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Abstract

The Chinese word identification and sentence intelligibility are evaluated by grades 3 and 5 students in the classrooms with different reverberation times (RTs) from three primary school under different signal-to-noise ratios (SNRs). The relationships between subjective word identification and sentence in- telligibility scores and speech transmission index (STI) are analyzed. The results show that both Chinese word identification and sentence intelligibility scores for grades 3 and 5 students in the classroom in- creased with the increase of SNR (and STI), increased with the increase of the age of students, and decreased with the increase of RT. To achieve a 99% sentence intelligibility score, the STIs required for grades 3, grade 5 students, and adults are 0.71, 0.61, and 0.51, respectively. The required objective acoustical index determined by a certain threshold of the word identification test might be underestimated for younger children (grade 3 students) in classroom but overestimated for adults. A method based on the sentence test is more useful for speech intelligibility evaluation in classrooms than that based on the word test for different age groups. Younger children need more favorable classroom acoustical environment with a higher STI than older children and adults to achieve the optimum speech communication in the classroom.
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Abstract

Annoyance ratings for artificially created noises, resembling the main characteristics of temporal wind turbine noise, were studied by means of a listening experiment involving 21 participants with normal hearing. Three types of stimuli were examined: broadband noise (−4 dB/octave), noise generated by moving cars, and narrowband noise. All stimuli had the sound level fluctuations typical for wind turbine noise. The magnitude of the sound level fluctuations was measured in a quantitative way, by using the characteristics of amplitude modulated sound: modulation rate and modulation depth. Our aim was to examine how the modulation rate and the modulation depth influence the noise annoyance assessment of broadband and narrowband amplitude modulated noises. Three different modulation rates, 1, 2 and 4 Hz, and sound level fluctuations (a measure of the modulation depth), 3, 6, 9 dB, were applied to each type of stimuli (with exception of noise generated by the moving cars) and investigated. The participants in the listening experiment were presented with sound stimuli in laboratory conditions and asked to rate their annoyance on a numerical scale. The results have shown a significant difference between the investigated conditions. The effect was particularly strong between the annoyance judgments of different types of noise (narrow and broadband), and modulated versus unmodulated noises. Temporal fluctuations occurring in wind turbine noise are very pertinent to the perception of annoyance and could be responsible for its being a relatively annoying noise source. The obtained results were discussed and compared to the typical modulation rates and level changes that occur in recordings of real wind turbine noise.
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Abstract

The aim of this work was to measure subjective speech intelligibility in an enclosure with a long reverberation time and comparison of these results with objective parameters. Impulse Responses (IRs) were first determined with a dummy head in different measurement points of the enclosure. The following objective parameters were calculated with Dirac 4.1 software: Reverberation Time (RT), Early Decay Time (EDT), weighted Clarity (C50) and Speech Transmission Index (STI). For the chosen measurement points, a convolution of the IRs with the Polish Sentence Test (PST) and logatome tests was made. PST was presented at a background of a babble noise and speech reception threshold - SRT (i.e. SNR yielding 50% speech intelligibility) for those points were evaluated. A relationship of the sentence and logatome recognition vs. STI was determined. It was found that the final SRT data are well correlated with speech transmission index (STI), and can be expressed by a psychometric function. The difference between SRT determined in condition without reverberation and in reverberation conditions appeared to be a good measure of the effect of reverberation on speech intelligibility in a room. In addition, speech intelligibility, with and without use of the sound amplification system installed in the enclosure, was compared.
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Abstract

An acoustic emission method (AE) is widespread and often applied for partial discharge (PD) diagnostics, mainly due to its ease of application as well as noninvasiveness and relatively high sensitivity. This paper presents comparative analysis of AE signals measurement results archived under laboratory conditions as well as on-site actual AE signals generated by inside PDs in electrical power transformer during its normal service. Three different PD model sources are applied for laboratory research: point to point, multipoint to plate and surface type. A typical measuring set up commonly used for on-site transformer PD diagnostics is provided for the laboratory tasks: piezoelectric joint transducer, preamplifier, amplifier and measuring PC interface. During the on-site research there are three measuring tracks applied simultaneously. Time domain, time-frequency domain and statistical tools are used for registered AE signals analysis. A number of descriptors are proposed as a result of the analysis. In the paper, at- tempt of AE signals descriptors, archived under laboratory condition application possibilities for on-site PD diagnostics of power transformers during normal service is made.
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Abstract

Transmission of vibroacoustic energy from an internal combustion engine (ICE) to its surroundings largely depends on how it is mounted, on available transmission paths and on the construction of the vehicle body and/or its surrounding structures. This is especially true in low speed engines in enclosed areas which generate perceptually weak noise, but strong low-frequency waves which energy has a negative impact on human health, comfort and driving safety especially in prolonged exposure to the source. The primary aim of the article was to analyse components of the ICE unit which had a determining impact on the reduction of low-frequency waves. Thus, the structurally transmitted noise from the ICE to its surrounding structure (body of the passenger vehicle) was analysed. The results of the vibroacoustic measurements were compared to modal analysis in order to determine possible resonance sources in the vehicle body and/or for assessing the influence of the vehicles safety gear on the generated vibroacoustic energy transfer into the cabin area of the passenger vehicle. Measurements were made for a passenger vehicle at rest and operating in its most common operational speed as well as for the stationary ICE of a cogenerate unit (CGU). Measurements and FFT analysis were used for the detection of the vibroacoustic energy sound pressure level (noise) and mechanical vibration. Firstly, the low-frequency noise sources were determined and their direct effects on the human body were investigated. Finally, this paper suggests some measures which may contribute to the reduction of undesirable vibroacoustic energy in enclosed areas.
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Abstract

This paper describes the development phases of a numerical-experimental integrated approach aimed at obtaining sufficiently accurate predictions of the noise field emitted by an external gear pump by means of some vibration measurements on its external casing. Harmonic response methods and vibroacoustic analyses were considered as the main tools of this methodology. FFT acceleration spectra were experimentally acquired only in some positions of a 8.5 cc/rev external gear pump casing for some working conditions and considered as external excitation boundary conditions for a FE quite simplified vibroacoustic model. The emitted noise field was computed considering the pump as a ‘black box’, without taking into account the complex dynamics of the gear tooth meshing process and the consequent fluid pressure and load distribution. Sound power tests, based on sound intensity measurements, as well as sound pressure measurements in some positions around the pump casing were performed for validation purposes. The comparisons between numerical and experimental results confirmed the potentiality of this approach in offering a good compromise between noise prediction accuracy and reduction of experimental and modelling requirements.
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Abstract

Urethane foam mattresses are commonly used as cushioning when placing panel flooring on the floor slab of a building. Urethane foam consists of elastic fibres with pores. Both elements can affect the performance of the insulation against impact sounds. However, these effects have not yet been detailed, and they may change if the material properties or constitution of the fibres and pores in the cushioning change. In this paper, we propose an analytical model for use in evaluating the performance of insulation against floor impact sound. This model was used to examine the contribution of the pores versus the elastic fibres to wave transmissions from the flooring surface to the slab. The results reveal that the constitution of the foam (either open or closed cells of pores) and the thickness and hardness of the cushion layer strongly affect the sound insulation performance of the floor.
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Abstract

The same speech sounds (phones) produced by different speakers can sometimes exhibit significant differences. Therefore, it is essential to use algorithms compensating these differences in ASR systems. Speaker clustering is an attractive solution to the compensation problem, as it does not require long utterances or high computational effort at the recognition stage. The report proposes a clustering method based solely on adaptation of UBM model weights. This solution has turned out to be effective even when using a very short utterance. The obtained improvement of frame recognition quality measured by means of frame error rate is over 5%. It is noteworthy that this improvement concerns all vowels, even though the clustering discussed in this report was based only on the phoneme a. This indicates a strong correlation between the articulation of different vowels, which is probably related to the size of the vocal tract.
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Abstract

In the paper, a noise map service designated for the user interested in environmental noise is presented. Noise prediction algorithm and source model, developed for creating acoustic maps, are working in the cloud computing environment. In the study, issues related to the noise modelling of sound propagation in urban spaces are discussed with a particular focus on traffic noise. Examples of results obtained through a web application created for that purpose are shown. In addition, these are compared to results obtained from the commercial software simulations based on two road noise prediction models. Moreover, the computing performance of the developed application is investigated and analyzed. In the paper, a flowchart simulating the operation of the noise web-based service is presented showing that the created application is easy to use even for people with little experience in computer technology.
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Abstract

Superparamagnetic iron oxide nanoparticles (SPION) synthesised chemically usually need the modification of the particle surface. Other natural sources of magnetic particles are various magnetotactic bacteria. Magnetosomes isolated from magnetotactic bacteria are organelles consisting of magnetite (Fe3O4) or greigite (Fe3S4) crystals enclosed by a biological membrane. Magnetotactic bacteria produce their magnetic particles in chains. The process of isolation of magnetosome chains from the body of bacteria consists of a series of cycles of centrifugation and magnetic decantation. Using a high-energy ultrasound it is possible to break the magnetosome chains into individual nanoparticles – magnetosomes. This study presents the effect of sonication of magnetosome suspension on their acoustic properties, that is speed and attenuation of the sound. Acoustic propagation parameters are measured using ultrasonic spectroscopy based on FFT spectral analysis of the received pulses. The speed and attenuation of ultrasonic waves in magnetosome suspensions are analysed as a function of frequency, temperature, magnetic field intensity, and the angle between the direction of the wave and the direction of the field.
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Abstract

The 16th International Symposium on Sound Engineering and Tonmeistering (ISSET) organized by the Institute of Radioelectronics and Multimedia Technology (Warsaw University of Technology), Department of Sound Engineering (Fryderyk Chopin University of Music) and the Polish Radio, under auspicious of the Polish Section of the Audio Engineering Society was held in Warsaw on October 8-10 in 2015. The main topics of the Symposium covered mostly all domains of audio engineering, i.e. musical acoustics, noise control, signal processing, room acoustics, radio and television, multimedia, sound engineering and tonmeistering, perception and quality assessment, and many others. The extra attention has been paid for the problems of loudness of audio programs in radio and TV broadcasting. Over 60 people from different branches of audio technology participated in this Symposium and shared their knowledge and experiences during the paper sessions, technical tours, workshops and special presentations. The selection of abstracts of the papers presented at the ISSET’2015 are inserted below.
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Abstract

The paper is an exploration of the optimal design parameters of a space-constrained electromagnetic vibration-based generator. An electromagnetic energy harvester is composed of a coiled polyoxymethylen circular shell, a cylindrical NdFeB magnet, and a pair of helical springs. The magnet is vertically confined between the helical springs that serve as a vibrator. The electrical power connected to the coil is actuated when the energy harvester is vibrated by an external force causing the vibrator to periodically move through the coil. The primary factors of the electrical power generated from the energy harvester include a magnet, a spring, a coil, an excited frequency, an excited amplitude, and a design space. In order to obtain maximal electrical power during the excitation period, it is necessary to set the system’s natural frequency equal to the external forcing frequency. There are ten design factors of the energy harvester including the magnet diameter (Dm), the magnet height (Hm), the system damping ratio (ζsys), the spring diameter (Ds), the diameter of the spring wire (ds), the spring length (ℓs), the pitch of the spring (ps), the spring’s number of revolutions (Ns), the coil diameter (Dc), the diameter of the coil wire (dc), and the coil’s number of revolutions (Nc). Because of the mutual effects of the above factors, searching for the appropriate design parameters within a constrained space is complicated. Concerning their geometric allocation, the above ten design parameters are reduced to four (Dm, Hm, ζsys, and Nc). In order to search for optimal electrical power, the objective function of the electrical power is maximized by adjusting the four design parameters (Dm, Hm, ζsys, and Nc) via the simulated annealing method. Consequently, the optimal design parameters of Dm, Hm, ζsys, and Nc that produce maximum electrical power for an electromagnetic energy harvester are found.
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Abstract

Transportation noise is a main source of noise pollution. It is assumed that it consists of recognizable noise events which come from moving aircrafts, trains and boats. The noise of an isolated sound event is assessed by the sound exposure level, LAE. Much legislation and many regulations and guidelines employ the A-weighted time-average sound level, LAeq,T, with the time interval T of one hour or longer. LAE measurements enable an approximation of LAeq,T. The key point is the uncertainty of this approximation. It has been shown that an increase in the number of LAE categories brings about a decrease in uncertainty. For illustrative purposes, LAE measurements of aircrafts taking off and landing were carried out.
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Abstract

The term roughness is used to describe a specific sound sensation which may occur when listening to stimuli with more than one spectral component within the same critical band. It is believed that the spectral components interact inside the cochlea, which leads to fluctuations in the neural signal and, in turn, to a sensation of roughness. This study presents a roughness model composed of two successive stages: peripheral and central. The peripheral stage models the function of the peripheral ear. The central stage predicts roughness from the temporal envelope of the signal processed by the peripheral stage. The roughness model was shown to account for the perceived roughness of various types of acoustic stimuli, including the stimuli with temporal envelopes that are not sinusoidal. It thus accounted for effects of the phase and the shape of the temporal envelope on roughness. The model performance was poor for unmodulated bandpass noise stimuli.
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Abstract

This study presents the indoor soundscape framework in detail by describing the variables and factors that form an indoor soundscape study. The main objective is to introduce a new indoor soundscaping framework and systematically explain the variables that contribute to the overall evaluation of an indoor soundscape. Hence, the dependencies of physical and psychoacoustical factors of the sound environment and the spatial factors of the built entity are statistically tested. The new indoor soundscaping framework leads to an overarching evaluation perspective of enclosed sound environments, combining objective room acoustics research and noise control engineering with architectural analysis. Therefore, it is hypothesised that case spaces with certain plan organisations, volumetric relations, and spatial referencing lead to differentiated sound pressure level (SPL) and loudness (N) values. SPL and N parametric variances of the sound environments are discussed through the statistical findings with respect to the architectural characteristics of each library case space. The results show that the relation between crowd level variances and sound environment parametric values is statistically significant. It is also found that increasing the atrium height and atrium void volume, the atrium’s presence as a common architectural element, and its interpenetrating reference and domain containment results in unwanted variances and acoustic formations, leading to high SPL and N values.
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Abstract

On behalf of Upper Silesian Division of the Polish Acoustical Society (main organizer) and Committee of Acoustics of the Polish Academy of Sciences (co-organizer) we are pleased to announce that it is nearing the end of preparations for the conduct traditionally organized Conferences, which will take place as previously in Szczyrk. At the moment, more than 50 people have confirmed participation in Conferences. They will present more than 40 lectures and reports. 45th Winter School on Wave and Quantum Acoustics constitutes a platform for sharing the results and achievements obtained in different branches of physical acoustics (e.g. molecular acoustics, quantum acoustics, acousto-optics, magnetoacoustics, photoacoustics, acoustics of solid state etc.). Moreover, researches in some selected topics related to those mentioned above (e.g. optoelectronics, relaxation processes) will also be presented during the school. Currently, during the Conference Workshop on Acoustoelectronics and Workshop on Molecular Acoustics, Relaxation and Calorimetric Methods will be held – however, there is a possibility of organizing workshops on other subjects in the future. We would like to invite scientific centers and groups to cooperate in organizing workshops on the subjects of their interests. XLIVth Winter School on Environmental Acoustics and Vibroacoustics is the forum for all environmental and vibroacoustics fields. Particularly it concerns traffic noise, vibroacoustics of machines, room acoustics, building acoustics, noise protection and similar problems.
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Abstract

Sonification is defined as presentation of information by means of non-speech audio. In assistive technologies for the blind, sonification is most often used in electronic travel aids (ETAs) - devices which aid in independent mobility through obstacle detection or help in orientation and navigation. The presented review contains an authored classification of various sonification schemes implemented in the most widely known ETAs. The review covers both those commercially available and those in various stages of research, according to the input used, level of signal processing algorithm used and sonification methods. Additionally, a sonification approach developed in the Naviton project is presented. The prototype utilizes stereovision scene reconstruction, obstacle and surface segmentation and spatial HRTF filtered audio with discrete musical sounds and was successfully tested in a pilot study with blind volunteers in a controlled environment, allowing to localize and navigate around obstacles.
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Abstract

The paper presents the possible applications of using acoustic diagnostics in inspecting the technical condition of an internal combustion engine with autoignition on the example of the Fiat drive unit with common rail system. As a result of measuring the sound pressure level for specific faults and comparing the noise generated by the motor running smoothly, the detailed maps of changes in the acoustic spectrum are possible to generate. These results may be helpful in the future diagnostics of internal combustion engines. In the paper, the results of scientific work in the area of research, design and operation of internal combustion engines, conducted at the Department of Automotive Engineering, in cooperation with the Laboratory of Hydraulic Drives & Vibroacoustics of Machines at the Wroclaw University of Technology are included.
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Abstract

In this article the authors investigated and presented the experiments on the sentence boundaries annotation from Polish speech using acoustic cues as a source of information. The main result of the investigation is an algorithm for detection of the syntactic boundaries appearing in the places of punctuation marks. In the first stage, the algorithm detects pauses and divides a speech signal into segments. In the second stage, it verifies the configuration of acoustic features and puts hypotheses of the positions of punctuation marks. Classification is performed with parameters describing phone duration and energy, speaking rate, fundamental frequency contours and frequency bands. The best results were achieved for Naive Bayes classifier. The efficiency of the algorithm is 52% precision and 98% recall. Another significant outcome of the research is statistical models of acoustic cues correlated with punctuation in spoken Polish.
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Abstract

Subspace-based methods have been effectively used to estimate enhanced speech from noisy speech samples. In the traditional subspace approaches, a critical step is splitting of two invariant subspaces associated with signal and noise via subspace decomposition, which is often performed by singular-value decomposition or eigenvalue decomposition. However, these decomposition algorithms are highly sensitive to the presence of large corruptions, resulting in a large amount of residual noise within enhanced speech in low signal-to-noise ratio (SNR) situations. In this paper, a joint low-rank and sparse matrix decomposition (JLSMD) based subspace method is proposed for speech enhancement. In the proposed method, we firstly structure the corrupted data as a Toeplitz matrix and estimate its effective rank value for the underlying clean speech matrix. Then the subspace decomposition is performed by means of JLSMD, where the decomposed low-rank part corresponds to enhanced speech and the sparse part corresponds to noise signal, respectively. An extensive set of experiments have been carried out for both of white Gaussian noise and real-world noise. Experimental results show that the proposed method performs better than conventional methods in many types of strong noise conditions, in terms of yielding less residual noise and lower speech distortion.
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Abstract

The active noise-reducing casing developed and promoted by the authors in recent publications have multiple advantages over other active noise control methods. When compared to classical solutions, it allows for obtaining global reduction of noise generated by a device enclosed in the casing. Moreover, the system does not require loudspeakers, and much smaller actuators attached to the casing walls are used instead. In turn, when compared to passive casings, the walls can be made thinner, lighter and with much better thermal transfer than sound-absorbing materials. For active noise control a feedforward structure is usually used. However, it requires an in-advance reference signal, which can be difficult to be acquired for some applications. Fortunately, usually the dominant noise components are due to rotational operations of the enclosed device parts, and thus they are tonal and multitonal. Therefore, it can be adequately predicted and the Internal Model Control structure can be used to benefit from algorithms well developed for feedforward systems. The authors have already tested that approach for a rigid casing, where interaction of the walls was significantly reduced. In this paper the idea is further explored and applied for a light-weight casing, more frequently met in practice, where each vibrating wall of the casing influences all the other walls. The system is verified in laboratory experiments.
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Abstract

The paper presents results of research on an influence of listening fatigue on the detection of changes in spectrum and envelope of musical samples. The experiment was carried out under conditions which normally exist in a studio or on the stage when sound material is recorded and/or mixed. The equivalent level of presented sound samples is usually 90 dB and this is an average value of sound level existing in control room at various recording activities. Such musical material may be treated as a noise so Temporary Threshold Shift phenomenon may occur after several sessions and this may lead to a listening fatigue effect. Fourteen subjects participated in the first part of the experiment and all of them have the normal hearing thresholds. The stimuli contained the musical material with introduced changes in sound spectrum up to ±6 dB in low (100 Hz), middle (1 kHz) and high frequency (10 kHz) octave bands. In the second part of research five subjects listened to musical samples with introduced envelope changes up to ±6 dB in interval of 1 s. The time of loud music exposure was 60, 90 and 120 minutes and this material was completely different from the tested samples. It turned out that listening to the music with an Leq = 90 dB for 1 hour influences the hearing thresholds for middle frequency region (about 1-2 kHz) and this has been reflected in a perception of spectral changes. The perceived peaks/notches of 3 dB have the detection ability at 70% and the changes of low and high ranges of spectrum were perceived at the similar level. After the longer exposure, the thresholds shifted up to 4.5 dB for the all investigated stimuli. It has been also found that hearing fatigue after 1 hour of a listening influences the perception of envelope which gets worse of 2 dB in comparison to the fresh-ear listening. When time of listening to the loud music increases, the changes in envelopes which can be detected rise to the value of 6 dB after 90-minutes exposure and it does not increase with further prolongation of listening time.
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