Demo Data from VLDB 2026 / reviewers in the wild / expert
Bruno S. Masiero
dblp:186/9409 · also Bruno Sanches Masiero
· DBLP profile ↗
4ranked-venue papers
2as first author
2since 2021 · last 2023
0000-0002-2246-4450ORCID · verified
Domains — the database's venue-derived domains; a paper can count in several
Artificial intelligence and machine learning · 2 · 1 first-author · 1 since 2021Graphics, computer vision, multimedia, augmented reality and games · 2 · 1 first-author · 1 since 2021
Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.
| Computer graphics and multimedia
2 papers |
Audio and music processing · 96% Virtual and augmented reality · 4% |
Topics — the 5 heaviest of 5, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing
sound event detection |
0.7 | 1 | 2023 | Sound Events Localization and Detection Using Bio-Inspired Gammatone Filters and Temporal Convolutional Neural Networks · IEEE ACM Trans. Audio Speech Lang. Process. 2023 |
Audio and music processing
audio feature extraction |
0.2 | 1 | 2023 | Sound Events Localization and Detection Using Bio-Inspired Gammatone Filters and Temporal Convolutional Neural Networks · IEEE ACM Trans. Audio Speech Lang. Process. 2023 |
Audio and music processing › spatial audio › binaural reproduction
crosstalk cancellation |
0.2 | 1 | 2014 | A Framework for the Calculation of Dynamic Crosstalk Cancellation Filters · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Audio and music processing
spatial audio |
0.2 | 1 | 2014 | A Framework for the Calculation of Dynamic Crosstalk Cancellation Filters · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Virtual and augmented reality
immersive audio |
0.1 | 1 | 2014 | A Framework for the Calculation of Dynamic Crosstalk Cancellation Filters · IEEE ACM Trans. Audio Speech Lang. Process. 2014 |
Methods — techniques the papers use, named apart from their topics
temporal convolutional network · 0.7data augmentation · 0.7convolutional recurrent neural network · 0.7weighted filter calculation · 0.2minimum-phase regularization · 0.2frequency-domain filter calculation · 0.2
| Year | Publication | Venue | Position |
|---|---|---|---|
| 2023 | Sound Events Localization and Detection Using Bio-Inspired Gammatone Filters and Temporal Convolutional Neural NetworksabstractThe auditory brain circuits are biologically constructed to recognize and localize sounds by encoding a combination of cues that help individuals interpret sounds. The development of computational methods inspired by human capacities has established opportunities for improving machine hearing. Recent studies based on deep learning show that using convolutional recurrent neural networks (CRNNs) is a promising approach for sound event detection and localization in spatial sound. Nevertheless, depending on the sound environment, the performance of these systems is still far from reaching perfect metrics. Therefore, this work intends to boost the performance of state-of-the-art (SOTA) systems by using bio-inspired gammatone auditory filters and intensity vectors (IVs) for the acoustic feature extraction stage, along with the implementation of a temporal convolutional network (TCN) block into a CRNN model, to capture long term dependencies. Three data augmentation techniques are applied to increase the small number of samples in spatial audio datasets. The mentioned stages constitute our proposed Gammatone-based Sound Events Localization and Detection (G-SELD) system, which exceeded the SOTA results on four spatial audio datasets with different levels of acoustical complexity and with up to three sound sources overlapping in time. Karen Gissell Rosero Jacome, Felipe Grijalva, Bruno S. Masiero |
IEEE ACM Trans. Audio Speech Lang. Process. | 3 |
| 2022 | L3DAS22 Challenge: Learning 3D Audio Sources in a Real Office EnvironmentabstractThe L3DAS22 Challenge is aimed at encouraging the development of machine learning strategies for 3D speech enhancement and 3D sound localization and detection in office-like environments. This challenge improves and extends the tasks of the L3DAS21 edition1. We generated a new dataset, which maintains the same general characteristics of L3DAS21 datasets, but with an extended number of data points and adding constrains that improve the baseline model’s efficiency and overcome the major difficulties encountered by the participants of the previous challenge. We updated the baseline model of Task 1, using the architecture that ranked first in the previous challenge edition. We wrote a new supporting API, improving its clarity and ease-of-use. In the end, we present and discuss the results submitted by all participants. L3DAS22 Challenge website: www.l3das.com/icassp2022. Eric Guizzo, Christian Marinoni, Marco Pennese, Xinlei Ren, Xiguang Zheng, Bruno S. Masiero, Aurelio Uncini, Danilo Comminiello |
ICASSP | 7 |
| 2017 | Revisiting the Kronecker Array TransformabstractIt is known that the calculation of a matrix-vector product can be accelerated if this matrix can be recast (or approximated) by the Kronecker product of two smaller matrices. In array signal processing, the manifold matrix can be described as the Kronecker product of two other matrices if the sensor array displays a separable geometry. This forms the basis of the Kronecker Array Transform (KAT), which was previously introduced to speed up the calculations of acoustic images with microphone arrays. If, however, the array has a quasi-separable geometry, e.g., an otherwise separable array with a missing sensor, then the KAT acceleration can no longer be applied. In this letter, we review the definition of the KAT and provide a much simpler derivation that relies on an explicit new relation developed between Kronecker and Khatri-Rao matrix products. Additionally, we extend the KAT to deal with quasi-separable arrays, alleviating the restriction on the need of perfectly separable arrays. Bruno S. Masiero, Vítor H. Nascimento |
IEEE Signal Process. Lett. | 1 |
| 2014 | A Framework for the Calculation of Dynamic Crosstalk Cancellation FiltersabstractDynamic crosstalk cancellation (CTC) systems commonly find use in immersive virtual reality (VR) applications. Such dynamic setups require extremely high filter update rates, so filter calculation is usually performed in the frequency-domain for higher efficiency. This paper proposes a general framework for the calculation of dynamic CTC filters to be used in immersive VR applications. Within this framework, we introduce a causality constraint to the frequency-domain calculation to avoid undesirable wrap-around effects and echo artifacts. Furthermore, when regularization is applied to the CTC filter calculation, in order to limit the output levels at the loudspeakers, noncausal artifacts appear at the CTC filters and the resulting ear signals. We propose a global minimum-phase regularization to convert these anti-causal ringing artifacts into causal artifacts. Finally, an aspect that is especially critical for dynamic CTC systems is the filter switch between active loudspeakers distributed in a surround audio-visual display system with 360°of freedom of operator orientation. Within this framework we apply a weighted filter calculation to control the filter switch, which allows the loudspeakers' contribution to be windowed in space, resulting in a smooth filter transition. Bruno S. Masiero, Michael Vorländer |
IEEE ACM Trans. Audio Speech Lang. Process. | 1 |