Download Hyper-Dimensional Digital Waveguide Mesh for Reverberation Modeling Characteristics of digital waveguide meshes with more than three physical dimensions are studied. Especially, the properties of a 4-D mesh are analyzed and compared to waveguide structures of lower dimensionalities. The hypermesh produces a response with a dense and irregular modal pattern at high frequencies, which is beneficial in modeling the reverberation of rooms or musical instrument bodies. In addition, it offers a high degree of decorrelation between output points selected at different locations, which is advantageous for multi-channel reverberation. The frequencydependent decay of the hypermesh response can be controlled using boundary filters introduced recently by one of the authors. Several hypermeshes can be effectively combined in a multirate system, in which each mesh produces reverberation on a finite frequency band. The paper presents two hypermesh application examples: the modeling of the impulse response of a lecture hall and the simulation of the response of a clavichord soundbox.
Download Two-step modal identification for increased resolution analysis of percussive sounds Modal synthesis is a practical and efficient way to model sounding structures with strong resonances. In order to create realistic sounds, one has to be able to extract the parameters of this model from recorded sounds produced by the physical system of interest. Many methods are available to achieve this goal, and most of them require a careful parametrization and a post-selection of the modes to guarantee a good quality/complexity trade-off. This paper introduces a two step analysis method aiming at an automatic and reliable identification of the modes. The first step is performed at a global level with few assumptions about the spectro/temporal content of the considered signal. From the knowledge gained with this global analysis, one can focus on specific frequency regions and perform a local analysis with strong assumptions. The gains of such a two step approach are a better estimation of the number of modal components as well as a better estimate of their parameters.
Download Score based real-time performance with a virtual violin This paper describes the implementation of a violin physical model tied with the control of music scores to enable the real-time performance of music pieces. The violin model is made of four strings, which allows the performance of double stops, chords and specific resonant effects that can be encountered in violin playing. A graphic tablet is used to control the bowing parameters and to trigger automatically note events contained in a specifically formatted MIDI file. The automatic pitch change helps reducing the violin playing complexity and enables the user to focus on sound shaping and phrasing. The device can be used for pure sound synthesis purposes as well as for experiments related to violinists’ sound control. However, the simplified interface for sound and score events is particularly suitable for non violinists wishing to explore expressive capabilities of the instrument and to experience specific features of violin playing.
Download Tunable Collisions: Hammer-String Simulation with Time-Variant Parameters In physical modelling synthesis, articulation and tuning are effected via time-variation in one or more parameters. Adopting hammered strings as a test case, this paper develops extended forms of such control, proposing a numerical formulation that affords online adjustment of each of its scaled-form parameters, including those featuring in the one-sided power law for modelling hammerstring collisions. Starting from a modally-expanded representation of the string, an explicit scheme is constructed based on quadratising the contact energy. Compared to the case of time-invariant contact parameters, updating the scheme’s state variables relies on the evaluation of two additional analytic partial derivatives of the auxiliary variable. A numerical energy balance is derived and the numerical contact force is shown to be strictly non-adhesive. Example results with time-variant tension and time-variant contact stiffness are detailed, and real-time viability is demonstrated.
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 Efficient simulation of the yaybahar using a modal approach This work presents a physical model of the yaybahar, a recently invented acoustic instrument. Here, output from a bowed string is passed through a long spring, before being amplified and propagated in air via a membrane. The highly dispersive character of the spring is responsible for the typical synthetic tonal quality of this instrument. Building on previous literature, this work presents a modal discretisation of the full system, with fine control over frequency-dependent decay times, modal amplitudes and frequencies, all essential for an accurate simulation of the dispersive characteristics of reverberation. The string-bow-bridge system is also solved in the modal domain, using recently developed noniterative numerical methods allowing for efficient simulation.
Download Non-iterative Modal Parameter Estimation for Plate Reverbs via Matrix-Pencil-Guided State Space Model Initialization Modal parameter identification for plate reverbs remains a challenging problem in virtual-analog audio effect emulation. Though neural network-based black-box approaches achieve high modeling accuracy, they generally lack interpretability and do not provide access to physically meaningful modal parameters. In this work, we present our solution to Task B of the DAFx Plate Reverb Parameter Estimation Challenge. Our method first estimates the total number of modes and then employs a Matrix Pencil (MP)-guided eigenvalue initialization strategy for a diagonal complex-valued State Space Model (SSM), which can be interpreted as a bank of parallel second-order all-pole filters. Exploiting the linearity of the resulting system, we compute the state impulse responses and replace gradient-based optimization with a closed-form least-squares estimation of the modal gains. The proposed approach enables accurate recovery of the modal parameters while maintaining an interpretable system representation.
Download Extraction of the excitation point location on a string using weighted least-square estimation of a comb filter delay This paper focuses on the extraction of the excitation point location on a guitar string by an iterative estimation of the structural parameters of the spectral envelope. We propose a general method to estimate the plucking point location, working into two stages: starting from a measure related to the autocorrelation of the signal as a first approximation, a weighted least-square estimation is used to refine a FIR comb filter delay value to better fit the measured spectral envelope. This method is based on the fact that, in a simple digital physical model of a plucked-string instrument, the resonant modes translate into an all-pole structure while the initial conditions (a triangular shape for the string and a zero-velocity at all points) result in a FIR comb filter structure.
Download Simulation-based Inference Plate Reverberation Inverse Problems We address Task A of the 1st DAFx Parameter Estimation Challenge, which aims to retrieve the physical parameters of a plate model from an impulse response. To do so, we use the Simulation-Based Inference (SBI) framework, in which we train a neural network to estimate a density over plate parameters given an impulse response, using a dataset generated by the simulator. Inference for a new impulse response then requires only a forward pass through the network, without involving the simulator. For each test observation, we fine-tune a specific network: additional simulation rounds are performed by sampling parameters from the current estimated distribution, simulating the corresponding impulse responses, and fine-tuning to produce the specialized network.
Download Synthetic Transaural Audio Rendering (STAR): a Perceptive Approach for Sound Spatialization The principles of Synthetic Transaural Audio Rendering (STAR) were first introduced at DAFx-06. This is a perceptive approach for sound spatialization, whereas state-of-the-art methods are rather physical. With our STAR method, we focus neither on the wave field (such as HOA) nor on the sound wave (such as VBAP), but rather on the acoustic paths traveled by the sound to the listener ears. The STAR method consists in canceling the cross-talk signals between two loudspeakers and the ears of the listener (in a transaural way), with acoustic paths not measured but computed by some model (thus synthetic). Our model is based on perceptive cues, used by the human auditory system for sound localization. The aim is to give the listener the sensation of the position of each source, and not to reconstruct the corresponding acoustic wave or field. This should work with various loudspeaker configurations, with a large sweet spot, since the model is neither specialized for a specific configuration nor individualized for a specific listener. Experimental tests have been conducted in 2015 and 2019 with different rooms and audiences, for still, moving, and polyphonic musical sounds. It turns out that the proposed method is competitive with the state-of-the-art ones. However, this is a work in progress and further work is needed to improve the quality.