Download The Helmholtz Resonator Tree The Helmholtz resonator is a prototype of a single acoustic resonance, which can be modeled with a digital resonator. This paper extends this concept by coupling several Helmholtz resonators. The resulting structure is called a Helmholtz resonator tree. The height of the tree is defined by the number of resonator layers that are interconnected. The overall number of resonance frequencies of a Helmholtz resonator tree is the same as its height. A Helmholtz resonator tree can be modeled using wave digital filters (WDF), when electro-acoustic analogies are applied. A WDF tool for implementing Helmholtz resonator trees has been developed in C++. A VST plugin and an Android mobile application were created, which can run short Helmholtz resonator trees in real time. Helmholtz resonator trees can be used for the real-time synthesis of percussive sounds and for realizing novel filtering which can be tuned using intuitive physical parameters.
Download A pickup model for the Clavinet In this paper recent findings on magnetic transducers are applied to the analysis and modeling of Clavinet pickups. The Clavinet is a stringed instrument having similarities to the electric guitar, it has magnetic single coil pickups used to transduce the string vibration to an electrical quantity. Data gathered during physical inspection and electrical measurements are used to build a complete model which accounts for nonlinearities in the magnetic flux. The model is inserted in a Digital Waveguide (DWG) model for the Clavinet string for its evaluation.
Download Model-based synthesis and transformation of voiced sounds In this work a glottal model loosely based on the Ishizaka and Flanagan model is proposed, where the number of parameters is drastically reduced. First, the glottal excitation waveform is estimated, together with the vocal tract filter parameters, using inverse filtering techniques. Then the estimated waveform is used in order to identify the nonlinear glottal model, represented by a closedloop configuration of two blocks: a second order resonant filter, tuned with respect to the signal pitch, and a regressor-based functional, whose coefficients are estimated via nonlinear identification techniques. The results show that an accurate identification of real data can be achieved with less than regressors of the nonlinear functional, and that an intuitive control of fundamental features, such as pitch and intensity, is allowed by acting on the physically informed parameters of the model. 10
Download Topologizing Sound Synthesis via Sheaves In recent years, a range of topological methods have emerged for
processing digital signals. In this paper we show how the construction of topological filters via sheaves can be used to topologize
existing sound synthesis methods. I illustrate this process on two
classes of synthesis approaches: (1) based on linear-time invariant digital filters and (2) based on oscillators defined on a circle.
We use the computationally-friendly approach to modeling topologies via a simplicial complex, and we attach our classical synthesis
methods to them via sheaves. In particular, we explore examples
of simplicial topologies that mimic sampled lines and loops. Over
these spaces we realize concrete examples of simple discrete harmonic oscillators (resonant filters), and simple comb filter based
algorithms (such as Karplus-Strong) as well as frequency modulation.
Download Real-Time Implementation of the Dynamic Stiff String Using Finite-Difference Time-Domain Methods and the Dynamic Grid Digital musical instruments based on physical modelling have gained increased popularity over the past years. This is partly due to recent advances in computational power, which allow for their real-time implementation. One of the great potentials for digital musical instruments based on physical models, is that one can go beyond what is physically possible and change properties of the instruments which are static in real life. This paper presents a real-time implementation of the dynamic stiff string using finitedifference time-domain (FDTD) methods. The defining parameters of the string can be varied in real time and change the underlying grid that these methods rely on based on the recently developed dynamic grid method. For most settings, parameter changes are nearly instantaneous and do not cause noticeable artefacts due to changes in the grid. A reliable way to prevent artefacts for all settings is under development.
Download RIR2FDN: An Improved Room Impulse Response Analysis and Synthesis This paper seeks to improve the state-of-the-art in delay-networkbased analysis-synthesis of measured room impulse responses (RIRs). We propose an informed method incorporating improved energy decay estimation and synthesis with an optimized feedback delay network. The performance of the presented method is compared against an end-to-end deep-learning approach. A formal listening test was conducted where participants assessed the similarity of reverberated material across seven distinct RIRs and three different sound sources. The results reveal that the performance of these methods is influenced by both the excitation sounds and the reverberation conditions. Nonetheless, the proposed method consistently demonstrates higher similarity ratings compared to the end-to-end approach across most conditions. However, achieving an indistinguishable synthesis of measured RIRs remains a persistent challenge, underscoring the complexity of this problem. Overall, this work helps improve the sound quality of analysis-based artificial reverberation.
Download Large-scale Real-time Modular Physical Modeling Sound Synthesis Due to recent increases in computational power, physical modeling synthesis is now possible in real time even for relatively complex models. We present here a modular physical modeling instrument design, intended as a construction framework for string- and bar- based instruments, alongside a mechanical network allowing for arbitrary nonlinear interconnection. When multiple nonlinearities are present in a feedback setting, there are two major concerns. One is ensuring numerical stability, which can be approached using an energy-based framework. The other is coping with the computational cost associated with nonlinear solvers—standard iterative methods, such as Newton-Raphson, quickly become a computational bottleneck. Here, such iterative methods are sidestepped using an alternative energy conserving method, allowing for great reduction in computational expense or, alternatively, to real-time performance for very large-scale nonlinear physical modeling synthesis. Simulation and benchmarking results are presented.
Download Simplified Guitar Bridge Model for the Displacement Wave Representation in Digital Waveguides In this paper, we present a simplified model for the string-bridge interaction in guitars or other string instruments simulated by digital waveguides. The bridge model is devised for the displacement wave representation in order to be integrated with other models for string interactions with the player and with other parts of the instrument, whose simulation and implementation is easier in this representation. The model is based on a multiplierless scattering matrix representing the string-bridge interaction. Although not completely physically inspired, we show that this junction is sufficiently general to accommodate a variety of transfer functions under the sole requirement of passivity and avoids integration constants mismatch when the bridge is in turn modeled by a digital waveguide. The model is completed with simple methods to introduce horizontal and vertical polarizations of the string displacement and sympathetic vibrations of other strings. The aim of this paper is not to provide the most general methods for sound synthesis of guitar but, rather, to point at low computational cost and scalable solutions suitable for real-time implementations where the synthesizer is running together with several other audio applications.
Download Live Convolution with Time-variant Impulse Response This paper describes a method for doing convolution of two live signals, without the need to load a time-invariant impulse response (IR) prior to the convolution process. The method is based on stepwise replacement of the IR in a continuously running convolution process. It was developed in the context of creative live electronic music performance, but can be applied to more traditional use cases for convolution as well. The process allows parametrization of the convolution parameters, by way of real-time transformations of the IR, and as such can be used to build parametric convolution effects for audio mixing and spatialization as well.
Download Filtering within the Framework of Mass-Interaction Physical Modeling and of Haptic Gestural Interaction A variety of filters have been designed, synthesized and used in the history of electronic and computer music. All the approaches aimed to provide filters fulfilling several specifications such as frequency response, phase response, transient state characteristics like rise time and overshoot, realizably conditions concerning the technology used for the implementation and even economical considerations. One of the most important aspects concerning the filters dedicated to musical applications is the control structure they provide to the musician, who is in charge for the integration of the filtering operation in the compositional process and performance. Designing filters using the mass interaction scheme embedded in the CORDIS-ANIMA formalism (used for sound synthesis and composition by physical modelling) offers a different methodology in the control which is coherent with the philosophy of musical composition by ‘physical thinking’. This article introduces a technique to design filters using the CORDIS-ANIMA simulation language.