Download Model Bending: Teaching Circuit Models New Tricks A technique is introduced for generating novel signal processing systems grounded in analog electronic circuits, called model bending. By applying the ideas behind circuit bending to models of nonlinear analog circuits it is possible to create novel nonlinear signal processors which mimic the behavior of analog electronics, but which are not possible to implement in the analog realm. The history of both circuit bending and circuit modeling is discussed, as well as a theoretical basis for how these approaches can complement each other. Potential pitfalls to the practical application of model bending are highlighted and suggested solutions to those problems are provided, with examples.
Download Stable Limit Cycles as Tunable Signal Sources This paper presents a method for synthesizing audio signals from
nonlinear dynamical systems exhibiting stable limit cycles, with
control over frequency and amplitude independent of changes to
the system’s internal parameters. Using the van der Pol oscillator
and the Brusselator as case studies, it is demonstrated how parameters are decoupled from frequency and amplitude by rescaling the
angular frequency and normalizing amplitude extrema. Practical
implementation considerations are discussed, as are the limits and
challenges of this approach. The method’s validity is evaluated experimentally and synthesis examples show the application of tunable nonlinear oscillators in sound design, including the generation
of transients in FM synthesis by means of a van der Pol oscillator
and a Supersaw oscillator bank based on the Brusselator.
Download Additive Synthesis Of Sound By Taking Advantage Of Psychoacoustics In this paper we present an original technique designed in order to speed up additive synthesis. This technique consists in taking into account psychoacoustic phenomena (thresholds of hearing and masking) in order to ignore the inaudible partials during the synthesis process, thus saving a lot of computation time. Our algorithm relies on a specific data structure called “skip list” and has proven to be very efficient in practice. As a consequence, we are now able to synthesize an impressive number of spectral sounds in real time, without overloading the processor.
Download Towards Inverse Virtual Analog Modeling Several digital signal processing approaches, generally referred to as Virtual Analog (VA) modeling, are currently under development for the software emulation of analog audio circuitry. The main purpose of VA modeling is to faithfully reproduce the behavior of real-world audio gear, e.g., distortion effects, synthesizers or amplifiers, using efficient algorithms. In this paper, however, we provide a preliminary discussion about how VA modeling can be exploited to infer the input signal of an analog audio system, given the output signal and the parameters of the circuit. In particular, we show how an inversion theorem known in circuit theory, and based on nullors, can be used for this purpose. As recent advances in Wave Digital Filter (WDF) theory allow us to implement circuits with nullors in a systematic fashion, WDFs prove to be useful tools for inverse VA modeling. WDF realizations of a nonlinear audio system and its inverse are presented as an example of application.
Download Audio Rendering System Design for an Object Oriented Audio Visual Human Perception Assessment Tool The cognitive processes behind human bimodal (audio visual) perception are not well understood. This contribution presents an approach to reach a deeper understanding by means of subjective assessments of (interactive) audio visual applications. A tool developed for performing these assessments is introduced, and the audio rendering system design of that tool is explained: its modular character, the signal processing flow as well as the possible reproduction setups are discussed. Finally, an example for the assessment of geometrically based room simulation and preliminary test results are given.
Download Sound Processing in OpenMusic This article introduces some new possibilities of audio manipulations and sound processing in the Computer-Aided Composition environment OpenMusic. Interfaces with underlying sound processing systems are described, with an emphasis on the use of the symbolic and visual programming environment for the design of sound computation processes.
Download Do You Hear a Bump or a Hole? An Experiment on Temporal Aspects in Footsteps Recognition In this paper, we present a preliminary experiment whose goal is to assess the role of temporal aspects in sonically simulating the act of walking on a bump or a hole. In particular, we investigate whether the timing between heel and toe and the timing between footsteps affects the perception of walking on unflat surfaces. Results show that it is possible to simulate a bump or a hole by only using temporal information in the auditory modality.
Download Application of non-negative matrix factorization to signal-adaptive audio effects This paper proposes novel audio effects based on manipulating an audio signal in a representation domain provided by non-negative matrix factorization (NMF). Critical-band magnitude spectrograms Y of sounds are first factorized into a product of two lower-rank matrices so that Y ≈ BG. The parameter matrices B and G are then processed in order to achieve the desired effect. Three classes of effects were investigated: 1) dynamic range compression (or expansion) of the component spectra or gains, 2) effects based on rank-ordering the components (colums of B and the corresponding rows of G) according to acoustic features extracted from them, and then weighting each component according to its rank, and 3) distortion effects based on controlling the amount of components (and thus the reconstruction error) in the above linear approximation. The subjective quality of the effects was assessed in a listening test.
Download Perceptual Linear Filters: Low-Order ARMA Approximation for Sound Synthesis This paper deals with the approximation of a given frequency response by a low-order linear ARMA filter (Auto-Regressive Moving Average). The aim of this work is the audio synthesis, then to improve the perceptual quality, a criterion based on human listening is defined and minimized. Two complementary approaches are proposed here for solving this non-linear and non-convex problem: first, a weighted version of the Iterative Prefiltering, second, an adaptation of the Gauss-Newton method. This algorithm is adapted to guarantee the causality/stability of the obtained filter, and eventually its minimum phase property. The benefit of the new method is illustrated and evaluated.
Download DHM and FDTD based Hardware Sound Field Simulation Acceleration Sound field simulation is widely used for acoustic design; however, this simulation needs many computational resources. On the other hand, FPGA becomes major for acceleration. To take advantage of hardware acceleration by FPGA, hardware oriented algorithm which consumes small number of gates and memory is necessary. This paper addresses hardware acceleration of sound field simulation using FPGA. Improved Digital Huygens Model (DHM) for hardware is implemented and speed up ratio is examined. For 2D simulation, the implemented accelerator is 1,170 times faster than software simulation. For 3D simulation, it is shown that FDTD based method is suitable for hardware implementation and required hardware resource are estimated.