Paper Archive

We revisit pulsetable synthesis, an efficient technique for generating plausible and expressive wind instrument tones. Based on the principles of pulse forming theory, this method models sound production as the periodic repetition of shaped pulses characterizing the target instruments' spectral envelope. In this approach, single-cycle waveforms, referred to as pulses, are stored in pulsetables indexed by their corresponding fundamental frequency. During synthesis, the pulses are read from these tables to form a periodic waveform, which is further shaped by time-varying low-pass filtering, amplification, and reverberation. These processes are guided by control signal contours that describe how fundamental frequency, brightness, and loudness evolve over time. Through case studies with real-world wind instrument recordings, we show how the interplay between these control signals gives rise to articulations such as attack transients, vibrato, and growl. Finally, we discuss the potential of this framework for integration into Differentiable Digital Signal Processing (DDSP) models, where neural networks could learn synthesis parameters directly from training data.

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This paper introduces Kronos, a vectorizing Just in Time compiler designed for musical programming systems. Its purpose is to translate abstract mathematical expressions into high performance computer code. Musical programming system design criteria are considered and a three-tier model of abstraction is presented. The low level expression Metalanguage used in Kronos is described, along with the design choices that facilitate powerful, yet transparent vectorization of the machine code.

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Sound control by gesture is a peculiar topic in Human-Computer Interaction: many different approaches to it are available, focusing each time on diversified perspectives. Our point of view is an interdisciplinary one: taking into account technical considerations about control theory and sound processing, we try to explore the expressiveness world which is closer to psychology theories. Starting from a state of the art which outlines two main approaches to the problem of ”making sound with gestures”, we will delve into psychological theories about expressiveness, describing in particular possible applications dealing with intermodality and mixed reality environments related to the Gestalt Theory. HCI design can indeed benefit from this kind of approach because of the quantitative methods that can be applied to measure expressiveness. Interfaces can be used in order to convey expressiveness, which is a plus of information that can help interacting with the machine; this kind of information can be coded as spatio-temporal schemes, as it is stated in Gestalt theory.

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This paper introduces research issues on multimodal interaction and interfaces for expressive sound control. We introduce Multisensory Integrated Expressive Environments (MIEEs) as a framework for Mixed Reality applications in the performing arts. Paradigmatic contexts for applications of MIEEs are multimedia concerts, interactive dance / music / video installations, interactive museum exhibitions, distributed cooperative environments for theatre and artistic expression. MIEEs are user-centred systems able to interpret the high-level information conveyed by performers through their expressive gestures and to establish an effective multisensory experience taking into account expressive, emotional, affective content. The lecture discusses some main issues for MIEEs and presents the EyesWeb (www.eyesweb.org) open software platform which has been recently redesigned (version 4) in order to better address MIEE requirements. Short live demonstrations are also presented.

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Recent diffusion-based generative models have achieved strong results in domain-specific audio generation tasks such as speech, singing, and instrumental music synthesis. However, these models are typically specialized and do not generalize well to mixed or intermediate audio types. In this work, we adapt a diffusion-based model originally designed for multi-instrument music synthesis to voice conversion, covering both speech and singing within a unified framework. Specifically, we extend musical note-based conditioning to include phonetic posteriorgrams (PPGs) and pitch contours, and reinterpret timbre conditioning as speaker or singer identity via feature-wise linear modulation. Experiments show that the adapted model matches or surpasses a dedicated voice conversion system in terms of naturalness and performer similarity, while maintaining accurate pitch control across speech and singing. At the same time, we observe limitations in phonetic fidelity and a degradation in vocal quality when incorporating instrumental training data. Furthermore, we demonstrate that off-the-shelf feature extractors provide effective conditioning signals, enabling large-scale self-supervised training without manual annotations. These results highlight the potential of cross-domain model transfer towards unified audio generation systems capable of handling speech, singing, and music.

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In this paper, we present a method for estimating the local keys of an audio signal. We propose to address the problem of local key finding by investigating the possible combination and extension of different previous proposed global key estimation approaches. The specificity of our approach is that we introduce key dependency on the harmonic and the metric structures. In this work, we focus on the relationship between the chord progression and the local key progression in a piece of music. A contribution of our work is that we address the problem of finding a good analysis window length for local key estimation by introducing information related to the metric structure in our model. Key estimation is not performed on empirical-chosen segment length but on segments that are adapted to the analyzed piece and independent from the tempo. We evaluate and analyze our results on a new database composed of classical music pieces.

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This article explains how to apply time–frequency scattering, a convolutional operator extracting modulations in the time–frequency domain at different rates and scales, to the re-synthesis and manipulation of audio textures. After implementing phase retrieval in the scattering network by gradient backpropagation, we introduce scale-rate DAFx, a class of audio transformations expressed in the domain of time–frequency scattering coefficients. One example of scale-rate DAFx is chirp rate inversion, which causes each sonic event to be locally reversed in time while leaving the arrow of time globally unchanged. Over the past two years, our work has led to the creation of four electroacoustic pieces: FAVN; Modulator (Scattering Transform); Experimental Palimpsest; Inspection (Maida Vale Project) and Inspection II; as well as XAllegroX (Hecker Scattering.m Sequence), a remix of Lorenzo Senni’s XAllegroX, released by Warp Records on a vinyl entitled The Shape of RemiXXXes to Come.

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In this paper, we present a model for the modulation of multiperformer microtiming variation in musical groups. This is done using a multivariate Markov model, in which the relationship between players is modelled using an interdependence matrix (α) and a multidimensional state transition matrix (S). This method allows us to generate more natural sounding musical sequences due to the reduction of out-of-phase errors that occur in Gaussian pseudorandom and player-independent probabilistic models. We verify this using subjective listening tests, where we demonstrate that our multivariate model is able to outperform commonly used univariate models at producing human-like microtiming variability. Whilst the participants in our study judged the real time sequences performed by humans to be more natural than the proposed model, we were still able to achieve a mean score of 63.39% naturalness, suggesting microtiming interdependence between players captured in our model significantly enhances the humanisation of group musical sequences.

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We present a self-supervised, event-based sound matching model using a differentiable extended Karplus-Strong algorithm. To avoid relying on external onset and fundamental frequency detectors, we explore training methodologies combining parameter losses on synthetic data with audio losses. We demonstrate that time-domain fractional delay interpolation provides gradient accuracy comparable to frequency-sampling while avoiding time-aliasing in highly resonant time-varying scenarios. Through systematic gradient analysis, we reveal that standard spectral losses provide no meaningful directional gradients for onset times, heavily degrading joint training. Training exclusively with parameter losses on synthetic data effectively learns fundamental frequency, timbral parameters, and onset times, but struggles to generalise to monophonic studio recordings of plucked guitar. External detectors combined with audio losses generalise best, isolating the model to timbre optimisation. While our Karplus-Strong decoder recovers interpretable parameters and naturally captures the transient characteristics of plucked guitar, Harmonics plus Noise baselines yield higher reconstruction fidelity by most metrics.

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This paper investigates the optimization of Truncated Backpropagation Through Time (TBPTT) for training neural networks in digital audio effect modeling, with a focus on dynamic range compression. The study evaluates key TBPTT hyperparameters – sequence number, batch size, and sequence length – and their influence on model performance. Using a convolutional-recurrent architecture, we conduct extensive experiments across datasets with and without conditioning by user controls. Results demonstrate that carefully tuning these parameters enhances model accuracy and training stability, while also reducing computational demands. Objective evaluations confirm improved performance with optimized settings, while subjective listening tests indicate that the revised TBPTT configuration maintains high perceptual quality.

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