Download EVERTims: Open Source Framework for Real-time Auralization in Architectural Acoustics and Virtual Reality
This paper presents recent developments of the EVERTims project, an auralization framework for virtual acoustics and real-time room acoustic simulation. The EVERTims framework relies on three independent components: a scene graph editor, a room acoustic modeler, and a spatial audio renderer for auralization. The framework was first published and detailed in [1, 2]. Recent developments presented here concern the complete re-design of the scene graph editor unit, and the C++ implementation of a new spatial renderer based on the JUCE framework. EVERTims now functions as a Blender add-on to support real-time auralization of any 3D room model, both for its creation in Blender and its exploration in the Blender Game Engine. The EVERTims framework is published as open source software: http://evertims.ircam. fr.
Download Pywdf: An Open Source Library for Prototyping and Simulating Wave Digital Filter Circuits in Python
This paper introduces a new open-source Python library for the modeling and simulation of wave digital filter (WDF) circuits. The library, called pwydf, allows users to easily create and analyze WDF circuit models in a high-level, object-oriented manner. The library includes a variety of built-in components, such as voltage sources, capacitors, diodes etc., as well as the ability to create custom components and circuits. Additionally, pywdf includes a variety of analysis tools, such as frequency response and transient analysis, to aid in the design and optimization of WDF circuits. We demonstrate the library’s efficacy in replicating the nonlinear behavior of an analog diode clipper circuit, and in creating an allpass filter that cannot be realized in the analog world. The library is well-documented and includes several examples to help users get started. Overall, pywdf is a powerful tool for anyone working with WDF circuits, and we hope it can be of great use to researchers and engineers in the field.
Download Towards Ontological Representations of Digital Audio Effects
In this paper we discuss the development of ontological representations of digital audio effects and provide a framework for the description of digital audio effects and audio effect transformations. After a brief account on our current research in the field of highlevel semantics for music production using Semantic Web technologies, we detail how an Audio Effects Ontology can be used within the context of intelligent music production tools, as well as for musicological purposes. Furthermore, we discuss problems in the design of such an ontology arising from discipline-specific classifications, such as the need for encoding different taxonomical systems based on, for instance, implementation techniques or perceptual attributes of audio effects. Finally, we show how information about audio effect transformations is represented using Semantic Web technologies, the Resource Description framework (RDF) and retrieved using the SPARQL query language.
Download Fractal Modulation Effects
Fractal modulation is obtained by forming a power weighted superposition of scaled and modulated versions of the signal. The resulting signal is self-similar with fractal characteristics. In this paper we explore fractal modulation as a powerful method to generate rich signals, useful both for the synthesis of complex sounds, like the sounds from natural events or ecological sounds, or as control functions of audio effects. The wavelet transform can be used as an efficient tool in order to generate a subset of fractal modulated signals that are power homogeneous. Any signal used as a seed for fractal modulation is transformed into a multiscale sound by means of a tree-structured multirate filter bank. Moreover, by superimposing a structured modulation scheme one can generate pseudo-periodic sounds whose partials have fractal behavior.
Download Compression of Head-Related Transfer Functions Using Piecewise Cubic Hermite Interpolation
We present a spline-based method for compressing and reconstructing Head-Related Transfer Functions (HRTFs) that preserves perceptual quality. Our approach focuses on the magnitude response and consists of four stages: (1) acquiring minimumphase head-related impulse responses (HRIR), (2) transforming them into the frequency domain and applying adaptive Wiener filtering to preserve important spectral features, (3) extracting a minimal set of control points using derivative-based methods to identify local maxima and inflection points, and (4) reconstructing the HRTF using piecewise cubic Hermite interpolation (PCHIP) over the refined control points. Evaluation on 301 subjects demonstrates that our method achieves an average compression ratio of 4.7:1 with spectral distortion ≤ 1.0 dB in each Equivalent Rectangular Band (ERB). The method preserves binaural cues with a mean absolute interaural level difference (ILD) error of 0.10 dB. Our method achieves about three times the compression obtained with a PCA-based method.
Download Towards Efficient Emulation of Nonlinear Analog Circuits for Audio Using Constraint Stabilization and Convex Quadratic Programming
This paper introduces a computationally efficient method for the emulation of nonlinear analog audio circuits by combining state-space representations, constraint stabilization, and convex quadratic programming (QP). Unlike traditional virtual analog (VA) modeling approaches or computationally demanding SPICE-based simulations, our approach reformulates the nonlinear differential-algebraic (DAE) systems that arise from analog circuit analysis into numerically stable optimization problems. The proposed method efficiently addresses the numerical challenges posed by nonlinear algebraic constraints via constraint stabilization techniques, significantly enhancing robustness and stability, suitable for real-time simulations. A canonical diode clipper circuit is presented as a test case, demonstrating that our method achieves accurate and faster emulations compared to conventional state-space methods. Furthermore, our method performs very well even at substantially lower sampling rates. Preliminary numerical experiments confirm that the proposed approach offers improved numerical stability and real-time feasibility, positioning it as a practical solution for high-fidelity audio applications.
Download Black-box Modeling of Distortion Circuits with Block-Oriented Models
This paper describes black-box modeling of distortion circuits. The analyzed distortion circuits all originate from guitar effect pedals, which are widely used to enrich the sound of an electric guitar with harmonics. The proposed method employs a blockoriented model which consists of a linear block (filter) and a nonlinear block. In this study the nonlinear block is represented by an extended parametric input/output mapping function. Three distortion circuits with different nonlinear elements are analyzed and modeled. The linear and nonlinear parts of the circuit are analyzed and modeled separately. The Levenberg–Marquardt algorithm is used for iterative optimization of the nonlinear parts of the circuits. Some circuits could not be modeled with high accuracy, but the proposed model has shown to be a versatile and flexible tool when modeling distortion circuits.
Download Low bit-rate audio coding with hybrid representations
We present a general audio coder based on a structural decomposition : the signal is expanded into three features : its harmonic part, the transients and the remaining part (referred as the noise). The rst two of these layers can be very eciently encoded in a wellchosen basis. The noise is by construction modelized as a gaussian (colored) random noise. Furthermore, this decomposition allows a good time-frequency psycoacoustic modeling, as it dircetly provides us with the tonal and nontonal part of the signal.
Download Evaluation of a Stochastic Reverberation Model Based on the Source Image Principle
Various audio signal processing applications, such as source separation and dereverberation, require an accurate mathematical modeling of the input audio data. In the literature, many works have focused on source signal modeling, while the reverberation model is often kept very simplistic. This paper aims to investigate a stochastic room impulse response model presented in a previous article: this model is first adapted to discrete time, then we propose a parametric estimation algorithm, that we evaluate experimentally. Our results show that this algorithm is able to efficiently estimate the model parameters, in various experimental settings (various signal-to-noise ratios and absorption coefficients of the room walls).
Download Efficient Modeling and Synthesis of Bell-like Sounds
This paper describes two different techniques that can be used to model and synthesize bell-like sounds. The first one is a sourcefilter model based on frequency-zooming ARMA (pole-zero) modeling techniques. The frequency-zooming approach is powerful also in modal analysis of bell sound behavior. The second technique is based on a digital waveguide with a single loop filter that is designed to generate inharmonic partials by including one or more second-order allpass sections in the loop filter, possibly augmented with one or a few parallel resonators. A small handbell with inharmonic partials was recorded and used as a target of modeling and synthesis. Sound examples are found in http://www.acoustics.hut.fi/demos/dafx02/.