Download A Real-Time Approach for Estimating Pulse Tracking Parameters for Beat-Synchronous Audio Effects Predominant Local Pulse (PLP) estimation, an established method for extracting beat positions and other periodic pulse information from audio signals, has recently been extended with an online variant tailored for real-time applications. In this paper, we introduce a novel approach to generating various real-time control signals from the original online PLP output. While the PLP activation function encodes both predominant pulse information and pulse stability, we propose several normalization procedures to discern local pulse oscillation from stability, utilizing the PLP activation envelope. Through this, we generate pulse-synchronous Low Frequency Oscillators (LFOs) and supplementary confidence-based control signals, enabling dynamic control over audio effect parameters in real-time. Additionally, our approach enables beat position prediction, providing a look-ahead capability, for example, to compensate for system latency. To showcase the effectiveness of our control signals, we introduce an audio plugin prototype designed for integration within a Digital Audio Workstation (DAW), facilitating real-time applications of beat-synchronous effects during live mixing and performances. Moreover, this plugin serves as an educational tool, providing insights into PLP principles and the tempo structure of analyzed music signals.
Download Spatial Auditory Displays - A study on the use of virtual audio environments as interfaces for users with visual disabilities This paper presents the work on a prototype spatial auditory display. Using high-definition audio rendering a sample application was presented to a mixed group of users with visual disabilities and normal sighted users. The evaluation of the prototype provided insights into how effective spatial presentation of sound can be in terms of human-computer interaction (HCI). It showed that typical applications with the most common interaction tasks like menus, text input and dialogs can be presented very effectively using spatial audio. It also revealed that there is no significant difference in effectiveness between normal sighted and visually impaired users. We believe that spatial auditory displays are capable to provide the visually impaired and blind access to modern information technologies in a more efficient way than common technologies and that they will be inevitable for multimodal displays in future applications.
Download Towards Transient Restoration in Score-informed Audio Decomposition Our goal is to improve the perceptual quality of transient signal components extracted in the context of music source separation. Many state-of-the-art techniques are based on applying a suitable decomposition to the magnitude of the Short-Time Fourier Transform (STFT) of the mixture signal. The phase information required for the reconstruction of individual component signals is usually taken from the mixture, resulting in a complex-valued, modified STFT (MSTFT). There are different methods for reconstructing a time-domain signal whose STFT approximates the target MSTFT. Due to phase inconsistencies, these reconstructed signals are likely to contain artifacts such as pre-echos preceding transient components. In this paper, we propose a simple, yet effective extension of the iterative signal reconstruction procedure by Griffin and Lim to remedy this problem. In a first experiment, under laboratory conditions, we show that our method considerably attenuates pre-echos while still showing similar convergence properties as the original approach. A second, more realistic experiment involving score-informed audio decomposition shows that the proposed method still yields improvements, although to a lesser extent, under non-idealized conditions.
Download Upcylcing Android Phones into Embedded Audio Platforms There are millions of sophisticated Android phones in the world that get disposed of at a very high rate due to consumerism. Their computational power and built-in features, instead of being wasted when discarded, could be repurposed for creative applications such as musical instruments and interactive audio installations. However, audio programming on Android is complicated and comes with restrictions that heavily impact performance. To address this issue, we present LDSP, an open-source environment that can be used to easily upcycle Android phones into embedded platforms optimized for audio synthesis and processing. We conducted a benchmark study to compare the number of oscillators that can be run in parallel on LDSP with an equivalent audio app designed according to modern Android standards. Our study tested six phones ranging from 2014 to 2018 and running different Android versions. The results consistently demonstrate that LDSP provides a significant boost in performance, with some cases showing an increase of more than double, making even very old phones suitable for fairly advanced audio applications.
Download Winding Numbers and Monodromy of Vector Bundles over a Circular Buffer The Möbius strip is perhaps the most recognizable topological object of general knowledge. It can be described mathematically in various ways including the formalism of line bundles. In this paper we discuss the bundle idea in the context of digital processing over a circular buffer and show how the idea leads to a more general notion known as monodromy, which describes the effect of the space on traversing a circle once. In this formulation, the monodromy of the Möbius strip is an orientation inversion characterized by a change in sign. This in turns leads to the concept of the winding number, which describes how many windings it takes to return to the original state. We discuss variable monodromy and illustrate that the winding number is robust under this variation. This will allow us to interpret previous disparate results in audio signal processing from Möbius waveguides to chaotic oscillators in delay loops in one unified framework. We close by showing how extending from line to vector bundles opens up the notion of braids to describe monodromy.
Download Resolving Grouped Nonlinearities in Wave Digital Filters using Iterative Techniques In this paper, iterative zero-finding techniques are proposed to resolve groups of nonlinearities occurring in Wave Digital Filters. Two variants of Newton’s method are proposed and their suitability towards solving the grouped nonlinearities is analyzed. The feasibility of the approach with implications for WDFs containing multiple nonlinearities is demonstrated via case studies investigating the mathematical properties and numerical performance of reference circuits containing diodes and transistors; asymmetric and symmetric diode clippers and a common emitter amplifier.
Download A Production-Oriented Framework for Evaluation of SFX Generation Industrial sound design requires audio generation systems that not only produce realistic audio, but also preserve the perceptual identity of a reference, support controllable variation, and remain efficient for practical workflows. Existing evaluations are usually tied to text-to-audio (TTA), unconditional, or task-specific settings, limiting assessment for reference-guided sound effects (SFX) variation. To address this gap, we present a production-oriented evaluation framework for structured comparison of heterogeneous audio generation and editing methods. Our framework identifies nine production requirements and explicitly accounts for differences in model capabilities, enabling comparison under a common production objective. A two-stage protocol is introduced: (1) a reference-guided audio-to-audio (ATA) variation task, in which all methods are evaluated under the same ESC-50 SFX adaptation setup, and (2) capability-specific analyses of native operations such as SFX morphing, temporal and energy alignment, inpainting, and targeted editing. This framework combines objective metrics (including FAD, ImageBind-based reference alignment, and diversity across generated variants), together with a human study of perceptual identity preservation and transient diagnosis. Our study reveals complementary strengths and trade-offs across baselines for different production needs. Among the full-generation baselines evaluated under a shared ATA setting, AudioX provides the strongest overall trade-off between reference alignment and diversity while still supporting SFX morphing. Other baselines remain most suitable for specific editing operations. Our framework establishes a structured evaluation and decision protocol for reference-guided SFX variation and provides a practical basis for designing future unified industrial audio generation pipelines. Audio demos and further details can be found on the accompanying web page.
Download A Nonlinear Method for Manipulating Warmth and Brightness In musical timbre, two of the most commonly used perceptual dimensions are warmth and brightness. In this study, we develop a model capable of accurately controlling the warmth and brightness of an audio source using a single parameter. To do this, we first identify the most salient audio features associated with the chosen descriptors by applying dimensionality reduction to a dataset of annotated timbral transformations. Here, strong positive correlations are found between the centroid of various spectral representations and the most salient principal components. From this, we build a system designed to manipulate the audio features directly using a combination of linear and nonlinear processing modules. To validate the model, we conduct a series of subjective listening tests, and show that up to 80% of participants are able to allocate the correct term, or synonyms thereof, to a set of processed audio samples. Objectively, we show low Mahalanobis distances between the processed samples and clusters of the same timbral adjective in the low-dimensional timbre space.
Download A Feedback Canceling Reverberator A real-time auralization system is described in which room sounds are reverberated and presented over loudspeakers. Room microphones are used to capture room sound sources, with their outputs processed in a canceler to remove the synthetic reverberation also present in the room. Doing so suppresses feedback and gives precise control over the auralization. It also allows freedom of movement and creates a more dynamic acoustic environment for performers or participants in music, theater, gaming, and virtual reality applications. Canceler design methods are discussed, including techniques for handling varying loudspeaker-microphone transfer functions such as would be present in the context of a performance or installation. Tests in a listening room and recital hall show in excess of 20 dB of feedback suppression.
Download A system for data-driven concatenative sound synthesis In speech synthesis, concatenative data-driven synthesis methods prevail. They use a database of recorded speech and a unit selection algorithm that selects the segments that match best the utterance to be synthesized. Transferring these ideas to musical sound synthesis allows a new method of high quality sound synthesis. Usual synthesis methods are based on a model of the sound signal. It is very difficult to build a model that would preserve the entire fine details of sound. Concatenative synthesis achieves this by using actual recordings. This data-driven approach (as opposed to a rule-based approach) takes advantage of the information contained in the many sound recordings. For example, very naturally sounding transitions can be synthesized, since unit selection is aware of the context of the database units. The C ATERPILLAR software system has been developed to allow data-driven concatenative unit selection sound synthesis. It allows high-quality instrument synthesis with high level control, explorative free synthesis from arbitrary sound databases, or resynthesis of a recording with sounds from the database. It is based on the new software-engineering concept of component-oriented software, increasing flexibility and facilitating reuse.