Demo Data from EDBT 2026 / reviewers in the wild / expert
Carlo Tripodi
dblp:128/8236
· DBLP profile ↗
4ranked-venue papers
0as first author
3since 2021 · last 2022
0000-0002-4790-2189ORCID · verified
Domains — the database's venue-derived domains; a paper can count in several
Artificial intelligence and machine learning · 3 · 3 since 2021Graphics, computer vision, multimedia, augmented reality and games · 1
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 · 100% |
Topics — the 3 heaviest of 3, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing › acoustic signal processing
audio equalization |
0.6 | 1 | 2022 | Deep Optimization of Parametric IIR Filters for Audio Equalization · IEEE ACM Trans. Audio Speech Lang. Process. 2022 |
Audio and music processing
microphone array processing |
0.6 | 1 | 2022 | Experimental Analysis and Design Guidelines for Microphone Virtualization in Automotive Scenarios · IEEE ACM Trans. Audio Speech Lang. Process. 2022 |
Audio and music processing › room acoustics
room impulse response |
0.6 | 1 | 2022 | Deep Optimization of Parametric IIR Filters for Audio Equalization · IEEE ACM Trans. Audio Speech Lang. Process. 2022 |
Methods — techniques the papers use, named apart from their topics
regularization · 0.6minimum mean square error · 0.6least mean squares · 0.6deep learning · 0.6biasnet · 0.6backpropagation · 0.6
| Year | Publication | Venue | Position |
|---|---|---|---|
| 2022 | Digital Filters Design for Personal Sound Zones: a Neural ApproachabstractThe creation of Personal Sound Zones (PSZ) is a recent application of digital signal processing that allows differentiating the sound intensity in neighboring regions of space (e.g. a “bright” and a “dark” zone). Given the impulse responses of the environment, digital filters can be designed in order to obtain an attenuation of the signal in the dark zone as a result of the superposition of the filtered IR coming from each loudspeaker. A neural optimization approach was recently shown to enable PSZ by designing digital FIR filters. In this work we propose an improvement of that neural optimization approach using a simpler neural network architecture. Furthermore we extend the method to the design of IIR filters, which is computationally more effective for a real-time implementation. The neural technique is compared with two state-of-the-art methods, analyzing the performance in terms of Acoustic Contrast. Experiments have been performed using a vehicle composed of standard loudspeakers and two speaker arrays, and show that the proposed approach achieves remarkable Acoustic Contrast without sacrificing audio quality. Giovanni Pepe, Leonardo Gabrielli, Stefano Squartini, Carlo Tripodi, Nicolo Strozzi |
IJCNN | 4 |
| 2022 | Experimental Analysis and Design Guidelines for Microphone Virtualization in Automotive ScenariosabstractIn this paper, a performance analysis on the estimation of the so-called observation filter for the Virtual Microphone Technique (VMT) in a realistic automotive environment is presented. A performance comparison between adaptive and fixed observation filter estimation methods, namely Least Mean Square (LMS) and Minimum Mean Square Error (MMSE), respectively, was carried on. Two different experimental setups were implemented on a popular B-segment car. Eight microphones were placed at the monitoring and virtual positions in order to sense environmental acoustic noise propagating within the cabin of the car running at variable speed on a smooth asphalt. Our experimental results show that a large spectral coherence between monitoring and virtual microphone signals indicates a potentially effective and relatively wide-band virtual microphone signal reconstruction. The fixed observation filter estimation method achieves better performance than the adaptive one, guaranteeing remarkable broadband estimation accuracy. Moreover, for each considered setup, design guidelines are proposed to obtain a good trade-off between estimation accuracy and material costs. Alessandro Opinto, Marco Martalò, Alessandro Costalunga, Nicolo Strozzi, Carlo Tripodi, Riccardo Raheli |
IEEE ACM Trans. Audio Speech Lang. Process. | 5 |
| 2022 | Deep Optimization of Parametric IIR Filters for Audio EqualizationabstractThis paper describes a novel Deep Learning method for the design of IIR parametric filters for automatic multipoint audio equalization, that is the task of improving the sound quality of a listening environment at multiple listening points employing multiple loudspeakers. The filters are designed to approximate the inverse of the RIR and achieve almost flat magnitude response. A simple and effective neural architecture, named BiasNet, is proposed to determine the IIR equalizer parameters. This novel architecture is conceived for optimization and, as such, is able to produce optimal IIR equalizer parameters at its output, after training, with no input required. In absence of input, the presence of learnable non-zero bias terms ensures that the network works properly. An output scaling method is used to obtain accurate tuning of the IIR filters center frequency, quality factor and gain. All layers involved in the proposed method are shown to be differentiable, allowing backpropagation to optimize the network weights and achieve, after a number of training iterations, the optimal output according to a given RIR. The parameters are optimized with respect to a loss function based on a spectral distance between the measured and desired magnitude response, and a regularization term is used to keep the same microphone-loudspeaker energy balance after equalization. Two experimental scenarios are employed, a room and a car cabin, with several loudspeakers. The performance of the proposed method improves over the baseline techniques and achieves an almost flat band at a lower computational cost. Giovanni Pepe, Leonardo Gabrielli, Stefano Squartini, Carlo Tripodi, Nicolo Strozzi |
IEEE ACM Trans. Audio Speech Lang. Process. | 4 |
| 2020 | Deep Learning for Individual Listening ZoneabstractA recent trend in car audio systems is the generation of Individual Listening Zones (ILZ), allowing to improve phone call privacy and reduce disturbance to other passengers, without wearing headphones or earpieces. This is generally achieved by using loudspeaker arrays. In this paper, we describe an approach to achieve ILZ exploiting general purpose car loudspeakers and processing the signal through carefully designed Finite Impulse Response (FIR) filters. We propose a deep neural network approach for the design of filters coefficients in order to obtain a so-called bright zone, where the signal is clearly heard, and a dark zone, where the signal is attenuated. Additionally, the frequency response in the bright zone is constrained to be as flat as possible. Numerical experiments were performed taking the impulse responses measured with either one binaural pair or three binaural pairs for each passenger. The results in terms of attenuation and flatness prove the viability of the approach. Giovanni Pepe, Leonardo Gabrielli, Stefano Squartini, Luca Cattani, Carlo Tripodi |
MMSP | 5 |