Download Parameter Estimation via Differentiable Modal Plate Synthesis We present our submission to Task A of the 1st DAFx Parameter Estimation Challenge, which concerns the estimation of the physical parameters of a vibrating plate from a synthetic impulse response. Our approach introduces a differentiable modal plate synthesizer and estimates the plate parameters through inference-time gradient-based optimization of the synthesizer parameters. The six target parameters are recovered by minimizing a multi-scale spectral loss via backpropagation through the differentiable plate model. To handle the non-convexity of the loss landscape, we adopt a two-phase training strategy consisting of multiple short-term probe optimizations, followed by full-scale refinement initialized from the best candidate. We evaluate the approach on eight impulse responses synthesized with the official challenge dataset generator. Compared with a constant-value predictor and the particle swarm optimization baseline provided by the challenge, the proposed method reduces the prediction error by approximately one order of magnitude.
Download Robust Design of Very High-Order Allpass Dispersion Filters A nonparametric allpass filter design method is presented for matching a desired group delay as a function of frequency. The technique is useful in physical modeling synthesis of musical instruments and emulation of audio effects devices exhibiting dispersive wave propagation. While current group delay filter design methods suffer from numerical difficulties except at low filter orders, the technique presented here is numerically robust, producing an allpass filter in cascaded biquad form, and with the filter poles following a smooth loop within the unit circle. The technique was inspired by the observation that a pole-zero pair arranged in allpass form contributes exactly 2π radians to the integral of group delay around the unit circle, regardless of the (stable) pole location. To match a given group delay characteristic, the method divides the frequency axis into sections containing 2π total area under the desired group-delay curve, and assigns a polezero allpass pair to each. In this way, the method incorporates an order selection technique, and by adding a pure delay to the desired group delay, allows the trading of increased filter order for improved fit to the frequency-dependent group delay. Design examples are given for modeling the group delay of a dispersive string (such as a piano string), and a dispersive spring, such as in a spring reverberator.
Download Flexible software framework for modal synthesis Modal synthesis is an important area of physical modeling whose exploration in the past has been held back by a large number of control parameters, the scarcity of generalpurpose design tools and the difficulty of obtaining the computational power required for real-time synthesis. This paper presents an overview of a flexible software framework facilitating the design and control of instruments based on modal synthesis. The framework is designed as a hierarchy of polymorphic synthesis objects, representing modal structures of various complexity. As a method of generalizing all interactions among the elements of a modal system, an abstract notion of energy is introduced, and a set of energy transfer functions is provided. Such abstraction leads to a design where the dynamics of interactions can be largely separated from the specifics of particular modal structures, yielding an easily configurable and expandable system. A real-time version of the framework has been implemented as a set of C++ classes along with an integrating shell and a GUI, and is currently being used to design and play modal instruments, as well as to survey fundamental properties of various modal algorithms.
Download Using the waveguide mesh in modelling 3D resonators Most of the results found by several researchers, during these years, in physical modelling of two dimensional (2D) resonators by means of waveguide meshes, extend without too much difficulty to the three dimensional (3D) case. Important parameters such as the dispersion error, the spatial bandwidth, and the sampling efficiency, which characterize the behavior and the performance of a waveguide mesh, can be reformulated in the 3D case, giving the possibility to design mesh geometries supported by a consistent theory. A comparison between different geometries can be carried out in a theoretical context. Here, we emphasize the use of the waveguide meshes as efficient tools for the analysis of resonances in 3D resonators of various shapes. For this purpose, several mesh geometries have been implemented into an application running on a PC, provided with a graphical interface that allows an easy input of the parameters and a simple observation of the consequent system evolution and the output data. This application is especially expected to give information on the modes resonating in generic 3D shapes, where a theoretical prediction of the modal frequencies is hard to do.
Download Direct simulation for wind instrument synthesis There are now a number of methods available for generating synthetic sound based on physical models of wind instruments, including digital waveguides, wave digital filters, impedance-based methods and those involving impulse responses. Normally such methods are used to simulate the behaviour of the resonator, and the coupling to the excitation mechanism is carried out by making use of simple lumped finite difference schemes or digital filter structures. In almost all cases, a traveling wave, frequencydomain, or impulse response description of the resonator is used as a starting point—efficient structures may be arrived at when the bore is of a particularly simple form, such as a cylinder or cone. In recent years, however, due to the great computing power available, efficiency has become less of a concern—this is especially the case for musical instruments which may be well-modelled in 1D, such as wind instruments. In this paper, a fully time-space discrete algorithm for the simulation and synthesis of woodwind instrument sounds is presented; such a method, though somewhat more computationally intensive than an efficient waveguide structure, is still well within the realm of real-time performance. The main benefits of such a method are its generality (it is no longer necessary to make any assumptions about bore profile, which may be handled in an almost trivial manner), extensibility (i.e., the model may be generalized to handle nonlinear phenomena directly), ease of programming, and the possibility of direct proofs of numerical stability without invoking frequency domain concepts. Simulation results, sound examples and a graphical user interface, in the Matlab programming language are also presented.
Download Acoustic Modelling of a Convex Pipe Adapted for Digital Waveguide Simulation This work deals with the physical modelling of acoustic pipes for real-time simulation, using the “Digital Waveguide Network” approach and the horn equation. With this approach, a piece of pipe is represented by a two-port system with a loop which involves two delays for wave propagation, and some subsystems without internal delay. A well-known form of this system is the “Kelly-Lochbaum” framework, which allows the reduction of the computation complexity. We focus this work on the simulation of pipes with a convex profile. But, using the “Kelly-Lochbaum” framework with the horn equation, two problems occur: first, even if the outputs are bound, some substates have their values which diverge; second, there is an infinite number of such substates. The system is then unstable and cannot be simulated as such. The solution of this problem is obtained with two steps. First, we show that there is a simple standard form compatible with the “Waveguide” approach, for which there is an infinite number of solutions which preserve the input/output relations. Second, we look for one solution which guarantees the stability of the system and which makes easier the approximation in order to get a low-cost simulation.
Download A Framework for Sonification of Vicon Motion Capture Data This paper describes experiments on sonifying data obtained using the VICON motion capture system. The main goal is to build the necessary infrastructure in order to be able to map motion parameters of the human body to sound. For sonification the following three software frameworks were used: Marsyas, traditionally used for music information retrieval with audio analysis and synthesis, CHUCK, an on-the-fly real-time synthesis language, and Synthesis Toolkit (STK), a toolkit for sound synthesis that includes many physical models of instruments and sounds. An interesting possibility is the use of motion capture data to control parameters of digital audio effects. In order to experiment with the system, different types of motion data were collected. These include traditional performance on musical instruments, acting out emotions as well as data from individuals having impairments in sensor motor coordination. Rhythmic motion (i.e. walking) although complex, can be highly periodic and maps quite naturally to sound. We hope that this work will eventually assist patients in identifying and correcting problems related to motor coordination through sound.
Download A Highly Parametrized Scattering Delay Network Implementation for Interactive Room Auralization Scattering Delay Networks (SDNs) are an interesting approach to artificial reverberation, with parameters tied to the room’s physical properties and the computational efficiency of delay networks. This paper presents a highly-parametrized and real-time plugin of an SDN. The SDN plugin allows for interactive room auralization, enabling users to modify the parameters affecting the reverberation in real-time. These parameters include source and receiver positions, room shape and size, and wall absorption properties. This makes our plugin suitable for applications that require realtime and interactive spatial audio rendering, such as virtual or augmented reality frameworks and video games. Additionally, the main contributions of this work include a filter design method for wall sound absorption, as well as plugin features such as air absorption modeling, various output formats (mono, stereo, binaural, and first to fifth order Ambisonics), open sound control (OSC) for controlling source and receiver parameters, and a graphical user interface (GUI). Evaluation tests showed that the reverberation time and the filter design approach are consistent with both theoretical references and real-world measurements. Finally, performance analysis indicated that the SDN plugin requires minimal computational resources.
Download Onset Time Estimation for the Analysis of Percussive Sounds using Exponentially Damped Sinusoids Exponentially damped sinusoids (EDS) model-based analysis of sound signals often requires a precise estimation of initial amplitudes and phases of the components found in the sound, on top of a good estimation of their frequencies and damping. This can be of the utmost importance in many applications such as high-quality re-synthesis or identification of structural properties of sound generators (e.g. a physical coupling of vibrating devices). Therefore, in those specific applications, an accurate estimation of the onset time is required. In this paper we present a two-step onset time estimation procedure designed for that purpose. It consists of a “rough" estimation using an STFT-based method followed by a time-domain method to “refine" the previous results. Tests carried out on synthetic signals show that it is possible to estimate onset times with errors as small as 0.2ms. These tests also confirm that operating first in the frequency domain and then in the time domain allows to reach a better resolution vs. speed compromise than using only one frequency-based or one time-based onset detection method. Finally, experiments on real sounds (plucked strings and actual percussions) illustrate how well this method performs in more realistic situations.
Download Cross-Modal Variational Inference for Bijective Signal-Symbol Translation Extraction of symbolic information from signals is an active field of research enabling numerous applications especially in the Musical Information Retrieval domain. This complex task, that is also related to other topics such as pitch extraction or instrument recognition, is a demanding subject that gave birth to numerous approaches, mostly based on advanced signal processing-based algorithms. However, these techniques are often non-generic, allowing the extraction of definite physical properties of the signal (pitch, octave), but not allowing arbitrary vocabularies or more general annotations. On top of that, these techniques are one-sided, meaning that they can extract symbolic data from an audio signal, but cannot perform the reverse process and make symbol-to-signal generation. In this paper, we propose an bijective approach for signal/symbol translation by turning this problem into a density estimation task over signal and symbolic domains, considered both as related random variables. We estimate this joint distribution with two different variational auto-encoders, one for each domain, whose inner representations are forced to match with an additive constraint, allowing both models to learn and generate separately while allowing signal-to-symbol and symbol-to-signal inference. In this article, we test our models on pitch, octave and dynamics symbols, which comprise a fundamental step towards music transcription and label-constrained audio generation. In addition to its versatility, this system is rather light during training and generation while allowing several interesting creative uses that we outline at the end of the article.