Paper Archive

When a classical guitar string is plucked, vibration energy flows through the bridge into the body and is radiated as sound. Synthesising this process for a large collection of instruments requires both an efficient nonlinear string model and a robust method for extracting instrument-specific parameters from measurements. This paper addresses both issues. Starting from the publicly available dataset of Mores, which provides impulse-response measurements on 65 classical guitars, modal parameters of the bridge compliance and of the bridge-to-air radiation path are extracted for each instrument. These feed a nonlinear string model in which transverse vibration is governed by a geometrically exact elastic potential coupled at an interior bridge point to the measured body data. The nonlinear potential is quadratised via the Scalar Auxiliary Variable (SAV) method, so that the equations of motion become linear in a scalar variable and a known gradient vector, even at the continuous level. After time discretisation, the coupled system is inverted through two sequential Sherman–Morrison rank-one updates (one for the bridge coupling, one for the SAV nonlinearity), yielding an O(N) algorithm per time step. Two regularisation techniques prevent long-term drift of the auxiliary variable. The complete pipeline is demonstrated by synthesising plucked notes across all frets and strings for each of the 65 guitars.

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This paper concerns the implementation of a 3D transmission line matrix (TLM) algorithm based on a tetrahedral mesh structure and visualization of room acoustics simulation. Although a well known method, TLM algorithms implemented in 3D are less commonly found in the literature. We have implemented the TLM method using a tetrahedral mesh of pressure nodes with transmission lines lying superimposed on nearest neighbour bonds of a tetrahedral atomic lattice. Results of simulations are compared with those of a standard 3D cartesian mesh and a 2D mesh implementation of TLM. An important feature is a useful graphics interface designed for user-friendly control of room acoustics simulation and visualization in arbitrary shaped rooms containing objects of arbitrary size and number. The paper includes brief discussions of results of using different techniques for modeling totally absorptive or partially absorptive boundaries.

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In this paper, we present a sound system to be integrated in an accredited realistic full-flight simulator, used for the training of airline pilots. We discuss the design and implementation of a corresponding real-time signal-processing software providing threedimensional audio reproduction of the acoustic events on a flight deck. Here, the emphasis is on an aircraft of a specific type. We address issues of suitable data acquisition methods, and, most importantly, of functional signal analysis and synthesis techniques.

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User control over variable-rate time-stretching typically requires direct, manual adjustment of the time-dependent stretch rate. For time-stretching with transient preservation, rhythmic warping, rhythmic emphasis modification, or other effects that require additional timing constraints, however, direct manipulation is difficult. For a more user-friendly approach, we present work that allows a user to specify a time-dependent stiffness curve to warp the time axis of a recording, while maintaining other timing constraints, such as a desired overall recording length or musical rhythm quantization (e.g. straight-to-swing), providing a notion of stretchability to sound. To do so, the user-guided stiffness curve and timing constraints are translated into the desired time-dependent stretch rate via a constrained optimization program motivated by a physical spring system. Once the time-dependent stretch rate is computed, appropriately modified variable-rate time-stretch processors are used to process the sound. Initial results are demonstrated using both a phase-vocoder and pitch-synchronous overlap-add processor.

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A sound scene can be defined as any “environmental” sound that has a consistent background texture, with one or more potentially recurring foreground events. We describe a data-driven framework for analyzing, transforming, and synthesizing high-quality sound scenes, with flexible control over the components of the synthesized sound. Given one or more sound scenes, we provide well-defined means to: (1) identify points of interest in the sound and extract them into reusable templates, (2) transform sound components independently of the background or other events, (3) continually re-synthesize the background texture in a perceptually convincing manner, and (4) controllably place event templates over the background, varying key parameters such as density, periodicity, relative loudness, and spatial positioning. Contributions include: techniques and paradigms for template selection and extraction, independent sound transformation and flexible re-synthesis; extensions to a wavelet-based background analysis/synthesis; and user interfaces to facilitate the various phases. Given this framework, it is possible to completely transform an existing sound scene, dynamically generate sound scenes of unlimited length, and construct new sound scenes by combining elements from different sound scenes. URL: http://taps.cs.princeton.edu/

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This paper presents a controlled evaluation framework for recovering deterministic stereo timecode waveforms used in digital vinyl systems. Three lightweight decoding policies are compared: Fixed-Threshold Symbol Decoder (FTSD), Adaptive-Threshold Symbol Decoder (ATSD), and State-Constrained Temporal Decoder (SCTD). The work provides matched comparison and explicit separation of availability, reference-relative correctness, and temporal behavior across noise, dropout, and clipping conditions.

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This paper presents simulations of nonlinear plate vibrations in relation to sound synthesis of gongs and cymbals. The von Kármán equations are shown and then solved in terms of the modes of the associated linear system. The modal equations obtained constitute a system of nonlinearly coupled Ordinary Differential Equations which are completely general as long as the modes of the system are known. A simple second-order time-stepping integration scheme yields an explicit resolution algorithm with a natural parallel structure. Examples are provided and the results discussed.

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This paper describes research into signal transformation operators allowing to modify the vibrato extent in recorded sound signals. A number of operators are proposed that deal with the problem taking into account different levels of complexity. The experimental validation shows that the operators are effective in removing existing vibrato in real world recordings at least for the idealized case of long notes and with properly segmented vibrato sections. It shows as well that for instruments with significant noise level (flute) independent treatment of noise and harmonic signal components is required.

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For their low and constant computational cost, non-iterative methods for the solution of differential problems are gaining popularity in virtual analog provided their stability properties and accuracy level afford their use at no exaggerate temporal oversampling. At least in some application case studies, one recent family of noniterative schemes has shown promise to outperform methods that achieve accurate results at the cost of iterating several times while converging to the numerical solution. Here, this family is contextualized and studied against known classes of non-iterative methods. The results from these studies foster a more general discussion about the possibilities, role and prospective use of non-iterative methods in virtual analog.

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In the field of virtual analog modeling, a variety of methods have been proposed to systematically derive simulation models from circuit schematics. However, they typically rely on implicit numerical methods to transform the differential equations governing the circuit to difference equations suitable for simulation. For circuits with non-linear elements, this usually means that a non-linear equation has to be solved at run-time at high computational cost. As an alternative to fully-implicit numerical methods, a family of non-iterative discretization schemes has recently been proposed, allowing a significant reduction of the computational load. However, in the original presentation, several assumptions are made regarding the structure of the ODE, limiting the generality of these schemes. Here, we show that for the second-order accurate variant in particular, the method is applicable to general ODEs. Furthermore, we point out an interesting connection to the implicit midpoint method.

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