The computing performance optimization of the Short-Lag Spatial Coherence (SLSC) method applied to ultrasound data processing is presented. The method is based on the theory that signals from adjacent receivers are correlated, drawing on a simplified conclusion of the van Cittert-Zernike theorem. It has been proven that it can be successfully used in ultrasound data reconstruction with despeckling. Former works have shown that the SLSC method in its original form has two main drawbacks: time-consuming processing and low contrast in the area near the transceivers. In this study, we introduce a method that allows to overcome both of these drawbacks.

The presented approach removes the dependency on distance (the “lag” parameter value) between signals used to calculate correlations. The approach has been tested by comparing results obtained with the original SLSC algorithm on data acquired from tissue phantoms.

The modified method proposed here leads to constant complexity, thus execution time is independent of the lag parameter value, instead of the linear complexity. The presented approach increases computation speed over 10 times in comparison to the base SLSC algorithm for a typical lag parameter value. The approach also improves the output image quality in shallow areas and does not decrease quality in deeper areas.

The Open Seminar on Acoustics is an annual conference, the largest acoustics conference in the country. It has been bringing all Polish acousticians together for over sixty years. It is organized in turns by different divisions of Polish Acoustical Society – in 2019 by the Poznan Division with the Institute of Acoustics, Adam Mickiewicz University in Poznan and Committee on Acoustics of Polish Academy of Science. The conference presents all sections of acoustics, such as: physical acoustics, technical, environmental, speech, hearing, musical, architectural acoustics, etc. The seminar is joined with special session “New trends in psychoacoustics in tribute to professors: Józef Zwisłocki and Andrzej Rakowski” and the Workshop “Noise protection in regulations – current state and directions of changes” (in Polish). We also invite you to the special session “Advances in research in the field of audio acoustics and sound engineering – ISSET 2019”.

The human voice is one of the basic means of communication, thanks to which one also can easily convey the emotional state. This paper presents experiments on emotion recognition in human speech based on the fundamental frequency. AGH Emotional Speech Corpus was used. This database consists of audio samples of seven emotions acted by 12 different speakers (6 female and 6 male). We explored phrases of all the emotions – all together and in various combinations. Fast Fourier Transformation and magnitude spectrum analysis were applied to extract the fundamental tone out of the speech audio samples. After extraction of several statistical features of the fundamental frequency, we studied if they carry information on the emotional state of the speaker applying different AI methods. Analysis of the outcome data was conducted with classifiers: K-Nearest Neighbours with local induction, Random Forest, Bagging, JRip, and Random Subspace Method from algorithms collection for data mining WEKA. The results prove that the fundamental frequency is a prospective choice for further experiments.

The most challenging in speech enhancement technique is tracking non-stationary noises for long speech segments and low Signal-to-Noise Ratio (SNR). Different speech enhancement techniques have been proposed but, those techniques were inaccurate in tracking highly non-stationary noises. As a result, Empirical Mode Decomposition and Hurst-based (EMDH) approach is proposed to enhance the signals corrupted by non-stationary acoustic noises. Hurst exponent statistics was adopted for identifying and selecting the set of Intrinsic Mode Functions (IMF) that are most affected by the noise components. Moreover, the speech signal was reconstructed by considering the least corrupted IMF. Though it increases SNR, the time and resource consumption were high. Also, it requires a significant improvement under nonstationary noise scenario. Hence, in this article, EMDH approach is enhanced by using Sliding Window (SW) technique. In this SWEMDH approach, the computation of EMD is performed based on the small and sliding window along with the time axis. The sliding window depends on the signal frequency band. The possible discontinuities in IMF between windows are prevented by the total number of modes and the number of sifting iterations that should be set a priori. For each module, the number of sifting iterations is determined by decomposition of many signal windows by standard algorithm and calculating the average number of sifting steps for each module. Based on this approach, the time complexity is reduced significantly with suitable quality of decomposition. Finally, the experimental results show the considerable improvements in speech enhancement under non-stationary noise environments.

The aim of the study was to examine the relationship between tinnitus pitch and maximum hearing loss, frequency range of hearing loss, and the edge frequency of the audiogram, as well as, to analyze tinnitus loudness at tinnitus frequency and normal hearing frequency.

The study included 212 patients, aged between 21 to 75 years (mean age of 54.4 ± 13.5 years) with chronic subjective tinnitus and sensorineural hearing loss. For the statistical data analysis we used Chisquare test and Fisher’s exact test with level of significance p < 0:05.

Tinnitus pitch corresponding to the frequency range of hearing loss, maximum hearing loss and the edge frequency was found in 70.8%, 37.3%, and 16.5% of the patients, respectively. The majority of patients had tinnitus pitch from 3000 to 8000 Hz corresponding to the range of hearing loss (p < 0:001). The mean tinnitus pitch was 3545 Hz ± 2482. The majority (66%) of patients had tinnitus loudness 4–7 dB SL. The mean sensation level at tinnitus frequency was 4.9 dB SL ± 1.9, and 13 dB SL ± 2.9 at normal hearing frequency.

Tinnitus pitch corresponded to the frequency range of hearing loss in majority of patients. There was no relationship between tinnitus pitch and the edge frequency of the audiogram. Loudness matching outside the tinnitus frequency showed higher sensation level than loudness matching at tinnitus frequency.

The effects of friction were observed in electric guitar strings passing over an electric guitar saddle. The effects of changing the ratio of the diameter of the winding to the diameter of the core of the string, the angle through which the string is bent, and the length on either side of the saddle were measured. Relative tensions were deduced by plucking and measuring the frequencies of vibration of the two portions of string. Coefficients of friction consistent with the capstan equation were calculated and were found to be lower than 0.26 for wound strings (nickel plated steel windings on steel cores) and lower than 0.17 for unwound (tin plated steel) strings. The largest values of friction were associated with strings of narrower windings and wider cores and this may be due to the uneven nature of the contact between the string and saddle for wound strings or due the surface of the windings deforming more, encouraging fresh (and therefore higher friction) metal to metal contact. It is advised to apply lubrication under the saddle to string contact point after first bringing the string up to pitch rather than before in order to prevent this fresh metal to metal contact.

The paper presents the key-finding algorithm based on the music signature concept. The proposed music signature is a set of 2-D vectors which can be treated as a compressed form of representation of a musical content in the 2-D space. Each vector represents different pitch class. Its direction is determined by the position of the corresponding major key in the circle of fifths. The length of each vector reflects the multiplicity (i.e. number of occurrences) of the pitch class in a musical piece or its fragment. The paper presents the theoretical background, examples explaining the essence of the idea and the results of the conducted tests which confirm the effectiveness of the proposed algorithm for finding the key based on the analysis of the music signature. The developed method was compared with the key-finding algorithms using Krumhansl-Kessler, Temperley and Albrecht-Shanahan profiles. The experiments were performed on the set of Bach preludes, Bach fugues and Chopin preludes.

Choral singers are among intensive voice users whose excessive vocal effort puts them at risk of developing voice disorders. The aim of the work was to assess voice quality for choral singers in the choir at the Polish-Japanese Academy of Information Technology. This evaluation was carried out using the acoustic parameters from the COVAREP (A Collaborative Voice Analysis Repository For Speech Technologies) repository. A prototype of a mobile application was also prepared to allow the calculation of these parameters.

The study group comprised 6 male and 19 female choir singers. The control group consisted of healthy non-singing individuals, 50 men and 39 women. Auditory perceptual assessment (using the RBH scale) as well as acoustic analysis were used to test the voice quality of all the participants. The voice quality of the female choir singers proved to be normal in comparison with the control group.

The male choir singers were found to have tense voice in comparison with the controls. The parameters which proved most effective for voice evaluation were Peak Slope and Normalized Amplitude Quotient.

One of the most effective designs to control the road traffic noise is the T-shaped barrier. The aim of this study was to examine the performance of T-shape noise barriers covered with oblique diffusers using boundary element method. A 2D simulation technique based on the boundary element method (BEM) was used to compute the insertion loss at the center frequency of each one-third octave band. In designed barriers, the top surface of the T-shaped noise barriers was covered with oblique diffusers. The width and height of the barrier stem and the width of its cap were 0.3, 2.7, and 1 m, respectively. Angles of he oblique diffusers were 15, 30, and 45 degrees. The oblique diffusers were placed on the top surface with two designs including same oblique diffusers (SOD) and quadratic residue oblique diffusers (QROD). Barriers considered were made of concrete, an acoustically rigid material. The barrier with characteristics of QROD, forward direction, and sequence of angles (15, 30, and 45 degrees) had the greatest value of the overall A-weighted insertion loss equal to 18.3 to 21.8 dBA at a distance of 20 m with various heights of 0 to 6 m.

This article shows acoustic measurements from Svalbard archipelago in the Arctic, located in the Arctic Ocean. The aim of the research was to show the Svalbard soundscape as well as to record and analyse the spatial-temporal dynamics of the acoustic environment, the human impact on the soundscape and to collect baseline data for future comparative research. Svalbard is interesting for many science disciplines because it has an arctic climate and, at the same time, it is relatively easily accessible. Climatologists, geologists, glaciologists, biologists and even anthropologists could find interesting themes to investigate here. Additionally, the soundscape of Spitsbergen is worthy of detailed examination. This paper presents comparative analysis of the soundscape of various spots near Longyearbyen in Management Area 10. The soundscape analysis of selected valleys shows the strong influence of human activity on the soundscape as well as the variability and characteristic features of the natural Arctic soundscape.

Ultrasonic processing in the cavitation mode is used to produce the composite materials based on the metal matrix and reinforcing particles of micro- and nano-sizes. In such a case, the deagglomeration of aggregates and the uniform distribution of particles are the expected effects. Although the particles can not only fragment in the acoustic field, they also can coagulate, coarsen and precipitate. In this paper, a theoretical study of processes of deagglomeration and coagulation of particles in the liquid metal under ultrasonic treatment is made. The influence of various parameters of ultrasound and dispersion medium on the dynamics of particles in the acoustic field is considered on the basis of the proposed mathematical model. The criterion of leading process (coagulation or deagglomeration) has been proposed. The calculated results are compared with the experimental ones known from the scientific literature.

Main aim of this study is to combine the characteristics of the sonic crystal (SC) with acoustic panels and porous materials to improve the sound transmission loss (STL) through the triple-panel structure. SCs cause a bandgap centered around a certain frequency (Bragg’s frequency) due to generation of destructive interference. Initially, an analytical method is developed that extends the previous theory of double-panel structure to predict STL through a triple-panel structure. Finite element (FE) simulations are performed to obtain the STL through the triple-panel, which are validated with the analytical predictions. Various configurations are analyzed using the FE method based on the method of inserting the porous material and SCs between the panels to address the combined effect. STL through the triple-panel structure is compared with that through the double-panel structure having the same total weight and total thickness. It is found that the combined structure of the triple panel and the SC with glass wool as filler gives the best soundproof performance for the same external dimensions. For narrow air gaps, filing with glass wool is more advantageous than inserting one row of SC. In addition, the triple panel combined with a SC has better soundproofing than the two-panel counterparts.

An isogeometric boundary element method is applied to simulate wave scattering problems governed by the Helmholtz equation. The NURBS (non-uniform rational B-splines) widely used in the CAD (computer aided design) field is applied to represent the geometric model and approximate physical field variables. The Burton-Miller formulation is used to overcome the fictitious frequency problem when using a single Helmholtz boundary integral equation for exterior boundary-value problems. The singular integrals existing in Burton-Miller formulation are evaluated directly and accurately using Hadamard’s finite part integration. Fast multipole method is applied to accelerate the solution of the system of equations. It is demonstrated that the isogeometric boundary element method based on NURBS performs better than the conventional approach based on Lagrange basis functions in terms of accuracy, and the use of the fast multipole method both retains the accuracy for isogeometric boundary element method and reduces the computational cost.

The airflow in the mouth of an open and closed flue organ pipe of corresponding geometrical proportions is studied. The phase locked particle image velocimetry with subsequent analysis by the biorthogonal decomposition is employed in order to compare the flow mechanisms and related features. The most significant differences lie in the mean velocity distribution and rapidity of the jet lateral motion. Remarks on the pressure estimation from PIV data and its importance for the aeroacoustic source terms are made and a specific example is discussed.

The condition monitoring techniques like acoustic emission, vibration analysis, and infrared thermography, used for the failure diagnosis of bearings, require longer processing time, as they have to perform acoustical measurement followed by signal processing and further analysis using special software. However, for any bearing, its period of usage can be easily determined within an hour, by measuring the bearing sound, using sound level meter (SLM). In this paper the acoustical analysis of the spindle bearing of a radial drilling machine was performed using SLM, by measuring the sound pressure level of the bearing in decibels, for different frequencies, while muting all the other noises. Then using an experimental set up, two database readings were taken, one for new bearing and the other for completely damaged bearing, both are SKF6207, which itself is the spindle bearing. From these three sets of sound pressure level readings, the period of usage of the spindle bearing, was calculated using an interpolation equation, by substituting the life of the bearing from the manufacturer’s catalogue. Therefore, for any machine with a SKF6207 bearing, its usage time can be estimated using the database readings and one measurement on that machine, all with the same speed.

In the paper, the results of investigations on the properties of acoustic emission signals generated in a tested pressure vessel are presented. The investigations were performed by repeating several times the following procedure: an increase in pressure, maintaining a given pressure level, a further increase in pressure, and then maintaining the pressure at new determined level. During the tests the acoustic emission signals were recorded by the measuring system 8AE-PD with piezoelectric sensors D9241A. The used eight-channel measuring system 8AE-PD enables the monitoring, recording and then basic and advanced analysis of signals.

The results of basic analysis carried out in domain of time and the results of advanced analysis carried out in the discrimination threshold domain of the recorded acoustic emission signals are presented in the paper.

In the framework of the advanced analysis, results are described by the defined by the author descriptors with acronyms ADC, ADP and ADNC. Such description is based on identifying the properties of amplitude distributions of acoustic emission signals by assigning them the level of advancement. It is shown that for signals including continoues AE or single burst AE signals descriptions of such registered signals by means of ADC, ADP and ADNC descriptors and by Upp and Urms descriptors provide identical ordering of registered acoustic emission signals. For complex signals, the description using ADC, ADP and ADNC descriptors based on the analysis of amplitude distributions of recorded signals gives the order of signals with more accurate connection with deformational processes being sources of acoustic emission signals.

The paper investigates the interdependence between the perceptual identification of the vocalic quality of six isolated Polish vowels traditionally defined by the spectral envelope and the fundamental frequency F0. The stimuli used in the listening experiments were natural female and male voices, which were modified by changing the F0 values in the ±1 octave range. The results were then compared with the outcome of the experiments on fully synthetic voices. Despite the differences in the generation of the investigated stimuli and their technical quality, consistent results were obtained. They confirmed the findings that in the perceptual identification of vowels of key importance is not only the position of the formants on the F1 × F2 plane but also their relationship to F0, the connection between the formants and the harmonics and other factors. The paper presents, in quantitative terms, all possible kinds of perceptual shifts of Polish vowels from one phonetic category to another in the function of voice pitch. An additional perceptual experiment was also conducted to check a broader range of F0 changes and their impact on the identification of vowels in CVC (consonant, vowel, consonant) structures. A mismatch between the formants and the glottal tone value can lead to a change in phonetic category.

Speech enhancement is fundamental for various real time speech applications and it is a challenging task in the case of a single channel because practically only one data channel is available. We have proposed a supervised single channel speech enhancement algorithm in this paper based on a deep neural network (DNN) and less aggressive Wiener filtering as additional DNN layer. During the training stage the network learns and predicts the magnitude spectrums of the clean and noise signals from input noisy speech acoustic features. Relative spectral transform-perceptual linear prediction (RASTA-PLP) is used in the proposed method to extract the acoustic features at the frame level. Autoregressive moving average (ARMA) filter is applied to smooth the temporal curves of extracted features. The trained network predicts the coefficients to construct a ratio mask based on mean square error (MSE) objective cost function. The less aggressive Wiener filter is placed as an additional layer on the top of a DNN to produce an enhanced magnitude spectrum. Finally, the noisy speech phase is used to reconstruct the enhanced speech. The experimental results demonstrate that the proposed DNN framework with less aggressive Wiener filtering outperforms the competing speech enhancement methods in terms of the speech quality and intelligibility.

The main objective of the research presented in this paper is to enhance driver-passengers comfort of a vehicle that in turn leads to better vehicle safety and stability. The focus was put on studying the interior vibration and noise contributions originated from tire-road and engine-transmission subsystems, due to their significant impact on the dynamic performance of the vehicle. The noise and vibration measurements were recorded at the driver’s head position and on the driver legs room. Furthermore, the influence of different tire types and road surface textures on the vehicle interior noise and vibration were considered. The results indicate that the widely used conventional engine mounts and tires in commercial vehicles cannot fulfill the conflicting requirements for the best isolation concerning both road surface and engine-transmission induced excitations. The values of driver’s head position sound pressure level and floor vibration acceleration broadband averages originate for engine-transmission are lower than that for tire-road interaction. Furthermore, the values of RMS, crest factor, kurtosis and IRI for the vehicle waveform were estimated for vehicle speeds, tire types and road surface textures. Moreover, the percentage contribution for both interior noise and vibration originated from tire-road interaction is higher than the one from vehicle engine-transmission system in all the vehicle speed range, tire type and road surface texture considered.

Biography of Jozef John Zwislocki (March 22, 1922 – May 14, 2018) – Polish-born American neuroscientist. Granted fellowship into the Acoustical Society of America, as well as membership to the United States National Academy of Sciences, Polish Academy of Sciences, American Association for the Advancement of Science, and Association for Research in Otolaryngology (among others). He worked at Federal Institute of Technology in Zurich, Switzerland, taught at the University of Basel, was on a research fellowship at Harvard University, and was member of the Syracuse University faculty. Owner of twelve patents.

In this paper, the applications of the multivariate data analysis and optimization on vibration signals from compressors have been tested on the assembly line to identify nonconforming products. The multivariate analysis has wide applicability in the optimization of weather forecasting, agricultural experiments, or, as in this case study, in quality control. The techniques of discriminant analysis and linear program were used to solve the problem. The acceleration and velocity signals used in this work were measured in twenty-five rotating compressors, of which eleven were classified as good baseline compressors and fourteen with manufacturing defects by the specialists in the final acoustic test of the production line. The results obtained with the discriminant analysis separated the conforming and nonconforming groups with a significance level of 0.01, which validated the proposed methodology.

A novel method of active noise control using adaptive radiation sound sources is investigated. A finite element model of a modal enclosed sound field is excited harmonically, representing a noise field in the low-frequency range. The control sources are comprised of elementary dipole sources for which the driving signals are adjusted by an optimization method. Two set-up cases of the proposed compound sources are investigated. The coupling of the control sources with the modal sound field is discussed. The simulated performance of the proposed method is compared with that of a system with distributed simple sources and the results show the effectiveness of the sources with adaptive radiation for active noise control in small enclosures.

The increment in the number of automobiles and the densification of the city has increased noise pollution rates. In addition, the lack of regulation in Chile regarding the acoustic insulation of façades is a problem of a growing concern. The main objective of the present study was to obtain a model of the Sound Insulation of housing, façades, stratified in Santiago, Chile, based on constructive variables. It is expected to serve as a basis for one future regulation for acoustic façades of houses. In the present study, tests based on the international ISO 140-5 standard were carried out in situ. An estimation model of the Standardized Level Difference Dls,2m,nT,w + C, was obtained based on the opening/façade proportion, and the type of glass used for the windows.

Combine harvesters are the source a large amount of noise in agriculture. Depending on different working conditions, the noise of such machines can have a significant effect on the hearing condition of drivers. Therefore, it is highly important to study the noise signals caused by these machines and find solutions for reducing the produced noise. The present study was carried out is order to obtain the fractal dimension (FD) of the noise signals in Sampo and John Deere combine harvesters in different operational conditions. The noise signals of the combines were recorded with different engine speeds, operational conditions, gear states, and locations. Four methods of direct estimations of the FD of the waveform in the time domain with three sliding windows with lengths of 50, 100, and 200 ms were employed. The results showed that the Fractal Dimension/Sound Pressure Level [dB] in John Deere and Sampo combines varied in the ranges of 1.44/96.8 to 1.57/103.2 and 1.23/92.3 to 1.51/104.1, respectively. The cabins of Sampo and John Deere combines reduced and enhanced these amounts, respectively. With an increase in the length of the sliding windows and the engine speed of the combines, the amount of FD increased. In other words, the size of the suitable window depends on the extraction method of calculating the FD. The results also showed that the type of the gearbox used in the combines could have a tangible effect on the trend of changes in the FD.