Paper Archive

Impulse response (IR) collections are useful in virtual acoustics, sound design, and field-based acoustic research, but they remain difficult to explore as continuous resources in lightweight real-time plugin workflows. This demo paper presents IRIS, a VST3 plugin for arranging, navigating, and auditioning measured or user-defined IR collections in a two-dimensional navigation plane. Each IR is represented as a node whose position can be imported from metadata or assigned manually. During navigation, nearby responses are combined using Gaussian distance-based weighting, while a bounded active set limits the number of simultaneous convolutions. The system also includes smoothing, hysteresis, optional preprocessing, boundary attenuation, OSC control, and coupled multichannel handling. The demo focuses on workflow and audible behavior rather than perceptual validation. A short timing characterization reports practical real-time limits as a function of IR length, buffer size, and active-set size. IRIS is presented as a practical tool for exploratory, analytical, and creative navigation of IR collections rather than as a physically optimal interpolation method.

Download

This work presents sfFDN, a modular and real-time-capable C++ library for Feedback Delay Networks (FDNs), together with the companion FDN Sandbox application designed for interactive experimentation, analysis, and parameter optimization. The library implements the canonical FDN as well as several recent extensions, including filter feedback matrices, velvet-noise decorrelation filters, and two-stage graphic equalizers for attenuation and tone correction. The Sandbox application exposes these features through a graphical interface, providing a suite of real-time visualizations, as well as an optimization framework supporting nine algorithms from the ensmallen library, with built-in loss functions for both colorless reverberation and room impulse response matching. Both the library and the application are open-source.

Download

We present FDverb, a frequency-domain artificial reverberator, based on the idea of a vocoder with a noise carrier signal. Using a short-time Fourier transform (STFT) for analysis and synthesis, FDverb generates late reverberation by weighting spectral noise components with envelopes. We extend FDverb with early reflections, nonlinear decay, and pitch shifting. These extensions enable creative sound-design applications. We provide FDverb as an open-source DAW plug-in, using the JUCE framework.

Download

This paper accompanies a demonstration of a real-time audio plug-in for a dynamic spatial reverb. The reverb is based on acoustic measurements of the Bunkervik creative arts space in Brescia, Italy. A modal synthesis reverberation engine was created based on measured impulse responses from three locations in the tunnel. Using a common set of modal frequencies the position of the receiver can be dynamically moved through the space by interpolating between data sets of residue weights and FIR filter taps. The audio plug-in also allows real-time manipulation of the high-frequency content, damping, and microphone rotation, all of which can be modulated using two LFOs.

Download

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.

Download

In electric guitars, the vibrations of the strings are typically sensed by coils of wire combined with a magnet, called pickups. The pickups and their position along the strings contribute strongly to the instrument's sound. Most guitars feature one to three pickups, each spanning across all strings with fixed positions and generating a single mono output. The work of this Master's Thesis at ETH Zürich introduces a new pickup system called PolyMap, which senses each string individually and at multiple locations. The system is demonstrated with a custom-made eight-string guitar that contains eight pickups per string for a total of 64 pickups. The signals from these 64 pickups are individually digitized inside the guitar and transmitted over a multichannel audio digital interface (MADI), a low-latency digital audio interface, to a computer for further processing. PolyMap enables high-resolution sensing of an electric guitar's strings and enables extensive post-processing capabilities for musicians, audio engineers, and researchers. To the best of our knowledge, this is the first polyphonic guitar pickup system with such a complete feature set.

Download

WaveNet-style convolutional networks emulate tube amplifiers and distortion pedals with high fidelity, but their computational cost has confined them to desktops or dedicated DSP hardware. We present a sparse-enabled WaveNet inference engine for iOS that runs heavily pruned neural guitar amplifier models in real time on iPhones. Aggressive iterative magnitude pruning removes 90% of the network weights with no perceptible loss in quality. A custom sparse C++ engine turns this sparsity directly into compute savings, sustaining low-latency real-time operation on a CPU-only iPhone implementation where the dense model cannot. On-device output matches the trained model to within int16 quantization error. At the demonstration, visitors will play a guitar through the app on iPhone hardware and A/B the on-device pruned model against the physical pedal it emulates. Source code and audio examples are available online.

Download

This demonstration presents Praat AudioTools, an open-source hybrid toolkit that repurposes Praat's phonetic-analysis environment for electroacoustic composition, sound design, and offline analysis–resynthesis workflows. Rather than treating analysis data as temporary measurements hidden inside an audio processor, Praat AudioTools exposes pitch contours, formant structures, temporal segmentations, spectral descriptors, phrase boundaries, stochastic trajectories, and host-application exchange files as editable compositional objects. These objects can be inspected, modified, chained, reused, and rendered into new sound transformations. The demonstration focuses on seven offline workflows: Neural Ambient Drone Designer, Praat for Max and Max for Live, Phase-Space Composer, Reich Generator, MCMC Musical Variation, Messagesquisse Opening, and Vector/Full-Chain composition workflows. None of the examples are presented as real-time effects. Instead, they show an "edit-in-the-middle" model in which sound is analyzed, intermediate representations are made visible, compositional decisions are applied to those representations, and the result is rendered as audio. The aim is to demonstrate a transparent alternative to both conventional black-box audio effects and end-to-end generative audio systems: a compositional environment where analysis objects become controllers, traces, scores, and reproducible technical artifacts.

Download

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.

Download

This work examines the use of the time-varying delay line (TVDL) to implement loopback frequency modulation (LBFM), an oscillator that loops back to modulate its own frequency. Digital delay lines are used regularly in sound synthesis/processing to model the pure delay associated with one-dimensional acoustic propagation. When the delay is made time varying, the TVDL time warps the input according to a delay function, altering the input's instantaneous frequency and phase. As a result, the TVDL is well suited for delay-based effects and, in particular, those involving frequency/phase modulation for which the TVDL delay function is oscillatory and thus bounded by a maximum and minimum delay. Sustaining a constant change in sounding frequency however, corresponds to a delay function having a term that is linear in time, making it limited only by the length of the input signal. While TVDLs may still be used when there is a pitch shift, limiting the delay by simple wrapping of the delay function and/or cross fading between multiple TVDLs may not be adequate to avoid audible artifacts. In LBFM, the resulting phase has both linear and oscillating terms and the resulting signal undergoes a sustained shift in the fundamental frequency that makes a TVDL implementation more challenging. An alternate closed-form representation of the LBFM oscillator, however, provides the information necessary for accurately wrapping the TVDL delay function and ensuring it is suitably bounded so that the produced sound is free of phase distortion and audible artifacts.

Download