Paper Archive

This paper proposes POLAR, a framework for the optimisation of loudspeaker signals using end-to-end differentiable perceptual loss functions. The framework optimises multiple perceptual attributes across multiple listeners, offering a versatile method for a range of problems. This versatility stems from the ability to customise the number of loudspeakers, listeners, and the weighting applied to different perceptual attributes. Here, we apply the method to four problems: (a) source panning for a single listener in stereo reproduction, (b) single-listener colouration matching in stereo reproduction, (c) extended sweet spot using stereo pairs beamforming, and (d) multi-attribute perceptually driven panning in stereo reproduction. The first three problems are evaluated against solutions traditionally used for these tasks: solutions of (a) are shown to be similar to those obtained with tangent panning law and vector-base amplitude panning (VBAP), solutions of (b) are shown to be similar to those obtained for cross-talk cancellation, and solutions of (c) are shown to be similar to those obtained in earlier work on directivity pattern optimisation for sweet spot widening. Each of these solutions was previously obtained using fundamentally different methodologies, demonstrating the flexibility and broad applicability of the proposed framework.

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This work extends our previously proposed Quadratic Programming (QP) approach for the emulation of nonlinear analog audio circuits by formalizing its main numerical ingredients and introducing a residual-driven adaptive multi-rate scheme. Starting from a state-space Differential Algebraic System of Equations (DAE) formulation, the nonlinear algebraic circuit device relations are replaced inside the QP by a first-order surrogate linear constraint, and the post-step nonlinear residual is shown to act as a valid defect indicator for adaptive step-size control. This yields a single-step simulation procedure that avoids the usual combination of nonlinear iterative solves and separate integration updates. The method is evaluated on a diode clipper, a BJT common-emitter amplifier, and a Colpitts oscillator, using SPICE as a baseline reference. The results show that adaptive step sizing considerably improves agreement with the reference solution, that the pseudo-inverse implementation is essentially equivalent to the full equality-constrained QP in the tested cases, and that the proposed formulation remains effective beyond the baseline clipper example, including for a self-oscillating circuit. These results position the proposed method as a promising bridge between SPICE-like interpretability and the efficiency demands of virtual analog (VA) audio applications.

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Robust Principal Component Analysis (RPCA) is a technique to decompose signals into sparse and low rank components, and has recently drawn the attention of the MIR field for the problem of separating leading vocals from accompaniment, with appealing results obtained on small excerpts of music. However, the performance of the method drops when processing entire music tracks. We present an adaptive formulation of RPCA that incorporates music content information to guide the decomposition. Experiments on a set of complete music tracks of various genres show that the proposed algorithm is able to better process entire pieces of music that may exhibit large variations in the music content, and compares favorably with the state-of-the-art.

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In this paper we present a working system for separating a piano recording into events representing individual piano notes. Each note is parameterized with a transient-plus-harmonics model that, should all the parameters be reliably estimated, would produce near perfect reconstruction for each note as well as for the whole recording. However, interference between overlapping notes makes it hard to estimate parameters from their combination. In this work we propose to assess the estimability of sinusoidal parameters via their apparent degree of interference, estimate the estimable ones using algorithms suitable for different interference situations, and infer the hard-to-estimate parameters from the estimated ones. The outcome is a sequence of separate, parameterized piano notes that perceptually highly resemble, if are not identical to, the notes in the original recording. This allows for later analysis and processing stages using algorithms designed for separate notes.

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We introduce FoleySet, a human-annotated Foley sound dataset designed to support research on sound-event understanding and Foley sound generation. The dataset provides annotations at multiple levels of granularity, capturing both broad event categories and more detailed semantic or production-related attributes. This multi-level structure supports tasks such as classification, retrieval, captioning, and controllable generation. FoleySet is intended to address the limited availability of systematically annotated Foley material and to provide a common resource for evaluating models across different levels of semantic detail.

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Physical modeling sound synthesis has achieved remarkable success in terms of its fidelity to reality. In many cases, since modeling of the physical system is performed on the sounding objects that already exist in the real world, observation precedes the model itself. Departing from this convention, this paper aims to physically model the acoustic characteristics of objects that do not necessarily exist in reality. Specifically, we study wave propagation on compact two-dimensional (2D) manifolds that are non-orientable surfaces, such as the Klein bottle that cannot be embedded in three-dimensional Euclidean space without self-intersection. We derive closed-form expressions for the eigenfrequencies and mode shapes of non-orientable 2D topologies and study their acoustic characteristics. The modal structures are verified through comparison with finite-difference time-domain simulations. The results demonstrate how the topological character formed by the boundaries influences the acoustic resonances, and how the quotient-space framework provides a practical route to reverb synthesis on geometries with no physical counterpart.

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In this paper we present a revised and improved version of a recently proposed extended source-filter model for sound synthesis, transformation and hybridization of harmonic instruments. This extension focuses mainly on the application for impulsively excited instruments like piano or guitar, but also improves synthesis results for continuously driven instruments including their hybrids. This technique comprises an extensive analysis of an instruments sound database, followed by the estimation of a generalized instrument model reflecting timbre variations according to selected control parameters. Such an instrument model allows for natural sounding transformations and expressive control of instrument sounds regarding its control parameters.

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In this paper we investigate the problem of singer identification on acapella recordings of isolated notes. Most of studies on singer identification describe the content of signals of singing voice with features related to the timbre (such as MFCC or LPC). These features aim to describe the behavior of frequencies at a given instant of time (local features). In this paper, we propose to describe sung tone with the temporal variations of the fundamental frequency (and its harmonics) of the note. The periodic and continuous variations of the frequency trajectories are analyzed on the whole note and the features obtained reflect expressive and intonative elements of singing such as vibrato, tremolo and portamento. The experiments, conducted on two distinct data-sets (lyric and pop-rock singers), prove that the new set of features capture a part of the singer identity. However, these features are less accurate than timbre-based features. We propose to increase the recognition rate of singer identification by combining information conveyed by local and global description of notes. The proposed method, that shows good results, can be adapted for classification problem involving a large number of classes, or to combine classifications with different levels of performance.

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The synthesis of sound textures, such as rain, wind, or crowds, is an important application for cinema, multimedia creation, games and installations. However, despite the clearly defined requirments of naturalness and flexibility, no automatic method has yet found widespread use. After clarifying the definition, terminology, and usages of sound texture synthesis, we will give an overview of the many existing methods and approaches, and the few available software implementations, and classify them by the synthesis model they are based on, such as subtractive or additive synthesis, granular synthesis, corpus-based concatenative synthesis, wavelets, or physical modeling. Additionally, an overview is given over analysis methods used for sound texture synthesis, such as segmentation, statistical modeling, timbral analysis, and modeling of transitions. 2

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This paper presents a review on techniques for signal reconstruction without phase, i.e. when only the spectrogram (the squared magnitude of the Short Time Fourier Transform) of the signal is known. The now standard Griffin and Lim algorithm will be presented, and compared to more recent blind techniques. Two important issues are raised and discussed: first, the definition of relevant criteria to evaluate the performances of different algorithms, and second the question of the unicity of the solution. Some ways of reducing the complexity of the problem are presented with the injection of additional information in the reconstruction. Finally, issues that prevents optimal reconstruction are examined, leading to a discussion on what seem the most promising approaches for future research.

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