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Electronics
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Applications of Audio and Acoustic Signal
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Applications of Audio and Acoustic Signal

A special issue of Electronics (ISSN 2079-9292). This special issue belongs to the section "Computer Science & Engineering".

Deadline for manuscript submissions: closed (30 June 2022) | Viewed by 13697

Special Issue Editor

Dr. Daniele Salvati

E-Mail Website
Guest Editor
Department of Mathematics, Computer Science and Physics, University of Udine, 33100 Udine, Italy
Interests: acoustic signal processing; acoustic sensor array processing; acoustic source localization and tracking; audio beamforming; deep learning methods for acoustic array processing; simultaneous localization and mapping of sources and sensors; binaural hearing; acoustic scene classification; acoustic event detection and classification; human–machine audio interfaces

Special Issue Information

Dear Colleagues,

Sound is, in most cases, an inextricable aspect of the human experience. Humans are constantly immersed in sound fields, and audio and acoustic signals are hence of primary importance in the information and communication technologies that model, reproduce, and mediate our acoustic perception of reality. Applications concerning audio and acoustic signals are rapidly growing and include telecommunication systems, human–computer interaction, musical industries, autonomous systems and robotics, hearing aids, and bioacoustics, to cite a few. In recent years, research activity has focused on numerous topics such as audio and speech recognition, acoustic beamforming and source localization, classification of acoustic scenes, signal enhancement, spatial audio recording and reproduction, music signal analysis, synthesis and modification, music information retrieval, and audio for multimedia.

This Special Issue intends to present original contributions describing advances in applications involving audio and acoustic signals, with an emphasis on recent technology trends. The Special Issue welcomes research papers in the field covering innovative approaches, new improvements, and novel applications.

Dr. Daniele Salvati
Guest Editor

Manuscript Submission Information

Manuscripts should be submitted online at www.mdpi.com by registering and logging in to this website. Once you are registered, click here to go to the submission form. Manuscripts can be submitted until the deadline. All submissions that pass pre-check are peer-reviewed. Accepted papers will be published continuously in the journal (as soon as accepted) and will be listed together on the special issue website. Research articles, review articles as well as short communications are invited. For planned papers, a title and short abstract (about 100 words) can be sent to the Editorial Office for announcement on this website.

Submitted manuscripts should not have been published previously, nor be under consideration for publication elsewhere (except conference proceedings papers). All manuscripts are thoroughly refereed through a single-blind peer-review process. A guide for authors and other relevant information for submission of manuscripts is available on the Instructions for Authors page. Electronics is an international peer-reviewed open access semimonthly journal published by MDPI.

Please visit the Instructions for Authors page before submitting a manuscript. The Article Processing Charge (APC) for publication in this open access journal is 2400 CHF (Swiss Francs). Submitted papers should be well formatted and use good English. Authors may use MDPI's English editing service prior to publication or during author revisions.

Keywords

  • Microphone array applications
  • Spatial and statistical blind source separation
  • Speech signal applications
  • Acoustic signal enhancement and audio restoration
  • Detection and classification of acoustic scenes and events
  • Autonomous system and robot audition applications
  • Music analysis and information retrieval
  • Bioacoustics
  • Audio security

Published Papers (7 papers)

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Research

15 pages, 2032 KiB  
Article
Very Simple System for Walking-Speed Measurement in Geriatric Patients
by Graziella Scandurra, Giorgio Basile and Carmine Ciofi
Electronics 2022, 11(19), 3159; https://doi.org/10.3390/electronics11193159 - 1 Oct 2022
Viewed by 1094
Abstract
Walking speed in geriatric patients is an important index for inferring the patient’s state of health and estimating the success rate of some surgical procedures. Although different solutions for monitoring the gait of a subject exist in scientific literature and on the market, [...] Read more.
Walking speed in geriatric patients is an important index for inferring the patient’s state of health and estimating the success rate of some surgical procedures. Although different solutions for monitoring the gait of a subject exist in scientific literature and on the market, there is a need for a system that is very simple, especially to wear, considering that elderly subjects often have movement difficulties. For this reason, we investigated the possibility of using a standard miniaturized wireless microphone, that can be easily attached to patients’ clothes by means of a clip, as the sole sensing device to be worn by the test subject. A transceiver, a sound card and a PC complete the system, which turns out to be quite simple to be set up and use, thanks to a proper graphic user interface that controls its entire operation. The system essentially tracks the position of the test subject over time by measuring the propagation times of repeated sound pulses from the speaker to the microphone. To avoid hearing discomfort, the frequency of the pulses is chosen at the higher end of the audio spectrum, so that they are essentially undetectable by adults. The measurement range is in excess of 6 m, that is sufficient for the standard 4 m walking-speed test. Tests performed in a laboratory environment have confirmed the effectiveness of the approach we propose. Full article
(This article belongs to the Special Issue Applications of Audio and Acoustic Signal)
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15 pages, 796 KiB  
Article
Low-Complexity Acoustic Scene Classification Using Time Frequency Separable Convolution
by Duc H. Phan and Douglas L. Jones
Electronics 2022, 11(17), 2734; https://doi.org/10.3390/electronics11172734 - 30 Aug 2022
Viewed by 1330
Abstract
Replacing 2D-convolution operations by depth-wise separable time and frequency convolutions greatly reduces the number of parameters while maintaining nearly equivalent performances in the context of acoustic scene classification. In our experiments, the models’ sizes can be reduced by 6 to 14 times with [...] Read more.
Replacing 2D-convolution operations by depth-wise separable time and frequency convolutions greatly reduces the number of parameters while maintaining nearly equivalent performances in the context of acoustic scene classification. In our experiments, the models’ sizes can be reduced by 6 to 14 times with similar performances. For a 3-class audio classification, replacing 2D-convolution in a CNN model gives roughly a 2% increase in accuracy. In a 10-class audio classification with multiple recording devices, replacing 2D-convolution in Resnet only reduces around 1.5% of the accuracy. Full article
(This article belongs to the Special Issue Applications of Audio and Acoustic Signal)
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16 pages, 2972 KiB  
Article
Method for Direct Localization of Multiple Impulse Acoustic Sources in Outdoor Environment
by Milan Mišković, Nenad Vukmirović, Dragan Golubović and Miljko Erić
Electronics 2022, 11(16), 2509; https://doi.org/10.3390/electronics11162509 - 11 Aug 2022
Cited by 1 | Viewed by 1547
Abstract
A method for the direct outdoor localization of multiple impulse acoustic sources by a distributed microphone array is proposed. This localization problem is of great interest for gunshot, firecracker and explosion detection localization in a civil environment, as well as for gun, mortar, [...] Read more.
A method for the direct outdoor localization of multiple impulse acoustic sources by a distributed microphone array is proposed. This localization problem is of great interest for gunshot, firecracker and explosion detection localization in a civil environment, as well as for gun, mortar, small arms, artillery, sniper detection localization in military battlefield monitoring systems. Such a kind of localization is a complicated technical problem in many aspects. In such a scenario, the permutation of impulse arrivals on distributed microphones occurs, so the application of classical two-step localization methods, such as time-of-arrival (TOA), time-difference-of-arrival (TDOA), angle-of-arrival (AOA), fingerprint methods, etc., is faced with the so-called association problem, which is difficult to solve. The association problem does not exist in the proposed method for direct (one-step) localization, so the proposed method is more suitable for localization in a given acoustic scenario than the mentioned two-step localization methods. Furthermore, in the proposed method, direct localization is performed impulse by impulse. The observation interval used for the localization could not be arbitrarily chosen; it is limited by the duration of impulses. In the mathematical model formulated in the paper, atmospheric factors in acoustic signal propagation (temperature, pressure, etc.) are included. The results of simulations show that by using the proposed method, centimeter localization accuracy can be achieved. Full article
(This article belongs to the Special Issue Applications of Audio and Acoustic Signal)
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22 pages, 3468 KiB  
Article
Two Sound Field Control Methods Based on Particle Velocity
by Song Wang and Cong Zhang
Electronics 2022, 11(14), 2275; https://doi.org/10.3390/electronics11142275 - 21 Jul 2022
Cited by 1 | Viewed by 1315
Abstract
In recent years, a variety of sound field control methods have been proposed for the generation of separated sound regions. Different algorithms control the physical properties of the generated sound field to different degrees. The existing methods mainly focus on sound pressure restoration [...] Read more.
In recent years, a variety of sound field control methods have been proposed for the generation of separated sound regions. Different algorithms control the physical properties of the generated sound field to different degrees. The existing methods mainly focus on sound pressure restoration and its related improvement. When the loudspeaker array is non-uniformly placed, the reconstruction system is not stable enough. To solve this problem, this paper proposes two sound field control methods related to particle velocity. The first method regulates the reconstruction error of particle velocity in the bright zone and the square of particle velocity in the dark zone; the second method regulates the reconstruction error of sound pressure and particle velocity in the bright zone and the square of sound pressure and particle velocity in the dark zone. Five channel and twenty-two channel non-uniform loudspeaker systems were used for two-dimensional and three-dimensional computer simulation testing. Experimental results show that the two proposed methods have better tradeoffs in terms of acoustic contrast, reproduction error and array effort than traditional methods, especially the second proposed method. In the two-dimensional experiment, the maximum reductions of the average array efforts generated by the proposed methods were about 10 dB and 11 dB compared with the average array efforts generated by two traditional methods. In the three-dimensional experiment, the maximum reductions of the average array efforts generated by the proposed methods were about 8 dB and 2 dB compared with the average array efforts generated by two traditional methods. The smaller the array effort, the more stable the loudspeaker system. Therefore, the reconstruction systems produced by the proposed methods are more stable than those produced by the traditional methods. Full article
(This article belongs to the Special Issue Applications of Audio and Acoustic Signal)
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12 pages, 501 KiB  
Article
Quality Enhancement of MPEG-H 3DA Binaural Rendering Using a Spectral Compensation Technique
by Hyeongi Moon and Young-cheol Park
Electronics 2022, 11(9), 1491; https://doi.org/10.3390/electronics11091491 - 6 May 2022
Viewed by 1589
Abstract
The latest MPEG standard, MPEG-H 3D Audio, employs the virtual loudspeaker rendering (VLR) technique to support virtual reality (VR) and augmented reality (AR). During the rendering, the binaural downmixing of channel signals often induces the so-called comb filter effect, an undesirable spectral artifact, [...] Read more.
The latest MPEG standard, MPEG-H 3D Audio, employs the virtual loudspeaker rendering (VLR) technique to support virtual reality (VR) and augmented reality (AR). During the rendering, the binaural downmixing of channel signals often induces the so-called comb filter effect, an undesirable spectral artifact, due to the phase difference between the binaural filters. In this paper, we propose an efficient algorithm that can mitigate such spectral artifacts. The proposed algorithm performs spectral compensation in both the panning gain and downmix signal domains depending on the frequency range. In the low-frequency bands where a band has a wider bandwidth than the critical-frequency scale, panning gains are directly compensated. In the high-frequency bands, where a band has a narrower bandwidth than the critical-frequency scale, a signal compensation similar to the active downmix is performed. As a result, the proposed algorithm optimizes the performance and the complexity within MPEG-H 3DA framework. By implementing the algorithm on MPEG-H 3DA BR, we verify that the additional computation complexity is minor. We also show that the proposed algorithm improves the subjective quality of MPEG-H 3DA BR significantly. Full article
(This article belongs to the Special Issue Applications of Audio and Acoustic Signal)
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24 pages, 7968 KiB  
Article
Acoustic Descriptors for Characterization of Musical Timbre Using the Fast Fourier Transform
by Yubiry Gonzalez and Ronaldo C. Prati
Electronics 2022, 11(9), 1405; https://doi.org/10.3390/electronics11091405 - 27 Apr 2022
Cited by 6 | Viewed by 3062
Abstract
The quantitative assessment of the musical timbre in an audio record is still an open-ended issue. Evaluating the musical timbre allows not only to establish precise musical parameters but also the recognition, classification of musical instruments, and assessment of the musical quality of [...] Read more.
The quantitative assessment of the musical timbre in an audio record is still an open-ended issue. Evaluating the musical timbre allows not only to establish precise musical parameters but also the recognition, classification of musical instruments, and assessment of the musical quality of a sound record. In this paper, we present a minimum set of dimensionless descriptors, motivated by musical acoustics, using the spectra obtained by the Fast Fourier Transform (FFT), which allows describing the timbre of wooden aerophones (Bassoon, Clarinet, Transverse Flute, and Oboe) using individual sound recordings of the musical tempered scale. We postulate that the proposed descriptors are sufficient to describe the timbral characteristics in the aerophones studied, allowing their recognition using the acoustic spectral signature. We believe that this approach can be further extended to use multidimensional unsupervised machine learning techniques, such as clustering, to obtain new insights into timbre characterization. Full article
(This article belongs to the Special Issue Applications of Audio and Acoustic Signal)
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12 pages, 1398 KiB  
Article
Estimation of Azimuth and Elevation for Multiple Acoustic Sources Using Tetrahedral Microphone Arrays and Convolutional Neural Networks
by Saulius Sakavičius and Artūras Serackis
Electronics 2021, 10(21), 2585; https://doi.org/10.3390/electronics10212585 - 22 Oct 2021
Cited by 6 | Viewed by 2151
Abstract
A method for multiple acoustic source localization using a tetrahedral microphone array and a convolutional neural network (CNN) is presented. Our method presents a novel approach for the estimation of acoustic source direction of arrival (DoA), both azimuth and elevation, utilizing a non-coplanar [...] Read more.
A method for multiple acoustic source localization using a tetrahedral microphone array and a convolutional neural network (CNN) is presented. Our method presents a novel approach for the estimation of acoustic source direction of arrival (DoA), both azimuth and elevation, utilizing a non-coplanar microphone array. In our approach, we use the phase component of the short-time Fourier transform (STFT) of the microphone array’s signals as the input feature for the CNN and a DoA probability density map as the training target. Our findings imply that our method outperforms the currently available methods for multiple sound source DoA estimation in both accuracy and speed. Full article
(This article belongs to the Special Issue Applications of Audio and Acoustic Signal)
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