Paper Archive

This paper proposes a new macroscopic physical model of ferromagnetic coils used in audio circuits. To account for realistic saturation and hysteretic phenomena, this model combines statistical physics results, measurement-driven refinements and portHamiltonian formulations that guarantee passivity, thermodynamic consistency and composability according to both electric and thermal ports. As an illustration, the model is used to simulate a passive high-pass filter. Different types of audio inputs are considered and simulations are compared to measurements.

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Physical accuracy of virtual acoustics receives increasing attention due to renewed interest in virtual and augmented reality applications. So far, the modeling of vibrating objects as point sources is a common simplification which neglects effects caused by their spatial extent. In this contribution, we propose a technique for the interconnection of a distributed source to a room model, based on a modal representation of source and room. In particular, we derive a connection matrix that describes the coupling between the modes of the source and the room modes in an analytical form. Therefore, we consider the example of a string that is oscillating in a room. Both, room and string rely on well established physical descriptions that are modeled in terms of transfer functions. The derived connection of string and room defines the coupling between the characteristic string and room modes. The proposed structure is analyzed by numerical evaluations and sound examples on the supplementary website.

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In the design of real-time spring reverberation algorithms, a modal architecture offers several advantages, including computational efficiency and parametric control flexibility. Due to the complex, highly dispersive behavior of helical springs, computing physically accurate parameters for such a model presents specific challenges. In this paper these are addressed by applying an implicit higher-order-in-space finite difference scheme to a two-variable model of helical spring dynamics. A novel numerical boundary treatment is presented, which utilises multiple centered boundary expressions of different stencil width. The resulting scheme is unconditionally stable, and as such allows adjusting the numerical parameters independently of each other and of the physical parameters. The dispersion relation of the scheme is shown to be accurate in the audio range only for very high orders of accuracy in combination with a small temporal and spatial step. The frequency, amplitude, and decay rate of the system modes are extracted from a diagonalised form of this numerical model. After removing all modes with frequencies outside the audio range and applying a modal amplitude correction to compensate for omitting the magnetic transducers, a light-weight modal reverb algorithm is obtained. Comparison with a measured impulse response shows a reasonably good match across a wide frequency range in terms of echo density, decay characteristics, and diffusive nature.

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Slide-string instruments allow continuous control of pitch by articulation with a slide object whose position of contact with the string is time-varying. This paper presents a method for simulation of such articulation. Taking into account sensing and musical practice considerations, an appropriate physical model configuration is determined, which is then formulated in numerical form using a finite difference approach. The model simulates the attachment and detachment phases of slide articulation which generally involve rattling, while finger damping is modelled in a more phenomenological manner as a regionally induced time-varying damping. A stability bound for the numerical model is provided via energy analysis, which also reveals the driving power contributions of the separate articulatory sources. The approach is exemplified with simulations of slide articulatory gestures that involve glissando, vibrato and finger damping.

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This paper presents a novel method of analysing drum sounds, demonstrating that this can form the basis of a highly realistic synthesis technique for real-time use. The synthesis method can be viewed as an extension of IFFT synthesis; here we exploit the fact that audio signals can be recovered from solely the real component of their discrete Fourier transform (RDFT). All characteristics of an entire drum sample can therefore be conveniently encoded in a single, real-valued, frequency domain signal. These signals are interpreted, incorporating the physics of the instrument, and modelled to investigate how the perceptual features are encoded. The model was able to synthesize drum sound components in such detail that they could not be distinguished in an ABX test. This method may therefore be capable of outperforming existing synthesis techniques, in terms of realism. Sound examples available here.

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Feedback delay network reverberators have decay filters associated with each delay line to model the frequency dependent reverberation time (T60) of a space. The decay filters are typically designed such that all delay lines independently produce the same T60 frequency response. However, in real rooms, there are multiple, concurrent T60 responses that depend on the geometry and physical properties of the materials present in the rooms. In this paper, we propose the Grouped Feedback Delay Network (GFDN), where groups of delay lines share different target T60s. We use the GFDN to simulate coupled rooms, where one room is significantly larger than the other. We also simulate rooms with different materials, with unique decay filters associated with each delay line group, designed to represent the T60 characteristics of a particular material. The T60 filters are designed to emulate the materials’ absorption characteristics with minimal computation. We discuss the design of the mixing matrix to control inter- and intra-group mixing, and show how the amount of mixing affects behavior of the room modes. Finally, we discuss the inclusion of air absorption filters on each delay line and physically motivated room resizing techniques with the GFDN.

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We present a method for accurately synthesizing the acoustic response of a water bottle using modal signal processing. We start with extensive measurements of two water bottles with considerations for how the level of water inside the bottles, the area covered by stickers attached to the exterior of the bottles, and the method of striking the bottles affect their sound. We perform modal analysis of these measurements and implement a real-time modal water bottle synthesizer.

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This paper describes a computationally efficient synthesis model for snare drum sounds. Its parameters can be modulated at audio rate while being played. The input to the model is an acoustic excitation signal which carries spectral information to color the output sound. This makes it suitable for acoustic interfaces – devices which provide excitation signal and control data simultaneously. The presented synthesis model builds up on work done by Miller Puckette and processes audio input from a piezoelectric microphone into a nonlinear reverberator. This paper details a strikingly simple but novel approach on how to make use of the momentary DC offset generated by piezoelectric microphones when pressed to simulate the changes in drumhead tension. This technique is especially of interest for interfaces without pressure sensing capabilities. In the design process we pursued an experimental approach rather than a purely mathematical. Implementations of the synthesis model are provided for Pure Data and FAUST as open source.

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A mechanical system is said to be bistable when its moving parts can rest at two equilibrium positions. The aim of this work is to model the vibration behaviour of a bistable system and use it to create a sound effect, taking advantage of the nonlinearities that characterize such systems. The velocity signal of the bistable system excited by an audio signal is the output of the digital effect. The latter is coded in C++ language and compiled into VST3 format that can be run as an audio plugin within most of the commercial digital audio workstation software in the market and as a standalone application. A Web Audio API demonstration is also available online as a support material.

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Biquad filters are a common tool for filter design. In this writing, we develop two structures for creating biquad filters with nonlinear elements. We provide conditions for the guaranteed stability of the nonlinear filters, and derive expressions for instantaneous pole analysis. Finally, we examine example filters built with these nonlinear structures, and show how the first nonlinear structure can be used in the context of analog modelling.

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