Paper Archive

In this work, the authors develop a higher order Ambisonic (HOA) decoder that compensates for listening room reverberation by cascading a conventional decode matrix with a crosstalk matrix derived from room impulse responses (RIR) constrained over the listener area, namely by sampling over a spherical boundary according to HOA convention. A decoder is generated for simulated RIRs of mixed specular and diffuse reverberation, and spectral and spatiotemporal energy distributions are shown for directional and diffuse HOA signals rendered through the reverberant room. Results demonstrate that the decoder renders temporally-variant directional sources with higher directivity as compared to a conventional decoder.ing Ambisonic decoders in reverberant environments using closed-hull crosstalk convolution. By modeling the acoustic path from each loudspeaker to the listener's ears via measured room impulse responses, we formulate equalization filters that compensate for room-induced distortions in the decoded signals. The proposed closed-hull approach constrains the solution to preserve perceptually relevant spatial cues while minimizing spectral coloration. Listening tests demonstrate improved externalization, timbral transparency, and localization accuracy compared to standard free-field decoding in reverberant conditions.

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Processing and recording multitrack audio from an electronic drum set is demanding of computational power and hardware resources. In this paper, we present a complete musical instrument system capable of up to 16-channel percussion sampling, processing, and recording, all in real time. The system leverages a field programmable gate array (FPGA) for parallel audio processing and includes audio effects such as pitch shift, delay, reverb, distortion, a virtual analog low-pass filter, and bit crush. The FPGA also provides interfaces for other system hardware, including an Ethernet audio interface and various audio effect control interfaces. The final design has a cost of under $500 and utilizes about half of the hardware resources on an entry-level FPGA, providing a future platform for more advanced percussion synthesis using real-time physical modeling.

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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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This paper is a technical report on the methodology submitted for Task A of the 1st DAFx Parameter Estimation Challenge. The goal of the challenge’s task is to invert the multi-dimensional physical and geometric parameters of a virtual plate reverberator given a target reference impulse response. To achieve this, we present a multi-stage gradient-free optimization framework. This three-stage optimization is computed using an efficient physics-based simulator, starting with an optimization of only mode frequency-determining physical parameters, followed by a 6-DoF parameter optimization with position-determining ones and a final phase for frequency- and position-independent mode amplitude estimation.

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We describe a Task B submission for the 1st DAFx Parameter Estimation Challenge. A matching-pursuit anchor stage seeds a multi-view subband autoregressive (AR) pole harvester on the raw IR and its first two finite differences; since linear filtering preserves pole locations while changing residues, the three views expose complementary subsets of the same pole set. Bands in which the AR order saturates are recursively split, and any remaining under-resolved region is completed from an IR-derived saturation indicator. Gains are assigned by a global ridge least-squares (LS) fit in the damped-biquad atom convention. The pipeline uses only the unnormalized IR; so it does not use plate parameters, analytical modal-frequency or decay laws, .wav files, or Task A information.

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Wave Digital Filters are a circuit-modeling paradigm well-suited for reusable software implementation, but existing software implementations often incur significant overhead due to run-time abstractions and data layout constraints. This paper presents a performance-oriented toolchain for implementing Wave Digital circuit models based on static code generation. The toolchain consists of a declarative circuit description language, a compiler that generates circuit simulation code with minimal persistent state and no run-time abstraction, and a minimal runtime library implementing specialized circuit components as Wave Digital Filters. Performance measurements across several test circuits demonstrate that the generated models consistently outperform existing implementations, and achieve near-ideal performance relative to a theoretical execution bound.

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