Microphone Positions Research Articles

Ergonomic Workstation Design for Online Shop Live Streamers: Reducing Musculoskeletal Disorders (Pt. Xyz Case Study)

This study was conducted at PT. XYZ to evaluate and reduce the risk of musculoskeletal disorders among host streamers through the design of ergonomic workstations. The research method used was a quantitative case study approach, with the population consisting of employees at PT. XYZ who worked as host streamers. Purposive sampling was employed to select 10 respondents who frequently experienced fatigue and complaints in specific body areas. Primary data were collected through observation, questionnaires, guidelines, and interviews. Posture measurements were taken using the REBA (Rapid Entire Body Assessment) and RULA (Rapid Upper Limb Assessment) methods. The results indicated that prior to the intervention, host streamers had REBA and RULA scores that signified a high risk of musculoskeletal disorders, particularly in the neck, shoulders, back, and legs. After implementing an adjustable live streaming workstation design, along with adjustments to lighting and microphone positions, there was a significant decrease in the REBA and RULA scores. The REBA score decreased from 10 to 1, and the average RULA score dropped from 7 to 3. This indicated a significant reduction in the risk of musculoskeletal disorders. The conclusion of the study was that the design of an adjustable ergonomic workstation significantly reduced the risk of musculoskeletal disorders among host streamers at PT. XYZ, thereby enhancing worker comfort and health during live streaming.

Improving speech intelligibility in noise and spatial perception: the critical role of hearing aid microphone position.

Hearing aid (HA) manufacturers have introduced behind-the-ear (BTE) models where the microphone is positioned in the ear canal, which could impact auditory performance by distorting the pinna's acoustic cues. This study aimed to compare two different BTE HAs with varying microphone positions: the receiver in the ear (RITE) and the transducer in the ear (TIE). The study involved 10 participants who had never used HAs before. They used both RITE and TIE HAs bilaterally for 3 weeks. Auditory performance was assessed through free field hearing assessments (hearing thresholds, speech recognition threshold, and speech discrimination score), the Turkish Matrix Sentence Test (TURMatrix), a sound localization test, and the Satisfaction with Amplification in Daily Living (SADL) questionnaire. There was no significant difference between TIE and RITE in the free field hearing assessments. However, TIE outperformed RITE in non-adaptive TURMatrix scores in quiet, adaptive TURMatrix scores in noise, and sound localization accuracy at various angles. SADL sub-scores (Positive Effect, Service and Cost, and Personal Image) and overall satisfaction scores were significantly better for TIE. The microphone position in HAs can influence auditory performance. This study demonstrated that TIE provided better speech intelligibility, localization accuracy, and user satisfaction compared to RITE.

Evaluation of the effect of respiratory airflow on captured speech quality: application to pop filters

Recording speech is often a matter of proper experimental design to obtain the best signal quality. This notably includes appropriate microphone positioning and gain adjustment. However, some radio communication devices require that the microphone be placed in a very close proximity to the mouth, therefore capturing the respiratory airflow, as well as the air bursts from plosive consonants. In this paper, we assess an objective evaluation method for the quality of speech signals captured in such adverse conditions. It employs an artificial mouth coupled with a steady airflow to compute the Speech Transmission Index (STI). We apply this method to the evaluation of four commercially available pop filters. The addition of a 30~L/min airflow reduces the STI with the microphone alone from excellent to fair. With soft fabric filters, we gain back the original STI, while it remains very slightly degraded with a metal mesh filter. The objective metric is finally confronted with a listening test. The perceived quality of simulated samples with the added airflow is similar to the STI results. However, no significant difference between the filters is found with real speech samples containing plosive consonants and no respiration noise.

Sound Source Localization at the Rolling Tire

A vehicle's tire rolling on pavement causes complex noise generating mechanisms which are a dominant sub-source of vehicle traffic noise. For the quantification of noise levels due to these tire-pavement interactions, standardized acoustic measurement methods exist, such as the Close-Proximity (CPX) method. These methods require microphones mounted near a rolling tire. However, they do not provide insight in the distribution and amplitudes of present sound sources. In this contribution, we aim at identifying the dominant sound sources of the rolling tire at discrete frequencies. This is accomplished by an inverse method combining microphone array measurements and Finite Element (FE) simulations. The microphone measurements were recorded on Austrian highways with a large measurement trailer built for research purposes. With the considered inverse method, not only the amplitudes of the dominant sound sources but also their phases can be identified. With these identified sources and a validated FE model of the measurement trailer, the sound pressure field within the trailer may be computed. The thereby calculated sound pressure is compared to measurements at the CPX microphone positions and other locations within the trailer.

Effects of impact and microphone positions on heavy-weight floor impact sound pressure levels in concrete buildings

This study aims to analyze the spatial distribution of heavy-weight impact sound transmitted to a receiving room in 59 households within a concrete residential building. The performance of buildings is generally regulated by single number quantities such as L′iA,Fmax. To enhance performance, it is important to understand the effect of the impact sound pressure level at designated impact positions and microphones. Therefore, this study proposes a method to quantify the impac sound pressure level for determining L′iA,Fmax according to the impact and microphone positions. Analysis of the data revealed that significant similarities in frequency spectra were observed when generating impacts at two positions near the window, in addition to two at the area near the corridor, both of which have similar boundary conditions. The central impact position exhibited the highest contribution at 24.6%, while other positions had contributions of 18.2–19.5%. For the microphone positions, those located at the corners on either side of the window showed contributions of 22.8–23.7%, whereas the remaining positions demonstrated lower contributions of 17.3–18.5%. These results can serve as basic data for developing construction methods to reduce floor impact sound.

Soundbox-based sound insulation measurement of composite panels with viscoelastic damping

Sound transmission loss (STL), an essential index for assessing the sound insulation performance of composite laminated structures, typically relies on experimental methods to measure. The soundbox method (SBM), a straightforward technique for measuring the STL, is sensitive to microphones’ positions. Within the framework of the Chebyshev-Ritz method, a semi-analytical vibro-acoustic model extended to composite laminated panels with viscoelastic damping (VED) is proposed for the first time. Based on the developed simplified layer-wise theory, the panel is modeled using three layers: the top face layer, the VED layer, and the bottom face layer. A closed cavity is added to the model as the soundbox enclosure used in actual measurements. By employing the Hamilton's principle, the governing equation for the coupling system is derived, and the vibration and internal acoustic responses of the coupling system are calculated. A discretization strategy is introduced to address the frequency-dependent properties of the VED layer, avoiding the need to reconstruct the stiffness matrix at each frequency. To obtain the STL of the panel, sound pressures at external measurement points are calculated based on the Rayleigh integral. The proposed model is validated against numerical results from finite element analyses. The influences of the microphone position inside and outside the cavity on the measured STL are studied. Furthermore, parametric studies over the microphones' positions are performed to enhance the SBM-based evaluation of the composite panel's sound insulation performance. The optimal locations for two internal microphones and one external microphone are recommended. Finally, experimental studies are carried out to guide the implementation of the SBM.

Doppler frequency analysis for sound-field sampling with moving microphones

Moving microphones allow for the fast acquisition of sound-field data that encode acoustic impulse responses in time-invariant environments. Corresponding decoding algorithms use the knowledge of instantaneous microphone positions for relating the dynamic samples to a positional context and solving the involved spatio-temporal channel estimation problem subject to the particular parameterization model. Usually, the resulting parameter estimates are supposed to remain widely unaffected by the Doppler effect despite the continuously moving sensor. However, this assumption raises issues from the physical point of view. So far, mathematical investigations into the actual meaning of the Doppler effect for such dynamic sampling procedures have been barely provided. Therefore, in this paper, we propose a new generic concept for the dynamic sampling model, introducing a channel representation that is explicitly based on the instantaneous Doppler shift according to the microphone trajectory. Within this model, it can be clearly seen that exact trajectory tracking implies exact Doppler-shift rendering and, thus, enables unbiased parameter recovery. Further, we investigate the impact of non-perfect trajectory data and the resulting Doppler-shift mismatches. Also, we derive a general analysis scheme that decomposes the microphone signal along with the encoded parameters into particular subbands of Doppler-shifted frequency components. Finally, for periodic excitation, we exactly characterize the Doppler-shift influences in the sampled signal by convolution operations in the frequency domain with trajectory-dependent filters.

Bayesian estimation of dissipation and sound speed in tube measurements using a transfer-function model.

This study discusses acoustic dissipation, which contributes to inaccuracies in impedance tube measurements. To improve the accuracy of these measurements, this paper introduces a transfer function model that integrates diverse dissipation prediction models. Bayesian inference is used to estimate the important parameters included in these models, describing dissipation originating from various mechanisms, sound speed, and microphone positions. By using experimental measurements and considering a hypothetical air layer in front of a rigid termination as the material under test, Bayesian parameter estimation allows a substantial enhancement in characterization accuracy by incorporating the dissipation and sound speed estimates. This approach effectively minimizes residual absorption coefficients attributed to both boundary-layer effects and air medium relaxation. The incorporation of dissipation models leads to a substantial reduction (to 1%) in residual absorption coefficients. Moreover, the use of accurately estimated parameters further enhances the accuracy of actual tube measurements of materials using the two-microphone transfer function method.

Assessing the influence of street canyon shape on aircraft noise: Results from measurements in courtyards near Amsterdam Schiphol Airport

Aircraft noise is a major stressor for communities in the vicinity of airports. Aircraft noise prediction models omit the built environment, based on an implicit assumption that the impact of buildings on the propagation of aircraft noise is neglectable. In this article a study is presented in which aircraft noise levels were measured near walls facing towards and away from aircraft flyovers in an urban test environment near Amsterdam Schiphol Airport. The test environment comprises three adjacent courtyards, each enclosed by stacked shipping containers. To examine the influence of street geometry on aircraft noise, specifically for slanted roofs and building insets, the shipping containers were stacked in a different pattern around each courtyard. In total, sound levels for 2383 aircraft flyovers were analysed across five months at ten microphone positions within the courtyards, for both arrivals and departures. Depending on the geometry of the courtyards, mean differences (LA,max) between facades with- and without a line of sight towards the aircraft ranged between −1,3dBA and 5,0dBA for arrivals, and 8,7dBA and 13,6dBA for departures. SEL values ranged between between −0,8dBA and 4,3dBA for arrivals, and 8,1dBA and 11,6dBA for departures. The results suggest that slanted roofs perpendicular to the flight direction deflect incident sound, substantially reducing the sounds levels inside courtyards. Contrarily, building insets at building sides facing away from the flight paths did not reduce sound levels underneath the overhangs significantly. The findings stress the importance of architectural and urban design to mitigate aircraft noise.

Application of adaptive finite focus beamforming method for localization of low-frequency transformer sound

In this paper, the developed adaptive linearly constrained minimum variance finite focus beamforming method is used to increase the accuracy and resolution of low-frequency transformer sound source localization. In the first step, simulations on virtual arrays are conducted to define the array size and compare the influence of three different beamforming algorithms on sound localization accuracy and resolution. In the second step, measurements of the regulating transformer as a noise source are conducted in the semi-anechoic chamber. A large rectangular microphone array, with 90 microphone positions spaced 0.44 meters apart, is used for that purpose. The developed algorithm is compared with infinite and finite focus beamforming for the localization of total noise and noise at specific noise harmonics.

Experimental Prediction Method of Free-Field Sound Emissions Using the Boundary Element Method and Laser Scanning Vibrometry

Acoustic emissions play a major role in the usability of many product categories. Therefore, mitigating the emitted sound directly at the source is paramount to improve usability and customer satisfaction. To reliably predict acoustic emissions, numerical methods such as the boundary element method (BEM) are employed, which allow for predicting, e.g., the acoustic emission into the free field. BEM algorithms need appropriate boundary conditions to couple the sound field with the structural motion of the vibrating body. In this contribution, firstly, an interpolation scheme is presented, which allows for appropriate interpolation of arbitrary velocity data to the computational grid of the BEM. Secondly, the free-field Helmholtz problem is solved with the open-source BEM software framework NiHu. The forward coupling between the device of interest and BEM is based on the surface normal velocities (i.e., a Neumann boundary condition). The BEM simulation results are validated using a previously established aeroacoustic benchmark problem. Furthermore, an application to a medical device (knee prosthesis frame) is presented. Furthermore, the radiated sound power is evaluated and contextualized with other low-cost approximations. Regarding the validation example, very good agreements are achieved between the measurements and BEM results, with a mean effective pressure level error of 0.63 dB averaged across three microphone positions. Applying the workflow to a knee prosthesis frame, the simulation is capable of predicting the acoustic radiation to four microphone positions with a mean effective pressure level error of 1.52 dB.

Externalization of Speech When Listening With Hearing Aids.

Subjective reports indicate that hearing aids can disrupt sound externalization and/or reduce the perceived distance of sounds. Here we conducted an experiment to explore this phenomenon and to quantify how frequently it occurs for different hearing-aid styles. Of particular interest were the effects of microphone position (behind the ear vs. in the ear) and dome type (closed vs. open). Participants were young adults with normal hearing or with bilateral hearing loss, who were fitted with hearing aids that allowed variations in the microphone position and the dome type. They were seated in a large sound-treated booth and presented with monosyllabic words from loudspeakers at a distance of 1.5 m. Their task was to rate the perceived externalization of each word using a rating scale that ranged from 10 (at the loudspeaker in front) to 0 (in the head) to -10 (behind the listener). On average, compared to unaided listening, hearing aids tended to reduce perceived distance and lead to more in-the-head responses. This was especially true for closed domes in combination with behind-the-ear microphones. The behavioral data along with acoustical recordings made in the ear canals of a manikin suggest that increased low-frequency ear-canal levels (with closed domes) and ambiguous spatial cues (with behind-the-ear microphones) may both contribute to breakdowns of externalization.

Data-driven estimation of sound absorption coefficients considering the positions of microphones

The sound absorption coefficient of walls is important to simulate the sound reflections in a room. However, measuring the sound absorption coefficients in detail requires many measurement points along the wall surface. Recently, machine-learning-based estimation methods have been proposed for the sound absorption coefficients using the room impulse responses as training data. In this study, to obtain the sound absorption coefficients of each wall surface with free microphone placement, we proposed the machine learning-based estimation with the room transfer functions and its measurement positions. The positional information of microphones is provided to the network in the mask data format. In the two-dimensional simulation experiments, we evaluate the estimation accuracy of the proposed method compared to the fixed microphone arrangement.

Impedance measurements on reflective surfaces: effect of nodes

The ISO 13472-2 standardizes the in-situ measurement of normal-incidence acoustic absorption of road pavements. The standard describes the use of the two-microphone method with an impedance tube specially configured for sealing against pavement surfaces. The method is of interest to the tire-pavement noise and vehicle pass-by noise communities. For roads constructed of dense graded asphalt, for example, pass-by noise test tracks conforming to ISO 10844, the pavement is a highly reflective surface. Under this measurement condition, the acoustic field within the tube develops nodes at one or the other microphone positions at characteristic frequencies. These nodes present as sharp peaks and valleys in the two-microphone transfer function. Such narrow bandwidth features are susceptible to measurement error, in particular, signal processing leakage as evidenced by significant drops in the coherence function at the frequencies of the nodes. In this paper we study the nature of the nodes and the effect they have on the two-microphone transfer function and coherence function. We further investigate how different formulations of the transfer function affect the overall 1/3rd octave band accuracy. The effect of the number of averages on both coherence and leakage error is assessed. The potential to solve this leakage effect by switching microphones to other positions is evaluated in relation to a procedure with two fixed microphones. It was concluded that although shifting microphone positions does have advantages in terms of preventing leakage, the mandatory application of a reference measurement reduces the overall error. Finally, recommendations for improving the procedure in ISO 13472-2 are formulated.

Improving the ISO 11819 standards for better characterization of noise reduction of porous pavements

The SPB method (ISO 11819-1) is designed to measure the acoustic properties in semi-free field of road pavements at a specific spot along a road, which in practice may characterize a 15-30 m length of the road lane. The CPX method (ISO 11819-2) is designed to correlate well to the SPB method in characterizing road pavements, while extending the measured road section to an arbitrary distance. It is known that both methods may underestimate the noise reduction of porous pavements compared to dense pavements. A remedy for the SPB method may be to extend it to measure equivalent sound exposure levels (SEL) instead. This will be discussed in the paper. Since the microphone locationsin the CPX method have been selected to give the best correlation between noise reductions measured by the CPX and the SPB methods, the inherent problem of the SPB method exists also in the CPX method. As a remedy for the underestimation of noise reduction when using the CPX method, some users have applied other microphone positions around the reference tires than the two mandatory ones. This paper will analyze this problem and suggest how one can avoid it; essentially by adding a microphone behind the tire.

Designing of two-dimensional acoustic beamforming array using machine learning

Acoustic beamforming arrays are used for locating sound sources and their performance is measured by their directivity pattern. Designing an acoustic beamforming array to meet specific performance requirements can be challenging and requires numerical optimization of various factors such as array width, microphone positions, and the number of microphones. This study employed a Deep Neural Network (DNN) approach to predict the arrangement of a two-dimensional acoustic beamforming array giving the desired beam pattern. The DNN was trained on 90,000 microphone array designs and beam patterns of 1-kHz sound. The prediction accuracy was evaluated based on the mean absolute errors (MAE) of beam patterns, frequency-averaged main lobe width (MLW), and maximum sidelobe level (MSL). The results for commonly designed arrays datasets such as Brüel & Kjær style array had a 26.54% error in beam patterns, 2.34 degree/m error in MLW, and 4.21 dB error in MSL. However, for predicted arrays generated by the random array designs, the results were more accurate, with deviations of 22.08% in beam patterns, 1.82 degree/m error in MLW, and 5.07 dB in MSL.

Measuring directivity of half-inch measurement microphones

The directivity of a microphone is the relative sensitivity variation as a function of incidence angle. This is an important characteristic when every sound arriving at the microphone position is relevant. This paper presents the measuring directivity of half-inch measurement microphones. The microphone is placed in a free sound field and exposed to a sound pressure. The unloaded output voltage is measured. Then, the microphone is rotated in 30o steps, while both the acoustic centers of the microphone and the sound source remain in fixed positions in the sound field, and is exposed to the same sound pressure. The unloaded output voltage for each angle of incidence is measured and them the difference relative to 0 deg. of incidence is calculated as a function of frequency. The directivity of eight measurement microphones from the same manufacturer but of different types and frequency response characteristics (i.e. pressure field, random incidence or free-field) were measured. Results showed that the microphones object of this investigation have similar directivities in the frequency range from 16 Hz to 12.5 kHz regardless of their frequency response characteristics.

A76Quality assurance of high-fidelity simulation courses at University Hospitals Birmingham (UHB)

The Clinical Education department at UHB has been running high fidelity simulation courses for undergraduates and postgraduates across three hospital sites for several years. Feedback from faculty and candidates has led to enhancement of all courses, yet a deliberate method of Quality Assurance (QA) was deemed requisite for educational governance, standardization and to attain accreditation with ASPiH. Drawing on ASPiH Standards [1], the departmental leads for simulation designed a framework with seven domains ( The 7 domains used for quality assurance of simulation courses at UHB Five undergraduate and four postgraduate courses were evaluated. Seven courses met or exceeded departmental standards in all seven domains, whilst the other two courses met or exceeded standards in six of the seven domains. All courses had robust purposes, organization and resources at their disposal, which exceeded departmental expectations with their consistent focus on how human factors affect patient safety, and sound adherence to the relevant curricula and learning objectives. Scenario designs also incorporated a range of human and technological resources and moulage. Despite an array of technology at the Trust’s disposal, its functioning did not always meet the departmental standards, with recurring audio difficulties and occasional software tethering issues. In response, microphone positions were changed, and the department’s Digital Innovation team consulted specialists to facilitate further improvements. In the courses observed, debriefs were conducted in a holistic manner and candidate evaluation was always discussed. However, many courses lacked enough faculty members resulting in Objective Structured Assessments of Debriefing (OSADs) being performed infrequently. Swift changes were made, with a tightening of procedural behaviours within the courses, and a decision was taken for postgraduate courses to run with a minimum of four faculty members. This comprehensive QA process demonstrated that high-fidelity simulation courses at UHB are predominantly being delivered at a good standard. Some recommended changes have already been implemented for aspects found to be substandard, with implementation of additional changes planned at annual course reviews. Annual QA must then take place to guarantee maintenance of high standards and to provide a solid foundation for appraising novel simulation courses throughout the Trust. Authors confirm that all relevant ethical standards for research conduct and dissemination have been met. The submitting author confirms that relevant ethical approval was granted, if applicable.

A Feasibility Study for a Hand-Held Acoustic Imaging Camera

Acoustic imaging systems construct spatial maps of sound sources and have potential in various applications, but large, cumbersome form factors limit their adoption. This paper investigates methodologies to miniaturize acoustic camera systems for improved mobility. Our approach optimizes planar microphone array design to achieve directional sensing capabilities on significantly reduced footprints compared to benchmarks. The current prototype utilizes a 128−microphone, 50 × 50 cm2 array with beamforming algorithms to visualize acoustic fields in real time but its stationary bulk hampers portability. We propose minimizing the physical aperture by carefully selecting microphone positions and quantities with tailored spatial filter synthesis. This irregular array geometry concentrates sensitivity toward target directions while avoiding aliasing artefacts. Simulations demonstrate a 32−element, ≈20 × 20 cm2 array optimized this way can outperform the previous array in directivity and noise suppression in a sub-range of frequencies below 4 kHz, supporting a 4× surface factor reduction with acceptable trade-offs. Ongoing work involves building and testing miniature arrays to validate performance predictions and address hardware challenges. The improved mobility of compact acoustic cameras could expand applications in car monitoring, urban noise mapping and other industrial fields limited by current large systems.

Comparison of facade noise reduction measurement methods: Field results

Many projects require acoustical measurements of facade noise reduction be conducted to verify that Building Code, “green” rating systems (e.g., LEED), or other project interior noise criteria are met. In addition, federally funded airport sound insulation programs in the United States require facade noise reduction measurements be conducted to verify the eligibility of residences receiving sound insulation treatments. Currently, there exists a variety of standards to measure facade noise reduction (e.g., ASTM E966, ISO 16283-3, SAE ARP 6973). These standards employ various measurement methods: artificial exterior noise source, calibrated exterior noise source, line source, etc. In addition, these standards provide guidance on the location of exterior microphones (flush versus near facade) and correction factors to be applied to the data. Recently, a field comparison of three measurement methods was made at multiple residences: artificial exterior noise source, artificial interior noise source, and flush versus near microphone position. This is follow-on research from the National Academies of Science Airport Cooperative Research Project 02-51 and its purpose is to determine whether valid facade noise reduction measurements can be made with the loudspeaker located at the interior of the residence. A comparison of results across the measurement methods will be presented.