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Advanced Technology in Acoustic Signal Processing
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Advanced Technology in Acoustic Signal Processing

A special issue of Sensors (ISSN 1424-8220). This special issue belongs to the section "Physical Sensors".

Deadline for manuscript submissions: closed (31 March 2024) | Viewed by 16347

Special Issue Editor

Dr. Alberto Bernardini

E-Mail Website
Guest Editor
Assistant Professor, Department of Electronics, Information and Bioengineering (DEIB) of the Polytechnic of Milan, Milan, Italy
Interests: digital signal processing; network theory; wave digital filters; audio signal processing; computational acoustics

Special Issue Information

Dear Colleagues,

Recent technological advancements in the field of audio transducers, along with the evolution of computational acoustics, are giving rise to a variety of acoustic signal processing applications that would have been unthinkable until a few years ago.

In particular, small-size transducers such as MEMS, piezoelectric microphones, and loudspeakers can be combined into arrays in order to develop new methods for soundfield analysis and rendering.

For example, small-size microphone arrays can be used to design almost-frequency-invariant spatial filtering techniques based on differential beamforming theory.

As a further example, parametric acoustic arrays exploit nonlinear acoustic theory to generate high-directivity and low-frequency broadband signals with small-aperture transducers.

In addition to the undeniable advantages, small-size acoustic devices are often characterized by some non-ideal qualities, such as distortion effects at low frequencies. Moreover, phase and amplitude mismatches due to unavoidable fabrication errors need to be taken into account when designing acoustic signal processing methods based on arrays of transducers.

Several analog and digital signal processing methods are currently being studied with the aim to mitigate these problems.

Application scenarios of modern acoustic signal processing include underwater acoustic engineering, multi-analyte detection, assisted diagnosis, touch sensing, chemical and biochemical sensing, sensing of the level and composition of liquids, temperature sensing, stress and torque sensing, automotive mobiles, infotainment applications, teleconferencing applications, and wireless remotes.

This Special Issue aims to bring together the research on acoustic signal processing methods for single transducers or arrays of transducers and thereby address the wide range of applications of sound signal sensors. Both review articles and original research papers related to acoustic signal processing are welcome.

Dr. Alberto Bernardini
Guest Editor

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Please visit the Instructions for Authors page before submitting a manuscript. The Article Processing Charge (APC) for publication in this open access journal is 2600 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

  • acoustic signal processing
  • acoustic signal sensors
  • MEMS microphones
  • MEMS loudspeakers
  • piezoelectric acoustic transducers
  • microphone arrays
  • parametric acoustic array
  • beamforming methods
  • parametric acoustic array
  • nonlinear acoustics
  • compensation of transducer nonidealities
  • audio circuits

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Published Papers (7 papers)

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Research

21 pages, 2476 KiB  
Article
Accurate and Low-Power Ultrasound–Radiofrequency (RF) Indoor Ranging Using MEMS Loudspeaker Arrays
by Chesney Buyle, Lieven De Strycker and Liesbet Van der Perre
Sensors 2023, 23(18), 7997; https://doi.org/10.3390/s23187997 - 20 Sep 2023
Viewed by 1883
Abstract
Accurately positioning energy-constrained devices in indoor environments is of great interest to many professional, care, and personal applications. Hybrid RF–acoustic ranging systems have shown to be a viable technology in this regard, enabling accurate distance measurements at ultra-low energy costs. However, they often [...] Read more.
Accurately positioning energy-constrained devices in indoor environments is of great interest to many professional, care, and personal applications. Hybrid RF–acoustic ranging systems have shown to be a viable technology in this regard, enabling accurate distance measurements at ultra-low energy costs. However, they often suffer from self-interference due to multipaths in indoor environments. We replace the typical single loudspeaker beacons used in these systems with a phased loudspeaker array to promote the signal-to-interference-plus-noise ratio towards the tracked device. Specifically, we optimize the design of a low-cost uniform planar array (UPA) through simulation to achieve the best ranging performance using ultrasonic chirps. Furthermore, we compare the ranging performance of this optimized UPA configuration to a traditional, single-loudspeaker system. Simulations show that vertical phased-array configurations guarantee the lowest ranging errors in typical shoe-box environments, having a limited height with respect to their length and width. In these cases, a P50 ranging error of around 3 cm and P95 ranging error below 30 cm were achieved. Compared to a single-speaker system, a 10 × 2 vertical phased array was able to lower the P80 and P95 up to an order of magnitude. Full article
(This article belongs to the Special Issue Advanced Technology in Acoustic Signal Processing)
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14 pages, 2751 KiB  
Article
Detection of Audio Tampering Based on Electric Network Frequency Signal
by Hsiang-Ping Hsu, Zhong-Ren Jiang, Lo-Ya Li, Tsai-Chuan Tsai, Chao-Hsiang Hung, Sheng-Chain Chang, Syu-Siang Wang and Shih-Hau Fang
Sensors 2023, 23(16), 7029; https://doi.org/10.3390/s23167029 - 8 Aug 2023
Cited by 6 | Viewed by 2212
Abstract
The detection of audio tampering plays a crucial role in ensuring the authenticity and integrity of multimedia files. This paper presents a novel approach to identifying tampered audio files by leveraging the unique Electric Network Frequency (ENF) signal, which is inherent to the [...] Read more.
The detection of audio tampering plays a crucial role in ensuring the authenticity and integrity of multimedia files. This paper presents a novel approach to identifying tampered audio files by leveraging the unique Electric Network Frequency (ENF) signal, which is inherent to the power grid and serves as a reliable indicator of authenticity. The study begins by establishing a comprehensive Chinese ENF database containing diverse ENF signals extracted from audio files. The proposed methodology involves extracting the ENF signal, applying wavelet decomposition, and utilizing the autoregressive model to train effective classification models. Subsequently, the framework is employed to detect audio tampering and assess the influence of various environmental conditions and recording devices on the ENF signal. Experimental evaluations conducted on our Chinese ENF database demonstrate the efficacy of the proposed method, achieving impressive accuracy rates ranging from 91% to 93%. The results emphasize the significance of ENF-based approaches in enhancing audio file forensics and reaffirm the necessity of adopting reliable tamper detection techniques in multimedia authentication. Full article
(This article belongs to the Special Issue Advanced Technology in Acoustic Signal Processing)
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22 pages, 3002 KiB  
Article
On the Virtualization of Audio Transducers
by Riccardo Giampiccolo, Alberto Bernardini, Oliviero Massi and Augusto Sarti
Sensors 2023, 23(11), 5258; https://doi.org/10.3390/s23115258 - 1 Jun 2023
Cited by 4 | Viewed by 1842
Abstract
In audio transduction applications, virtualization can be defined as the task of digitally altering the acoustic behavior of an audio sensor or actuator with the aim of mimicking that of a target transducer. Recently, a digital signal preprocessing method for the virtualization of [...] Read more.
In audio transduction applications, virtualization can be defined as the task of digitally altering the acoustic behavior of an audio sensor or actuator with the aim of mimicking that of a target transducer. Recently, a digital signal preprocessing method for the virtualization of loudspeakers based on inverse equivalent circuit modeling has been proposed. The method applies Leuciuc’s inversion theorem to obtain the inverse circuital model of the physical actuator, which is then exploited to impose a target behavior through the so called Direct–Inverse–Direct Chain. The inverse model is designed by properly augmenting the direct model with a theoretical two-port circuit element called nullor. Drawing on this promising results, in this manuscript, we aim at describing the virtualization task in a broader sense, including both actuator and sensor virtualizations. We provide ready-to-use schemes and block diagrams which apply to all the possible combinations of input and output variables. We then analyze and formalize different versions of the Direct–Inverse–Direct Chain describing how the method changes when applied to sensors and actuators. Finally, we provide examples of applications considering the virtualization of a capacitive microphone and a nonlinear compression driver. Full article
(This article belongs to the Special Issue Advanced Technology in Acoustic Signal Processing)
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17 pages, 3474 KiB  
Communication
Listen to the Brain–Auditory Sound Source Localization in Neuromorphic Computing Architectures
by Daniel Schmid, Timo Oess and Heiko Neumann
Sensors 2023, 23(9), 4451; https://doi.org/10.3390/s23094451 - 2 May 2023
Cited by 3 | Viewed by 2425
Abstract
Conventional processing of sensory input often relies on uniform sampling leading to redundant information and unnecessary resource consumption throughout the entire processing pipeline. Neuromorphic computing challenges these conventions by mimicking biology and employing distributed event-based hardware. Based on the task of lateral auditory [...] Read more.
Conventional processing of sensory input often relies on uniform sampling leading to redundant information and unnecessary resource consumption throughout the entire processing pipeline. Neuromorphic computing challenges these conventions by mimicking biology and employing distributed event-based hardware. Based on the task of lateral auditory sound source localization (SSL), we propose a generic approach to map biologically inspired neural networks to neuromorphic hardware. First, we model the neural mechanisms of SSL based on the interaural level difference (ILD). Afterward, we identify generic computational motifs within the model and transform them into spike-based components. A hardware-specific step then implements them on neuromorphic hardware. We exemplify our approach by mapping the neural SSL model onto two platforms, namely the IBM TrueNorth Neurosynaptic System and SpiNNaker. Both implementations have been tested on synthetic and real-world data in terms of neural tunings and readout characteristics. For synthetic stimuli, both implementations provide a perfect readout (100% accuracy). Preliminary real-world experiments yield accuracies of 78% (TrueNorth) and 13% (SpiNNaker), RMSEs of 41∘ and 39∘, and MAEs of 18∘ and 29∘, respectively. Overall, the proposed mapping approach allows for the successful implementation of the same SSL model on two different neuromorphic architectures paving the way toward more hardware-independent neural SSL. Full article
(This article belongs to the Special Issue Advanced Technology in Acoustic Signal Processing)
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16 pages, 1804 KiB  
Article
Improved Speech Spatial Covariance Matrix Estimation for Online Multi-Microphone Speech Enhancement
by Minseung Kim, Sein Cheong, Hyungchan Song and Jong Won Shin
Sensors 2023, 23(1), 111; https://doi.org/10.3390/s23010111 - 22 Dec 2022
Cited by 2 | Viewed by 2554
Abstract
Online multi-microphone speech enhancement aims to extract target speech from multiple noisy inputs by exploiting the spatial information as well as the spectro-temporal characteristics with low latency. Acoustic parameters such as the acoustic transfer function and speech and noise spatial covariance matrices (SCMs) [...] Read more.
Online multi-microphone speech enhancement aims to extract target speech from multiple noisy inputs by exploiting the spatial information as well as the spectro-temporal characteristics with low latency. Acoustic parameters such as the acoustic transfer function and speech and noise spatial covariance matrices (SCMs) should be estimated in a causal manner to enable the online estimation of the clean speech spectra. In this paper, we propose an improved estimator for the speech SCM, which can be parameterized with the speech power spectral density (PSD) and relative transfer function (RTF). Specifically, we adopt the temporal cepstrum smoothing (TCS) scheme to estimate the speech PSD, which is conventionally estimated with temporal smoothing. Furthermore, we propose a novel RTF estimator based on a time difference of arrival (TDoA) estimate obtained by the cross-correlation method. Furthermore, we propose refining the initial estimate of speech SCM by utilizing the estimates for the clean speech spectrum and clean speech power spectrum. The proposed approach showed superior performance in terms of the perceptual evaluation of speech quality (PESQ) scores, extended short-time objective intelligibility (eSTOI), and scale-invariant signal-to-distortion ratio (SISDR) in our experiments on the CHiME-4 database. Full article
(This article belongs to the Special Issue Advanced Technology in Acoustic Signal Processing)
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9 pages, 3024 KiB  
Communication
Three-Dimensional Acoustic Device for Testing the All-Directional Anisotropic Characteristics of Rock Samples
by Kai Zhang, Shengqing Li, Yuanda Su, Baohai Tan, Wenjie Wu and Shoutao Xin
Sensors 2022, 22(23), 9473; https://doi.org/10.3390/s22239473 - 4 Dec 2022
Cited by 1 | Viewed by 1683
Abstract
Many oil and gas fields, especially non-conventional shale and compacted sand reservoirs, have formation anisotropy. The acoustic anisotropy measurement of cores in these reservoirs can guide drilling, well logging, and exploitation. However, almost all core holders are designed for cylinder cores, which are [...] Read more.
Many oil and gas fields, especially non-conventional shale and compacted sand reservoirs, have formation anisotropy. The acoustic anisotropy measurement of cores in these reservoirs can guide drilling, well logging, and exploitation. However, almost all core holders are designed for cylinder cores, which are not suitable for all-directional measurements. A three-dimensional measurement device was designed on the basis of the cross-hole sonic logging method. This device mainly consisted of two pairs of transducers, a signal generator, an oscillograph, an omnidirectional positioning system, and a computer control system. By adjusting the measurement latitude and longitude circle automatically, this device scanned spherical sample rocks and obtained full-wave waveforms in all directions. Experiments were performed taking granite from the Jiaodong Peninsula, China, as an example, and the arrival times and velocities of the longitudinal and shear waves were calculated based on the full-wave waveforms. Thereafter, anisotropic physical characterizations were carried out on the basis of these velocities. These data play an important role in guiding formation fracturing and analyzing the stability of borehole walls. Full article
(This article belongs to the Special Issue Advanced Technology in Acoustic Signal Processing)
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11 pages, 1261 KiB  
Communication
Performance Analysis of Active Structural Acoustic Control Applied to a Washing Machine
by Stanislaw Wrona
Sensors 2022, 22(19), 7357; https://doi.org/10.3390/s22197357 - 28 Sep 2022
Cited by 4 | Viewed by 2166
Abstract
Great efforts are undertaken by scientists and manufacturers to reduce noise generated by devices present in the human environment. This task is particularly difficult in the case of low-frequency noise; however, active noise reduction methods can be a solution to such problems. The [...] Read more.
Great efforts are undertaken by scientists and manufacturers to reduce noise generated by devices present in the human environment. This task is particularly difficult in the case of low-frequency noise; however, active noise reduction methods can be a solution to such problems. The purpose of this article is to present the active noise-controlling casing method applied to a washing machine, whose noise generated during the spinning process is to be reduced. The paper presents a set of experimental measurements and analyses. It also provides hardware and software configuration details. The employed active control system can efficiently reduce noise, even by 16 dBA in the low-frequency range up to 300 Hz, which is an important step towards making the technology ready for manufacturing. The conclusions drawn are valid for different active noise-control applications, considering a variety of devices. Full article
(This article belongs to the Special Issue Advanced Technology in Acoustic Signal Processing)
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