frequency warping that are based on local approximations of the warping operators and allow for real-time implementation. Filter bank structures are derived that allow for efficient realization of the approximate technique. An analysis of the error is also presented, which shows that both numerical and perceptual errors are within acceptable limits. Furthermore, the approximate implementation allows for a larger variety of warping maps than that achieved by the classical (non-causal) first-order allpass cascade implementation.

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The rich spectra of classic waveforms (sawtooth, square and triangular) are obtained by discontinuities in the waveforms or their derivatives. At the same time, the discontinuities lead to aliasing when the waveforms are digitally generated. To remove or reduce the aliasing, researchers have proposed various methods, mostly based on limiting bandwidth or smoothing the waveforms. This paper introduces a new approach to generate the virtual analog oscillators with no aliasing. The approach relies on generating an impulse train using a feedback delay loop, often used for the physical modeling of musical instruments. Classic waveforms are then derived from the impulse train with a leaky integrator. Although the output generated by this method is not exactly periodic, it perceptually sounds harmonic. While additional processing is required for time-varying pitch shifting, resulting in some high-frequency attenuation when the pitch changes, the proposed method is computationally more efficient than other algorithms and the high-frequency attenuation can be also adjusted.

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This paper presents an automated Public Address processing unit, using delay and magnitude response adjustment. The aim is to achieve a flat frequency response and delay adjustment between different physically-placed speakers at the measuring point, which is nowadays usually made manually by the sound technician. The adjustment is obtained using three signal processing operations to the audio signal: time delay adjustment, crossover filtering, and graphic equalization. The automation is in the calculation of different parameter sets: estimation of the time delay, the selection of a suitable crossover frequency, and calculation of the gains for a third-octave graphic equalizer. These automatic methods reduce time and effort in the calibration of line-array PA systems, since only three sine sweeps must be played through the sound system. Measurements have been conducted in an anechoic chamber using a 1:10 scale model of a line array system to verify the functioning of the automatic calibration and equalization methods.

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