Augusto Sarti

Prof. Dr. Augusto Sarti


Research Focus & Metrics

Academic Career Length

27Years (2000-Present)

(Calculated once Scilit publication data is synced: current year − year of first publication + 1)

Representative Publications

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1 April 2020 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
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Journal Article
Closed access
MCMaximo Cobos
Maximo Cobos
FAFabio Antonacci
Fabio Antonacci
LCLuca Comanducci
Luca Comanducci
ASAugusto Sarti
Augusto Sarti
The generalized cross-correlation (GCC) is regarded as the most popular approach for estimating the time difference of arrival (TDOA) between the signals received at two sensors. Time delay estimates are obtained by maximizing the GCC output, where the direct-path delay is usually observed as a prominent peak. Moreover, GCCs play also an important role in steered response power (SRP) localization algorithms, where the SRP functional can be written as an accumulation of the GCCs computed from multiple sensor pairs. Unfortunately, the accuracy of TDOA estimates is affected by multiple factors, including noise, reverberation and signal bandwidth. In this paper, a sub-band approach for time delay estimation aimed at improving the performance of the conventional GCC is presented. The proposed method is based on the extraction of multiple GCCs corresponding to different frequency bands of the cross-power spectrum phase in a sliding-window fashion. The major contributions of this paper include: 1) a sub-band GCC representation of the cross-power spectrum phase that, despite having a reduced temporal resolution, provides a more suitable representation for estimating the true TDOA; 2) such matrix representation is shown to be rank one in the ideal noiseless case, a property that is exploited in more adverse scenarios to obtain a more robust and accurate GCC; 3) we propose a set of low-rank approximation alternatives for processing the sub-band GCC matrix, leading to better TDOA estimates and source localization performance. An extensive set of experiments is presented to demonstrate the validity of the proposed approach.

25 November 2021 in Sensors
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Journal Article
Open access
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MOMarco Olivieri
Marco Olivieri
MPMirco Pezzoli
Mirco Pezzoli
FAFabio Antonacci
Fabio Antonacci
ASAugusto Sarti
Augusto Sarti
In this manuscript, we describe a novel methodology for nearfield acoustic holography (NAH). The proposed technique is based on convolutional neural networks, with autoencoder architecture, to reconstruct the pressure and velocity fields on the surface of the vibrating structure using the sampled pressure soundfield on the holographic plane as input. The loss function used for training the network is based on a combination of two components. The first component is the error in the reconstructed velocity. The second component is the error between the sound pressure on the holographic plane and its estimate obtained from forward propagating the pressure and velocity fields on the structure through the Kirchhoff–Helmholtz integral; thus, bringing some knowledge about the physics of the process under study into the estimation algorithm. Due to the explicit presence of the Kirchhoff–Helmholtz integral in the loss function, we name the proposed technique the Kirchhoff–Helmholtz-based convolutional neural network, KHCNN. KHCNN has been tested on two large datasets of rectangular plates and violin shells. Results show that it attains very good accuracy, with a gain in the NMSE of the estimated velocity field that can top 10 dB, with respect to state-of-the-art techniques. The same trend is observed if the normalized cross correlation is used as a metric.

1 April 2022 in Sensors
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Journal Article
Open access
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MPMirco Pezzoli
Mirco Pezzoli
DPDavide Perini
Davide Perini
ABAlberto Bernardini
Alberto Bernardini
FBFederico Borra
Federico Borra
FAFabio Antonacci
Fabio Antonacci
ASAugusto Sarti
Augusto Sarti
In this paper, we propose a data-driven approach for the reconstruction of unknown room impulse responses (RIRs) based on the deep prior paradigm. We formulate RIR reconstruction as an inverse problem. More specifically, a convolutional neural network (CNN) is employed prior, in order to obtain a regularized solution to the RIR reconstruction problem for uniform linear arrays. This approach allows us to avoid assumptions on sound wave propagation, acoustic environment, or measuring setting made in state-of-the-art RIR reconstruction algorithms. Moreover, differently from classical deep learning solutions in the literature, the deep prior approach employs a per-element training. Therefore, the proposed method does not require training data sets, and it can be applied to RIRs independently from available data or environments. Results on simulated data demonstrate that the proposed technique is able to provide accurate results in a wide range of scenarios, including variable direction of arrival of the source, room T60, and SNR at the sensors. The devised technique is also applied to real measurements, resulting in accurate RIR reconstruction and robustness to noise compared to state-of-the-art solutions.

4 May 2017 in IEEE Transactions on Signal Processing
Verified
Journal Article
Closed access
MCMarco Compagnoni
Marco Compagnoni
APAlessia Pini
Alessia Pini
ACAntonio Canclini
Antonio Canclini
PBPaolo Bestagini
Paolo Bestagini
FAFabio Antonacci
Fabio Antonacci
ASAugusto Sarti
Augusto Sarti
The curse of outlier measurements in estimation problems is a well known issue in a variety of fields. Therefore, outlier removal procedures, which enables the identification of spurious measurements within a set, have been developed for many different scenarios and applications. In this paper, we propose a statistically motivated outlier removal algorithm for time differences of arrival (TDOAs), or equivalently range differences (RD), acquired at sensor arrays. The method exploits the TDOAspace formalism and works by only knowing relative sensor positions. As the proposed method is completely independent from the application for which measurements are used, it can be reliably used to identify outliers within a set of TDOA/RD measurements in different fields (e.g. acoustic source localization, sensor synchronization, radar, remote sensing, etc.). The proposed outlier removal algorithm is validated by means of synthetic simulations and real experiments.

28 May 2020 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
FBFederico Borra
Federico Borra
ABAlberto Bernardini
Alberto Bernardini
FAFabio Antonacci
Fabio Antonacci
ASAugusto Sarti
Augusto Sarti
We present a spatial filtering approach to first-order steerable Differential Microphone Arrays (DMAs) with arbitrary planar geometry. In particular, the design of the spatial filter is based on a recently proposed frequency-domain design methodology that approximates, in a least-square sense, a target beampattern using the Jacobi-Anger expansion involving Bessel functions. Despite the generality of that approach, however, its computational cost turns out to be excessive when working with limited processing resources. The beamforming technique proposed in this manuscript overcomes this issue by exploiting the fact that in DMAs the spacing between sensors is typically smaller than the smallest wavelength of audio signals of interest. This allows us to substitute zero- and first-order Bessel functions with their Taylor series approximation truncated to the first order. Moreover, we show that this approximation allows us to derive an efficient discrete-time-domain implementation of first-order steerable differential beamformers based on arrays with arbitrary geometries.

28 July 2020 in IEEE/ACM Transactions on Audio, Speech, and Language Processing
Verified
Journal Article
Closed access
MPMirco Pezzoli
Mirco Pezzoli
FBFederico Borra
Federico Borra
FAFabio Antonacci
Fabio Antonacci
STStefano Tubaro
Stefano Tubaro
ASAugusto Sarti
Augusto Sarti
In this manuscript we propose a methodology for the reconstruction of sound fields in arbitrary locations based on the signals acquired by a spatial distribution of compact microphone arrays (virtual miking). The proposed method is suitable for operating in reverberant environments, thanks to a two-stage analysis process, the former of which aims at separating the direct and the diffuse components of the sound field. The method that we propose is inherently parametric, as the sources of the acoustic scene are characterized by parameters describing location and directivity (spherical harmonics expansion), which are extracted from the exterior model of the direct component of the sound field. Once the parameters of the sources are extracted, the direct sound field at an arbitrary location is reconstructed. The diffuse component is reconstructed from the joint knowledge of the diffuse component at the locations of the distributed microphone arrays, under the assumption of isotropic behavior. Results show that the proposed technique is able to analyze the sound field and reconstruct the parameters of the sources that are active in the scene. In addition, the synthesis of the signals at the virtual microphone locations turns out to accurately match (in terms of spatial cues) the actual sound field, as measured by a microphone places in the desired location.