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Journal articles on the topic 'Microphones'
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1
Auliya, Rahmat Zaki, Muhamad Ramdzan Buyong, Burhanuddin Yeop Majlis, Mohd Farhanulhakim Mohd. Razip Wee, and Poh Choon Ooi. "Characterization of embedded membrane in corrugated silicon microphones for high-frequency resonance applications." Microelectronics International 36, no. 4 (October 7, 2019): 137–42. http://dx.doi.org/10.1108/mi-02-2019-0010.
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Purpose The purpose of this paper is to propose an alternative approach to improve the performance of microelectromechanical systems (MEMSs) silicon (Si) condenser microphones in terms of operating frequency and sensitivity through the introduction of a secondary material with a contrast of mechanical properties in the corrugated membrane. Design/methodology/approach Finite element method from COMSOL is used to analyze the MEMS microphones performance consisting of solid mechanic, electrostatic and thermoviscous acoustic interfaces. Hence, the simulated results could described the physical mechanism of the MEMS microphones, especially in the case of microphones with complex geometry. A 2-D model was used to simplify computation by applying axis symmetry condition. Findings The simulation results have suggested that the operating frequency range of the microphone could be extended to be operated beyond 20 kHz in the audible frequency range. The data showed that the frequency resonance of the microphone using a corrugated Si membrane with SiC as the embedded membrane is increased up to 70 kHz compared with 63 kHz for the plane Si membrane, whereas the microphone’s sensitivity is slightly decreased to −79 from −76 dB. Furthermore, the frequency resonance of a corrugated membrane microphone could be improved from 26 to 70 kHz by embedding the SiC material. Last, the sensitivity and frequency resonance value of the microphones could be modified by adjusting the height of the embedded material. Originality/value Based on these theoretical results, the proposed modification highlighted the advantages of simultaneous modifications of frequency and sensitivity that could extend the applications of sound and acoustic detections in the ultrasonic spectrum with an acceptable performance compared with the typical state-of-the-art Si condenser microphones.
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Mittal, Manan, Kanad Sarkar, Austin Lu, Ryan M. Corey, and Andrew C. Singer. "Source separation using bandlimited external microphones and a microphone array." Journal of the Acoustical Society of America 153, no. 3_supplement (March 1, 2023): A52. http://dx.doi.org/10.1121/10.0018131.
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Modern listening devices are equipped with air-conducted microphones and contact microphones. External microphones, like those on a listening device, have been used to estimate relative transfer functions (RTFs) at microphone arrays. With numerous active sound sources, the air-conducted microphones perform poorly while the contact microphones are robust to external noise. A drawback of contact microphones is that they are bandlimited. Past work has shown that the contact microphone and microphone array can be combined to estimate RTFs in the low frequencies. To overcome the limitations of the contact microphone, we propose a method that leverages the full-band signal at the microphone array to provide beamforming gains at higher frequencies. We demonstrate this method by separating three human talkers in a noisy environment.
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West, James E., Ian M. McLane, and Valerie Rennoll. "Sixty years of contributions to the world of microphones." Journal of the Acoustical Society of America 153, no. 3_supplement (March 1, 2023): A106. http://dx.doi.org/10.1121/10.0018320.
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For almost 60 years, electret microphones have been the preferred sensors for applications in communications, mainly because the microphones are linear over a broad frequency range and rather simple to manufacture. Because the electret microphone can be mass produced with only slight differences in phase and frequency response, multiple units can be combined to form a variety of directional arrays ranging from second-order unidirectional to two-dimensional arrays for focusing on a specific area. While electret microphones and arrays have similar utility for monitoring lung and heart sounds from the body, the body sounds captured can be easily corrupted by noise external to the body. Advanced signal processing techniques can mitigate contributions from airborne noise but are computationally intensive. By modifying the acoustic impedance of the electret microphone’s diaphragm to match that of the body, we are able to capture high-fidelity heart and lung sounds without corruption from airborne noise. This redesign of the original electret microphone could provide a method to continuously monitor lung and heart sounds from a subject regardless of their surrounding noise environment.
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Kawaguchi, Junki, and Mitsuharu Matsumoto. "Noise Reduction Combining a General Microphone and a Throat Microphone." Sensors 22, no. 12 (June 13, 2022): 4473. http://dx.doi.org/10.3390/s22124473.
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In this study, we propose a method to reduce noise from speech obtained from a general microphone using the information of a throat microphone. A throat microphone records a sound by detecting the vibration of the skin surface near the throat directly. Therefore, throat microphones are less prone to noise than ordinary microphones. However, as the acoustic characteristics of the throat microphone differ from those of ordinary microphones, its sound quality degrades. To solve this problem, this study aims to improve the speech quality while suppressing the noise of a general microphone by using the information recorded by a throat microphone as reference information to extract the speech signal in general microphones. In this paper, the framework of the proposed method is formulated, and several experiments are conducted to evaluate the noise suppression and speech quality improvement effects of the proposed method.
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Thompson, Stephen C. "Microphones: A bit of history." Journal of the Acoustical Society of America 153, no. 3_supplement (March 1, 2023): A106. http://dx.doi.org/10.1121/10.0018319.
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The earliest commercially successful microphone design was the carbon microphone, whose original patent was filed by Emile Berliner in 1877. An interesting variant of the basic carbon microphone was a dual diaphragm microphone patented by Granville Woods in 1883. The basic carbon microphone provided adequate performance at low cost and remained in service in the telephone system for at least the next century. Other microphone technologies can also be found in the early patent literature, though their commercial acceptance was delayed until electronic amplification was possible. The eventual development of vacuum tube amplifiers enabled the development of the condenser microphone, whose initial patent was filed in 1916. Condenser microphones provide the improvement of wide bandwidth and flat response compared to carbon microphones. The invention in the 1960s of high quality electret materials enabled the manufacture of low cost electret microphones that quickly displaced carbon mics in telephones and most commercial devices for at least the next 40 years. Then, in the early years of the 21st century, condenser microphones implemented in silicon, also known as MEMS microphones, took over in high all volume applications. This paper will discuss this history and the current state of the art.
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Zhao, Sipei, and Fei Ma. "A circular microphone array with virtual microphones based on acoustics-informed neural networks." Journal of the Acoustical Society of America 156, no. 1 (July 1, 2024): 405–15. http://dx.doi.org/10.1121/10.0027915.
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Acoustic beamforming aims to focus acoustic signals to a specific direction and suppress undesirable interferences from other directions. Despite its flexibility and steerability, beamforming with circular microphone arrays suffers from significant performance degradation at frequencies corresponding to zeros of the Bessel functions. To conquer this constraint, baffled or concentric circular microphone arrays have been studied; however, the former need a bulky baffle that interferes with the original sound field, whereas the latter require more microphones that increase the complexity and cost, both of which are undesirable in practical applications. To tackle this challenge, this paper proposes a circular microphone array equipped with virtual microphones, which resolves the performance degradation commonly associated with circular microphone arrays without resorting to physical modifications. The sound pressures at the virtual microphones are predicted from those measured by the physical microphones based on an acoustics-informed neural network, and then the sound pressures measured by the physical microphones and those predicted at the virtual microphones are integrated to design the beamformer. Experimental results demonstrate that the proposed approach not only eliminates the performance degradation but also suppresses spatial aliasing at high frequencies, thereby underscoring its promising potential.
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Sun, Huiyuan, Naoki Murata, Jihui Zhang, Tetsu Magariyachi, Prasanga N. Samarasinghe, Shigetoshi Hayashi, Thushara D. Abhayapala, and Tetsunori Itabashi. "Secondary channel estimation in spatial active noise control systems using a single moving higher order microphone." Journal of the Acoustical Society of America 151, no. 3 (March 2022): 1922–31. http://dx.doi.org/10.1121/10.0009837.
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Spatial active noise control (ANC) systems focus on minimizing unwanted acoustic noise over continuous spatial regions by generating anti-noise fields with secondary loudspeakers. Conventionally, error microphones are necessary inside the region to measure the channels from the secondary loudspeakers to the error microphones and record the residual sound field during the noise control. These error microphones highly limit the implementation of spatial ANC systems because of their impractical geometry and obstruction to the users from accessing the region. Recent advances, such as virtual sensing, focus on ANC with microphones placed away from the region. While these techniques relax the usage of error microphones during the noise control, an error microphone array remains necessary during the secondary channel estimation. In this paper, we propose a method to estimate secondary channels without using an error microphone array. Instead, a moving higher order microphone is applied to obtain the secondary channels from the secondary loudspeakers to the region of interest, which includes all desired error microphone locations. By simulation, we show that the proposed method is robust against various measuring errors introduced by the movement of the microphone and is suitable for the secondary channel estimation in spatial ANC systems.
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Gu, Wandong, Jiancheng Tao, and Xiaojun Qiu. "Coherence between error and acoustic reference microphones with two closely located noise sources." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 268, no. 6 (November 30, 2023): 2559–66. http://dx.doi.org/10.3397/in_2023_0375.
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It is well known that locating acoustic reference microphones near the error microphone can improve the coherence between reference and error signals in feedforward active noise control systems. In this paper, the effects of the number of reference microphones and the distance between noise sources and reference microphones on the coherence between the reference and error signals are investigated based on an analytical model. Two scenarios when the two noise sources are located on the same side or opposite sides of the error microphone are studied. The simulation results show that the coherence coefficient can reach 1.0 regardless of the distance between the noise sources and the reference microphones if the number of reference microphones is the same as that of the noise sources. When the number of reference microphones is less than that of noise sources, the coherence coefficient in the low frequency range decreases as the reference microphones move close to the noise sources.
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Shi, Li, Haishan Zou, Xiaojun Qiu, and Kai Chen. "An analysis of a feedback active noise control system using the remote microphone technique." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 268, no. 3 (November 30, 2023): 5634–44. http://dx.doi.org/10.3397/in_2023_0803.
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This paper examines strategies in virtual sensing feedback active noise control systems and compares the noise reduction performance of three strategies: using physical microphones for direct feedback control, using virtual microphones for direct feedback control, and using physical microphones for indirect virtual feedback control. For a primary sound field with incident waves in a single direction, simulation results indicate that employing a physical microphone for indirect virtual feedback control consistently yields superior noise reduction performance than that using a physical microphone for direct feedback control. However, if the causal requirement between physical and virtual microphones is not satisfied, the increase in the equivalent secondary path delay deteriorates noise reduction performance while utilizing physical microphones for indirect virtual feedback control. For a primary sound field containing waves from multiple directions, the simulation results show that the noise reduction performance is affected by both the coherence and causality between the physical and virtual microphone signals.
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Darras, Kevin, Bjørn Kolbrek, Andreas Knorr, and Volker Meyer. "Assembling cheap, high-performance microphones for recording terrestrial wildlife: the Sonitor system." F1000Research 7 (December 28, 2018): 1984. http://dx.doi.org/10.12688/f1000research.17511.1.
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Passive acoustic monitoring of wildlife requires microphones. Several cheap, high-performance open-source solutions currently exist for recording sounds, but all of them are still reliant on commercial microphones. Commercial microphones are relatively expensive, specialized on particular taxa, and often have opaque technical specifications. We designed Sonitor, an open-source microphone system to address all needs of ecologists that sample terrestrial wildlife acoustically. We evaluated the cost of our system and measured trade-offs that are seldom acknowledged but which universally limit microphones' functions: weatherproofing versus sound attenuation, windproofing versus transmission loss after rain, signal loss in long cables, and analog sound amplification and directivity with acoustic horns. We propose three microphone configurations suiting different budgets, sound qualities, and flexibility requirements, which all cover the entire sound frequency spectrum of sonant terrestrial wildlife at a fraction of the cost of commercial microphones.
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Tsunokuni, Izumi, Haruka Matsuhashi, and Yusuke Ikeda. "Direct sound field estimation based on sound source modeling with sparse equivalent sources." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 268, no. 2 (November 30, 2023): 6953–60. http://dx.doi.org/10.3397/in_2023_1038.
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The measurement of multi-point room impulse responses (RIRs) is challenging because it requires many microphones or a microphone traverse system. In our previous study, we proposed a spatial extrapolation method to measure early RIRs by the locally-positioned microphones. However, the estimation accuracy was degraded at a far distance from the microphones owing to the directivity of the sound source. In this study, we proposed a modeling method to estimate a direct sound field using a source radiation model based on the sparse equivalent source method. First, the source radiation was modeled using an enclosed microphone array. Then, we modeled the direct sound field using locally positioned microphones. The evaluation experiment was conducted in an anechoic chamber under 2.5D condition.
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Shah, Muhammad Ali, Ibrar Ali Shah, Duck-Gyu Lee, and Shin Hur. "Design Approaches of MEMS Microphones for Enhanced Performance." Journal of Sensors 2019 (March 6, 2019): 1–26. http://dx.doi.org/10.1155/2019/9294528.
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This paper reports a review about microelectromechanical system (MEMS) microphones. The focus of this review is to identify the issues in MEMS microphone designs and thoroughly discuss the state-of-the-art solutions that have been presented by the researchers to improve performance. Considerable research work has been carried out in capacitive MEMS microphones, and this field has attracted the research community because these designs have high sensitivity, flat frequency response, and low noise level. A detailed overview of the omnidirectional microphones used in the applications of an audio frequency range has been presented. Since the microphone membrane is made of a thin film, it has residual stress that degrades the microphone performance. An in-depth detailed review of research articles containing solutions to relieve these stresses has been presented. The comparative analysis of fabrication processes of single- and dual-chip omnidirectional microphones, in which the membranes are made up of single-crystal silicon, polysilicon, and silicon nitride, has been done, and articles containing the improved performance in these two fabrication processes have been explained. This review will serve as a starting guide for new researchers in the field of capacitive MEMS microphones.
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Nnonyelu, Chibuzo Joseph, Meng Jiang, Marianthi Adamopoulou, and Jan Lundgren. "Performance Analysis of Cardioid and Omnidirectional Microphones in Spherical Sector Arrays for Coherent Source Localization." Sensors 24, no. 23 (November 27, 2024): 7572. http://dx.doi.org/10.3390/s24237572.
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Traditional spherical sector microphone arrays using omnidirectional microphones face limitations in modal strength and spatial resolution, especially within spherical sector configurations. This study aims to enhance array performance by developing a spherical sector array employing first-order cardioid microphones. A model based on spherical sector harmonic (SSH) functions is introduced to extend the benefits of spherical harmonics to sector arrays. Modal strength analysis demonstrates that cardioid microphones in open spherical sectors enhance nonzero-order strengths and eliminate the nulls associated with spherical Bessel functions. We find that the spatial resolution of spherical cap arrays depends on the array’s maximum order and the limiting polar angle, but is independent of the microphone gain pattern. We assess direction-of-arrival (DOA) estimation performance for coherent wideband sources using the array manifold interpolation method, and compare cardioid and omnidirectional arrays through simulations in both open and rigid hemispherical configurations. The results indicate that cardioid arrays outperform omnidirectional ones on DOA estimation tasks, with performance improving alongside increased microphone directivity in the open hemispherical configuration. Specifically, hypercardioid microphones yielded the best results in the open configuration, while subcardioid microphones (without nulls) were optimal in rigid configurations. These findings demonstrate that spherical sector arrays of first-order cardioid microphones offer improved modal strength and DOA estimation capabilities over traditional omnidirectional arrays, providing significantly enhancing performance in spherical sector array processing.
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Oatley, James, and Craig Storey. "Applicability of MEMS microphones for environmental sound level monitoring." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 263, no. 6 (August 1, 2021): 875–85. http://dx.doi.org/10.3397/in-2021-1672.
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This paper explores the challenges associated with the integration of MEMS microphone technology into IEC 61672 classified or type-approved environmental sound level monitors. A comparison is drawn between MEMS microphones and electret condenser capsule microphones to highlight key performance differences within the technologies, and a basic integration method for both technologies is suggested. A review of the IEC 61672 and type-approval standards is conducted against the suggested integration method for a MEMS microphone; key shortcomings are reported and objectively reviewed. Development trends for MEMS microphones are explored, providing key insights into the progression of the technology against electret condenser capsule microphones. Furthermore, the evolution of environmental sound level monitoring systems is explored with a key focus on networked and sound localisation technology. The importance of MEMS microphones within the evolution of environmental sound level monitoring systems is presented alongside key arguments for the practical suitability of MEMS technology over electret condenser capsule technology.
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TOYOOKA, Shota, and Yoshinobu KAJIKAWA. "Optimal selection of reference microphone for active noise control." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 270, no. 7 (October 4, 2024): 4332–39. http://dx.doi.org/10.3397/in_2024_3446.
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This paper proposes two methods for selecting a reference microphone from multiple microphones, focusing on the correlation between the reference signal and the desired signal and the causality constraint of the signal, in order to effectively reduce noise arriving from various locations. In the first method, a threshold is set for the correlation between the reference signal and the desired signal, and a microphone that satisfies the causality constraint is selected as the reference microphone from microphones with correlation values that exceed this threshold. On the other hand, the second method selects a microphone with the highest correlation between the reference signal and the desired signal as the reference microphone among the microphones that satisfy the causality constraint. We demonstrate through simulation experiments that the active noise control system using these proposed methods can provide stable and effective noise reduction for noise arriving from various locations. It is also shown that both methods can reselect an appropriate reference microphone when the noise source moves.
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Awan, Shaheen N., Mohsin A. Shaikh, Maude Desjardins, Hagar Feinstein, and Katherine Verdolini Abbott. "The Effect of Microphone Frequency Response on Spectral and Cepstral Measures of Voice: An Examination of Low-Cost Electret Headset Microphones." American Journal of Speech-Language Pathology 31, no. 2 (March 10, 2022): 959–73. http://dx.doi.org/10.1044/2021_ajslp-21-00156.
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Purpose: The purpose of this study was to establish the frequency response of a selection of low-cost headset microphones that could be given to subjects for remote voice recordings and to examine the effect of microphone type and frequency response on key acoustic measures related to voice quality obtained from speech and vowel samples. Method: The frequency responses of three low-cost headset microphones were evaluated using pink noise generated via a head-and-torso model. Each of the headset microphones was then used to record a series of speech and vowel samples prerecorded from 24 speakers who represented a diversity of sex, age, fundamental frequency ( F o ), and voice quality types. Recordings were later analyzed for the following measures: smoothed cepstral peak prominence (CPP; dB), low versus high spectral ratio (L/H ratio; dB), CPP F o (Hz), and cepstral spectral index of dysphonia (CSID). Results: The frequency response of the microphones under test was observed to have nonsignificant effects on measures of the CPP and CPP F o , significant effects on the CSID in speech contexts, and strong and significant effects on the measure of spectral tilt (L/H ratio). However, the correlations between the various headset microphones and a reference precision microphone were excellent ( r s > .90). Conclusions: The headset microphones under test all showed the capability to track a wide range of diversity in the voice signal. Though the use of higher quality microphones that have demonstrated specifications is recommended for typical research and clinical purposes, low-cost electret microphones may be used to provide valid measures of voice, specifically when the same microphone and signal chain is used for the evaluation of pre- versus posttreatment change or intergroup comparisons.
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Hu, Siqi, Haitao Hu, Wei Xue, Dianyu Kang, and Jing Xiao. "Modeling and simulation study of acoustic response for dual-membrane capacitive MEMS microphone." Journal of Physics: Conference Series 2859, no. 1 (October 1, 2024): 012007. http://dx.doi.org/10.1088/1742-6596/2859/1/012007.
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Abstract As the micro-electro-mechanical systems (MEMS) technology matures, MEMS sensors have found widespread applications in mechatronics, robotics, and voice control. The high stability, high integration, and radio frequency interference resistance of MEMS microphones have rapidly led to their adoption in these domains, displacing electret condenser microphones.This article focuses on modeling and analyzing dual-membrane capacitive MEMS microphone, utilizing an lumped equivalent circuit model to calculate the microphone’s acoustic frequency response. Furthermore, the article employs the finite element method (FEM) to conduct a detailed analysis of the resonant frequency of the membrane, as well as to explore the effects of changes in the volume of the front and back chamber on the packaging performance of MEMS microphones. Finally, physical tests were conducted, and the simulation results of the two models showed considerable consistency with the curves obtained from the physical tests, verifying the accuracy of the models.
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Bentler, Ruth, Catherine Palmer, and Gustav H. Mueller. "Evaluation of a Second-Order Directional Microphone Hearing Aid: I. Speech Perception Outcomes." Journal of the American Academy of Audiology 17, no. 03 (March 2006): 179–89. http://dx.doi.org/10.3766/jaaa.17.3.4.
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This clinical trial was undertaken to evaluate the benefit obtained from hearing aids employing second-order adaptive directional microphone technology, used in conjunction with digital noise reduction. Data were collected for 49 subjects across two sites. New and experienced hearing aid users were fit bilaterally with behind-the-ear hearing aids using the National Acoustics Laboratory—Nonlinear version 1 (NAL-NL1) prescriptive method with manufacturer default settings for various parameters of signal processing (e.g., noise reduction, compression, etc.). Laboratory results indicated that (1) for the stationary noise environment, directional microphones provided better speech perception than omnidirectional microphones, regardless of the number of microphones; and (2) for the moving noise environment, the three-microphone option (whether in adaptive or fixed mode) and the two-microphone option in its adaptive mode resulted in better performance than the two-microphone fixed mode, or the omnidirectional modes.
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Schlieper, Roman, Song Li, Jürgen Peissig, and Stephan Preihs. "High-frequency acoustic impedance tube based on MEMS microphones." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 263, no. 2 (August 1, 2021): 4724–32. http://dx.doi.org/10.3397/in-2021-2810.
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Acoustic impedance tubes are commonly used to measure a test specimen's acoustic characteristics, such as reflection factor, absorption coefficient, or acoustic impedance, in combination with one or two condenser measurement microphones according to associated standards. In the development process of an impedance tube, the microphone diaphragm's size has an important role in the measurement quality. On the one hand, the microphone diameter has to be large enough to ensure the possibility of measuring at low sound pressure levels (SPLs), but on the other hand, the size of the microphone diaphragm should be small in order not to influence the sound propagation through the impedance tube due to the microphone coupling. Micro-Electro-Mechanical Systems (MEMS) microphones are recently widely applied in various acoustic applications due to their small size and high sensitivity. This paper proposes the development of an acoustic impedance tube equipped with 16 MEMS microphones and an inner diameter of 8 mm with an operating frequency range between 60 Hz and 16 kHz. The bottom port MEMS microphones are connected via a 1 mm hole to the tube. The system evaluation is based on standard test specimens like empty probe adapters, rigid termination, and porous absorbers.
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Zhang, Zhe, ChenLu Shi, Xiao Lv, and ZiHong Ling. "Active control of interior road noise using the remote microphone technique." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 265, no. 7 (February 1, 2023): 916–20. http://dx.doi.org/10.3397/in_2022_0130.
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A multichannel feedforward headrest system for the active control of interior road noise in a vehicle cabin is built. The remote microphone technique is applied, which enables the estimation of the sound pressure responses at the passenger's ear positions without direct deployment of error microphones there. The optimal observation filter for the remote microphone technique is formulated in a so-called training stage using signals measured at two error microphones on the passenger's ears and an array of four to five monitoring microphones on the headrest, passenger seat and vehicle ceiling. The estimation accuracy of the observation filter is investigated through simulations and road test. Regarding the causality error encountered in a certain test case where the passenger leans forward, thus making the noise signals arrive at the monitoring microphones prior to the error ones, a delay factor is added into the original remote microphone technique to correctly compensate for the time delay. The noise attenuation performance of the active headrest system is then experimentally and subjectively determined, indicating a larger noise abatement in a wider spatial environment by applying the remote microphone technique.
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Janota, Claus P., and Jeanette Olach Janota. "Intelligibility of Telephone Speech for the Hearing Impaired When Various Microphones Are Used for Acoustic Coupling." Journal of Speech, Language, and Hearing Research 34, no. 1 (February 1991): 183–88. http://dx.doi.org/10.1044/jshr.3401.183.
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This study set out to systematically measure the effect of using various candidate microphones for acoustic coupling of hearing aids to a telephone receiver. Intelligibility of words was determined for three microphones and three levels of interfering noise for a total of nine conditions. The subjects all had moderate hearing loss. It was found that microphones that exhibit pressure gradient sensitivity can, when properly positioned relative to the telephone receiver, increase intelligibility scores significantly. Results show that comparable listening performance is achieved with a pressure gradient microphone at a 10 dB higher level of interfering noise than with a microphone that is only sensitive to pressure. The results quantify the effect of microphones in acoustic coupling of hearing aids to telephones and point to a possible means for improving some types of hearing aids to facilitate telephone use for the hearing-impaired.
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Vennerod, Jakob, and Matthieu Lacolle. "Miniature optical MEMS microphone with 14dBA noise floor." Journal of the Acoustical Society of America 153, no. 3_supplement (March 1, 2023): A144. http://dx.doi.org/10.1121/10.0018444.
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This paper explains the fundamental technology used to create an optical microphone transducer. In recent years, microelectromechanical system (MEMS) capacitive microphones have demonstrated improved performance. State-of-the-art capacitive MEMS microphones can achieve SNR in the order of 73 dBA (21 dBA noise floor) with overall dynamic range in the order of 101 dB. There are fundamental challenges to driving the performance of capacitive MEMS microphone technology in very small packages to new heights. Piezoelectric MEMS microphones have not demonstrated SNR performance >65 dBA. The next breakthrough in miniature microphone technology will come from optical MEMS microphone technology. 80dB SNR (14 dBA noise floor) with 132 dB dynamic range (146dB maximum sound pressure level) has been achieved in a very small package. This paper will review the fundamentals of optical acoustic transduction and describe some of the approaches to miniaturization of the technology.
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Segers, Laurent, Jurgen Vandendriessche, Thibaut Vandervelden, Benjamin Johan Lapauw, Bruno da Silva, An Braeken, and Abdellah Touhafi. "CABE: A Cloud-Based Acoustic Beamforming Emulator for FPGA-Based Sound Source Localization." Sensors 19, no. 18 (September 10, 2019): 3906. http://dx.doi.org/10.3390/s19183906.
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Microphone arrays are gaining in popularity thanks to the availability of low-cost microphones. Applications including sonar, binaural hearing aid devices, acoustic indoor localization techniques and speech recognition are proposed by several research groups and companies. In most of the available implementations, the microphones utilized are assumed to offer an ideal response in a given frequency domain. Several toolboxes and software can be used to obtain a theoretical response of a microphone array with a given beamforming algorithm. However, a tool facilitating the design of a microphone array taking into account the non-ideal characteristics could not be found. Moreover, generating packages facilitating the implementation on Field Programmable Gate Arrays has, to our knowledge, not been carried out yet. Visualizing the responses in 2D and 3D also poses an engineering challenge. To alleviate these shortcomings, a scalable Cloud-based Acoustic Beamforming Emulator (CABE) is proposed. The non-ideal characteristics of microphones are considered during the computations and results are validated with acoustic data captured from microphones. It is also possible to generate hardware description language packages containing delay tables facilitating the implementation of Delay-and-Sum beamformers in embedded hardware. Truncation error analysis can also be carried out for fixed-point signal processing. The effects of disabling a given group of microphones within the microphone array can also be calculated. Results and packages can be visualized with a dedicated client application. Users can create and configure several parameters of an emulation, including sound source placement, the shape of the microphone array and the required signal processing flow. Depending on the user configuration, 2D and 3D graphs showing the beamforming results, waterfall diagrams and performance metrics can be generated by the client application. The emulations are also validated with captured data from existing microphone arrays.
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Titze, Ingo R., and William S. Winholtz. "Effect of Microphone Type and Placement on Voice Perturbation Measurements." Journal of Speech, Language, and Hearing Research 36, no. 6 (December 1993): 1177–90. http://dx.doi.org/10.1044/jshr.3606.1177.
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This study was conducted to explore the effects of microphone type (dynamic vs. condenser) and pattern (omnidirectional vs. cardioid) on the extraction of voice perturbation measures for sustained phonation. Also of interest were the effects of distance and angle between the source and the microphone. Four professional-grade and two consumer-grade microphones were selected for analysis. Synthesized phonation with different amplitude and frequency modulations at fundamental frequencies of 100 Hz and 300 Hz were presented over a loudspeaker. Human phonation was also included to test the validity of loudspeaker presentations. Three microphone distances (4 cm, 30 cm, 1 m) and three angles (0°, 45°, 90°) were used for microphone placement. Among the professional grade microphones, the cardioid condenser type had the smallest effect on perturbation measures. In general, condenser types gave better results than dynamic types. Microphones with an unbalanced output did not perform as well as those with balanced outputs. Microphone sensitivity and distance had the largest effect on perturbation measures, making it difficult to resolve normal vocal jitter at anything but a few centimeters from the mouth. Angle had little effect for short distances, but a greater effect for longer distances. These conclusions are preliminary because the sampling of microphones, distances, and signal types was very coarse. The study serves only to chart the course for future work.
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Milhomem, Thiago Antonio Bacelar. "Measuring directivity of half-inch measurement microphones." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 268, no. 3 (November 30, 2023): 5797–806. http://dx.doi.org/10.3397/in_2023_0827.
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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.
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Kim, Jonghoek. "Angle of Arrival Estimator Utilizing the Minimum Number of Omnidirectional Microphones." Journal of Marine Science and Engineering 12, no. 6 (May 24, 2024): 874. http://dx.doi.org/10.3390/jmse12060874.
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In sound signal processing, angle of arrival indicates the direction from which a propagating sound signal arrives at a point where multiple omnidirectional microphones are positioned. Considering a small underwater platform (e.g., underwater unmanned vehicle), this article addresses how to estimate a non-cooperative target’s signal direction utilizing the minimum number of omnidirectional microphones. It is desirable to use the minimum number of microphones, since one can reduce the cost and size of the platform by using small number of omnidirectional microphones. Suppose that each microphone measures a real-valued sound signal whose speed and frequency information are not known in advance. Since two microphones cannot determine a unique AOA solution, this study presents how to estimate the angle of arrival using a general configuration composed of three omnidirectional microphones. The effectiveness of the proposed angle of arrival estimator utilizing only three microphones is demonstrated by comparing it with the state-of-the-art estimation algorithm through computer simulations.
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Ghiselli, Sara, Erica Pizzol, Vincenzo Vincenti, Enrico Fabrizi, Daria Salsi, and Domenico Cuda. "Do Different Types of Microphones Affect Listening Effort in Cochlear Implant Recipients? A Pupillometry Study." Journal of Clinical Medicine 13, no. 4 (February 17, 2024): 1134. http://dx.doi.org/10.3390/jcm13041134.
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Background: It is known that subjects with a cochlear implant (CI) need to exert more listening effort to achieve adequate speech recognition compared to normal hearing subjects. One tool for assessing listening effort is pupillometry. The aim of this study is to evaluate the effectiveness of adaptive directional microphones in reducing listening effort for CI recipients. Methods: We evaluated listening in noise and listening effort degree (by pupillometry) in eight bimodal subjects with three types of CI microphones and in three sound configurations. Results: We found a correlation only between sound configurations and listening in noise score (p-value 0.0095). The evaluation of the microphone types shows worse scores in listening in noise with Opti Omni (+3.15 dB SNR) microphone than with Split Dir (+1.89 dB SNR) and Speech Omni (+1.43 dB SNR). No correlation was found between microphones and sound configurations and within the pupillometric data. Conclusions: Different types of microphones have different effects on the listening of CI patients. The difference in the orientation of the sound source is a factor that has an impact on the listening effort results. However, the pupillometry measurements do not significantly correlate with the different microphone types.
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TAşDELEN, Yusuf, Baran Berk BAğCı, Umut Murat GöK, and Osman Taha ŞEN. "Development of a gunshot location detection system based on acoustic data." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 270, no. 4 (October 4, 2024): 7495–500. http://dx.doi.org/10.3397/in_2024_3967.
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Determination of sound source location is a crucial problem, especially for the assault threat detection in crowded areas or military applications. The chief objective of this study is to design a microphone array to detect the location of explosion type sound sources, such as gunshot, which have significant amplitudes when compared to the background noise levels. Furthermore, the time arrival difference algorithm is utilized for the determination of time arrival differences and sound source location. Evidently, for stationary microphones that are positioned close to each other, determination of time arrival differences with high accuracy is a significant problem. Thus, a significant effort is given to enhance the accuracy of time arrival difference predictions via proper sampling. The design process of the microphone array is conducted computationally, where the sound data received by the microphones are generated. The time arrival differences are then calculated numerically, which are used for the prediction of sound source location. The number of microphones, relative positions of the microphones are determined accordingly via digital experiments. Finally, the designed microphone array is built and experiments are conducted. In conclusion, it is observed that the numerical model successfully predicts the sound source location of the computational model.
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Darras, Kevin, Bjørn Kolbrek, Andreas Knorr, Volker Meyer, Mike Zippert, and Arne Wenzel. "Assembling cheap, high-performance microphones for recording terrestrial wildlife: the Sonitor system." F1000Research 7 (February 19, 2021): 1984. http://dx.doi.org/10.12688/f1000research.17511.3.
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Passive acoustic monitoring of wildlife requires sound recording systems. Several cheap, high-performance, or open-source solutions currently exist for recording soundscapes, but all rely on commercial microphones. Commercial microphones are relatively expensive, specialized for particular taxa, and often have incomplete technical specifications. We designed Sonitor, an open-source microphone system to address all needs of ecologists that sample terrestrial wildlife acoustically. We evaluated the cost and durability of our system and measured trade-offs that are seldom acknowledged but which universally limit microphones' functions: weatherproofing versus sound attenuation, windproofing versus transmission loss after rain, signal loss in long cables, and analog sound amplification versus directivity with acoustic horns. We propose five microphone configurations suiting different budgets (from 8 to 33 EUR per unit), and fulfilling different sound quality and flexibility requirements. The Sonitor system consists of sturdy acoustic sensors that cover the entire sound frequency spectrum of sonant terrestrial wildlife at a fraction of the cost of commercial microphones.
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Darras, Kevin, Bjørn Kolbrek, Andreas Knorr, Volker Meyer, and Mike Zippert. "Assembling cheap, high-performance microphones for recording terrestrial wildlife: the Sonitor system." F1000Research 7 (November 5, 2019): 1984. http://dx.doi.org/10.12688/f1000research.17511.2.
Full textAbstract:
Passive acoustic monitoring of wildlife requires sound recording systems. Several cheap, high-performance open-source solutions currently exist for recording soundscapes, but all of them are still reliant on commercial microphones. Commercial microphones are relatively expensive, specialized for particular taxa, and often have incomplete technical specifications. We designed Sonitor, an open-source microphone system to address all needs of ecologists that sample terrestrial wildlife acoustically. We evaluated the cost and durability of our system and measured trade-offs that are seldom acknowledged but which universally limit microphones' functions: weatherproofing versus sound attenuation, windproofing versus transmission loss after rain, signal loss in long cables, and analog sound amplification versus directivity with acoustic horns. We propose five microphone configurations suiting different budgets (from 8 to 33 EUR per unit), and fulfilling different sound quality and flexibility requirements. The Sonitor system consists of sturdy acoustic sensors that cover the entire sound frequency spectrum of sonant terrestrial wildlife at a fraction of the cost of commercial microphones.
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Iwai, Kenta, Satoru Hase, and Yoshinobu Kajikawa. "Multichannel Feedforward Active Noise Control System with Optimal Reference Microphone SelectorBased on Time Difference of Arrival." Applied Sciences 8, no. 11 (November 19, 2018): 2291. http://dx.doi.org/10.3390/app8112291.
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In this paper, we propose a multichannel active noise control (ANC) system with an optimal reference microphone selector based on the time difference of arrival (TDOA). A multichannel feedforward ANC system using upstream reference signals can reduce various noises such as broadband noise by arranging reference microphones close to noise sources. However, the noise reduction performance of an ANC system degrades when the noise environment changes, such as the arrival direction. This is because some reference microphones do not satisfy the causality constraint that the unwanted noise propagates to the control point faster than the anti-noise used to cancel the unwanted noise. To solve this problem, we propose a multichannel ANC system with an optimal reference microphone selector. This selector chooses the reference microphones that satisfy the causality constraint based on the TDOA. Some experimental results demonstrate that the proposed system can choose the optimal reference microphones and effectively reduce unwanted acoustic noise.
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Riccardi, Peter J., Zane T. Rusk, John A. Case, Heui Young Park, Eric Rokni, and Stephen C. Thompson. "Low-cost measurement-grade microphone powered by MEMS elements and preamplifier housed in 3D printed enclosure." Journal of the Acoustical Society of America 152, no. 4 (October 2022): A50. http://dx.doi.org/10.1121/10.0015505.
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Acoustic measurement-grade microphones with flat frequency responses and adequate sensitivities are a necessary tool for many acousticians and vibroacoustic engineers. These microphones can often cost hundreds, if not thousands, of dollars. With the availability of microelectromechanical systems (MEMs) microphone elements and 3D printers, it is possible to construct drop-in replacements of these measurement grade microphones at the fraction of the cost. A MEMs system was designed with four elements in parallel to reduce uncorrelated noise. The system runs rail-to-rail on a 3.3VDC, Integrated Electronics Piezo-Electric (IEPE) powered preamplifier, producing a nominal sensitivity of 13 mV/Pa. The microphone body is 3D printed, and the final bill of material cost is less than 30 USD. The performance of the microphone was measured experimentally in a semi-anechoic chamber and verified by comparing the data to measurements of a commercial microphone: the PCB Piezotronics 378B02 ½” free-field microphone with a nominal sensitivity of 50 mV/Pa. This presentation will show that the MEMs microphone has a similar electrical noise floor, and lower acoustical noise floor below 1 kHz. The frequency response of the MEMs microphone is flat in the audio band through 10 kHz, making it a suitable replacement for acoustical measurements where its lower sensitivity can be tolerated.
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Stalder, Carly, and Stephane Leahy. "Comparative evaluation of omnidirectional and directional micro-electromechanical system microphone performance." Journal of the Acoustical Society of America 153, no. 3_supplement (March 1, 2023): A107. http://dx.doi.org/10.1121/10.0018324.
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As the need for directionality becomes a key requirement in audio applications, directional microphones have begun to enter the micro-electromechanical system (MEMS) design and market space, and their performance is approaching that of top-of-the-line omnidirectional MEMS microphones. This presentation examines and compares the performance limitations for both types of MEMS microphones and suggests more comprehensive methods of characterization that allow the qualities of directional MEMS microphones to be fully captured. Mechanical thermal noise caused by Brownian motion of air particles, measured with a laser Doppler vibrometer system, are shared and discussed for a variety of microphones and test structures. Parameters such as signal-to-noise ratio, total harmonic distortion, and frequency response are re-examined in order to capture the value of a microphone that exhibits directional behavior. Insights and suggestions are made for improvement and future work.
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Kuczynski, Jacek. "Developement of low-cost noise monitoring terminals (Nmt) based On MEMS microphones." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 265, no. 1 (February 1, 2023): 6657–65. http://dx.doi.org/10.3397/in_2022_1004.
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The article shows and discusses examples of Noise Monitoring Terminals (NMT) with MEMS microphones meeting class 1 and class 2 in accordance with the IEC 61672-1. The rapid development of MEMS microphones (Micro Electro-Mechanical Systems) in last decade years made it possible to use them in noise measurement instrumentation meeting the IEC 61672-1 specifications. Fifteen years ago, the available MEMS microphones offered only the 60 dB dynamic range, whereas modern MEMS microphones offer 100 dB dynamics! Such a wide dynamic range of MEMS microphones, along with their improved repeatability and a long-term stability, enabled the development of the low-cost noise monitoring terminals for noise monitoring. In particular one of such NMTs (SVANTEK SV 307) offers the measurement range of 25dBA Leq÷128 dBA Peak which proves to be optimal for urban noise monitoring applications. Even more hardware development possibilities are offered by implementation of fully digital MEMS microphones that are offered for the cost below 5 Euro. Such a low microphone cost enables the development of innovating designs for low-cost noise monitoring terminals with features such as a multi-microphones arrangement for a dynamic system check.
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Nnonyelu, Chibuzo J., Meng Jiang, and Jan Lundgren. "Spherical-sector harmonics domain processing for wideband source localization using spherical-sector array of directional microphones." Journal of the Acoustical Society of America 153, no. 3_supplement (March 1, 2023): A54. http://dx.doi.org/10.1121/10.0018140.
Full textAbstract:
The spherical microphone array can be uneconomical for applications where the sources arrive only from a known section of the sphere. For this reason, the spherical-sector harmonics was developed for processing spherical sector array. The orthonormal spherical sector harmonics (SSH) basis functions which accounts for the discontinuity arising from sectioning the sphere have been developed and shown to work for the array of omnidirectional microphones. In this work, the SSH basis functions are applied to far-field wideband sound source localization using spherical-sector array of first-order directional microphones (cardioid microphones). The array manifold interpolation method is used to produce the steered covariance matrix and the MUSIC algorithm applied for the direction of arrival estimation. The root-mean-square error performance of this spherical-sector array of the first-order cardioid microphones is compared against that of the omnidirectional microphones for different directions and signal-to-noise ratio.
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HABLOVIčOVá, Blanka, Vítězslav KřIVáNEK, Jan MACHANEC, and Petra MARKOVá. "Tyre/road noise frequency spectra analyses based on CPX measurement with six microphones." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 270, no. 8 (October 4, 2024): 3607–16. http://dx.doi.org/10.3397/in_2024_3348.
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Road traffic noise is affected by many factors, the road surface being one of them. The CPX method (ISO 11819-2) is the method used to measure tyre/road noise, i.e. noise generated by tyre-surface contact. Up to six microphones are positioned around reference tyre (ISO 11819-3) in specified positions as the standard requires. Two microphones are mandatory; four are optional. Conventional stone mastic asphalt, low-noise pavement, exposed aggregate cement concrete and cobblestones were measured by the CPX method using six microphones at a velocity of 80 km/h. Tyre/road noise frequency spectra from each microphone will be shown in this paper. Presented results should be used to discussion about positions or necessity of mandatory and optional microphones.
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Takeda, Aoi, Takeshi Nakaichi, and Nobuyuki Shibata. "Development of the ear insertion-type noise dosimeter with hearing protection." INTER-NOISE and NOISE-CON Congress and Conference Proceedings 268, no. 6 (November 30, 2023): 2249–56. http://dx.doi.org/10.3397/in_2023_0333.
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This study developed the ear insertion-type noise dosimeter with an earpiece. This dosimeter is composed of features two microphones, one located at the ear canal entrance, and the other positioned inside the ear canal. The dosimeter calculates LAeq and LApeak at one-second intervals. The device logs data from both the inside and outside microphones for up to 8 hours, and the recorded data can be conveniently accessed and displayed on a smartphone. The difference between the inside LAeq and the outside LAeq indicates the exposed level attenuated by the earpiece. The ear insertion-type noise dosimeter was evaluated in a free field by using an acoustic measurement manikin (KEMAR; GRAS Inc, Denmark). Results obtained in this study showed that the differences between the LAeq in the free field and that at the entrance of the ear canal were found to be slightly amplified by the pinna effect. Also, the LAeqs measured at the microphone that positioned inside the ear canal was consistently 3.2 dB lower on average than that recorded by KEMAR's microphone. These findings suggest that the LAeq on the eardrum can be estimated by using the LAeq measured by the inside microphone. Additionally, the sound attenuation value can be calculated by comparing the values obtained from the outside microphones with those from the inside microphones.
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Ershov, Victor, and Vadim Palchikovskiy. "DESIGNING PLANAR MICROPHONE ARRAY FOR SOUND SOURCE LOCALIZATION." Akustika 32 (March 1, 2019): 123–29. http://dx.doi.org/10.36336/akustika201932123.
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Mathematical background for designing planar microphone array for localization of sound sources are described shortly. The designing is based on optimization of objective function, which is maximum dynamic range of sound source localization. The design parameters are radial coordinates (distance along the beam from the center of the array) and angle coordinates (beam inclination) of the microphones. It is considered the arrays with the same radial coordinates of the microphones for each beam and the independent radial coordinates of each microphone, as well as the same inclination angle for all beams and the individual inclination angle of each beam. As constraints, it is used the minimum allowable distance between two adjacent microphones, and minimum and maximum diameter of the working area of the array. The solution of the optimization problem is performed by the Minimax method. An estimation of the resolution quality of designed arrays was carried out based on localization of three monopole sources. The array of 3 m in diameter without inclination of the beams and with different radial coordinates of the microphones on each beam was found to be the most efficient configuration among the considered ones.
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Bulkin, Vladislav, and Stanislav Kozlov. "Construction of a system for assessing the acoustic signal attenuation." E3S Web of Conferences 542 (2024): 05004. http://dx.doi.org/10.1051/e3sconf/202454205004.
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Acoustic noise is one of the main factors affecting humans. The task of the urban environment operational monitoring is urgent as it will facilitate the solution of population protection problem. This problem can be solved by carrying out measurements in a full-spectrum mode with an assessment of the acoustic signal attenuation along typical noise propagation paths deep into residential areas. It has been shown that monitoring the noise level or its changes within the propagation limits deep into residential areas is possible provided that measuring microphones are located along this path at least in two characteristic points. An estimate of the potentially required microphone separation distance in case of using a system with two microphones is given. It is determined that at the preliminary stage of the measuring system construction, the measuring microphones’ location at a distance of 1 km from each other is sufficient. A block diagram of the meter, including two radio transmitters combined with the microphones and a receiver-analyzer, is proposed. It has been determined that acoustic signal delay on the way to the second microphone must be taken into account and the system calibration problem must be solved.
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Md. Easir Arafat, Indraneel Misra, and Md. Ekramul Hamid. "A comparative study for throat microphone speech enhancement with different approaches." International Journal of Science and Research Archive 13, no. 1 (September 30, 2024): 850–59. http://dx.doi.org/10.30574/ijsra.2024.13.1.1631.
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Throat microphones (TM) offer significant advantages in noisy environments by capturing speech signals directly from the throat, thus minimizing external noise. However, TM signals often lack clarity and intelligibility compared to conventional microphones. This paper presents a comparative study of three prominent feature extraction techniques—Mel-frequency cepstral coefficients (MFCC), Linear Predictive Cepstral coefficients (LPCC), Perceptual Linear Prediction (PLP) for enhancing speech captured by throat microphones. Each technique is evaluated based on its ability to enhance speech clarity and reduce noise interference. Experimental results on the ATR503 dataset, consisting of throat and close-talk microphone recordings, reveal that LPCC achieved an average Signal-to-Noise Ratio (SNR) improvement of 3dB and a Perceptual Evaluation of Speech Quality (PESQ) score increase of 1.3133 and 0.9553 compared to MFCC and PLP. In subjective evaluations the highest mean rating of 8.46 for LPCC indicates it was perceived as the most intelligible and clear. LPC spectra analysis demonstrates that Linear Predictive Cepstral Coefficients (LPCC) in retrieving missing frequencies in speech captured by throat microphones. These findings suggest that LPCC is a robust method for throat microphone speech enhancement, offering significant improvements in speech intelligibility and quality in noisy environments.
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GE, SHUZHI SAM, AI POH LOH, and FENG GUAN. "ROBUST SOUND LOCALIZATION USING SMALL NUMBER OF MICROPHONES." International Journal of Information Acquisition 02, no. 01 (March 2005): 1–22. http://dx.doi.org/10.1142/s0219878905000386.
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In this paper, robust sound localization is proposed for systems with spatially distributed microphones fewer than 4, the minimum required criteria for 3D sound localization in the literature, in an effort to address the robustness issues of microphone failures. It is shown that, for three- and two-microphone systems, two and four samples of the Interaural Time Difference (ITD) measurements are required respectively. In a one-microphone system, five samples of the Interaural Intensity Difference (IID) measurement should be used instead. These investigations are important because if one microphone fails, the remaining system can still function to locate the sound source without significant impact on the localization system. In addition, our investigation also shows that a 3-microphone system can estimate the azimuth and elevation simultaneously, which usually requires 4 microphones in the literature. Simulation and experimental results are presented to illustrate the performance of a three-microphone system. Results show that the system can locate the sound source with satisfactory accuracy.
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Wright, Mark Peter. "The Thing about Microphones." Leonardo Music Journal 26 (December 2016): 60–63. http://dx.doi.org/10.1162/lmj_a_00977.
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This article examines the microphone and its connective political and nonhuman ecologies. A media archaeological excavation of Leon Theremin’s role in the development of a specific bugging device (“The Thing”) facilitates discussion throughout. Situating the microphone within a networked history of power relations and ethical consequences, the author draws upon contexts of surveillance, parasites and horror in order to ask whether microphones are agential actors and, if so, what the consequences might be.
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Ha, Sangwoo, Jungwoo Kim, Hyun-guk Kim, and Semyung Wang. "Horizontal Active Noise Control-Based Wave Field Reproduction Using a Single Circular Array in 3D Space." Applied Sciences 12, no. 20 (October 12, 2022): 10245. http://dx.doi.org/10.3390/app122010245.
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In this paper, we propose horizontal active noise control (ANC) using two-dimensional wave field information alone. By reducing the control space to a horizontal plane, the number of microphones and speakers was considerably reduced compared with ANC systems using three-dimensional wave field information. The radii of the reference, microphone, and loudspeaker array were determined based on the wave field reproduction error. Accordingly, the simulation and experimental results of the proposed ANC system were presented based on the use of five microphones and loudspeakers using conventional ANC algorithms. Overall, an average noise reduction of 20 dB was observed inside the microphone array with a radius of 0.5 m for tonal noise at 200 Hz. This performance is acceptable with a drastically reduced number of microphones and speakers. The findings of this study, along with further research conducted in a reverberant room, represent a significant contribution to global ANC commercialization.
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Cord, Mary T., Rauna K. Surr, Brian E. Walden, and Ole Dyrlund. "Relationship between Laboratory Measures of Directional Advantage and Everyday Success with Directional Microphone Hearing Aids." Journal of the American Academy of Audiology 15, no. 05 (May 2004): 353–64. http://dx.doi.org/10.3766/jaaa.15.5.3.
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The improvement in speech recognition in noise obtained with directional microphones compared to omnidirectional microphones is referred to as the directional advantage. Laboratory studies have revealed substantial differences in the magnitude of the directional advantage across hearing-impaired listeners. This investigation examined whether persons who were successful users of directional microphone hearing aids in everyday living tended to obtain a larger directional advantage in the test booth than persons who were unsuccessful users. Results revealed that the mean directional advantage did not differ significantly between patients who used the directional mode regularly and those who reported little or no benefit from directional microphones in daily living and, therefore, tended to leave their hearing aids set in the default omnidirectional mode. Success with directional microphone hearing aids in everyday living, therefore, cannot be reliably predicted by the magnitude of the directional advantage obtained in the clinic.
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Putri, Chery Chaen, and Denny Hermawanto. "MICROPHONE FRONT CAVITY DEPTH MEASUREMENT USING NON-CONTACT METHOD AT NATIONAL MEASUREMENT STANDARDS – NATIONAL STANDARDIZATION AGENCY OF INDONESIA." Jurnal Standardisasi 21, no. 2 (May 9, 2019): 143. http://dx.doi.org/10.31153/js.v21i2.748.
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<p>Front cavity depth of microphone has influence in the determination of laboratory standard microphone sensitivity. Therefore this parameter is included in the measurement uncertainty budget. To determine microphone sensitivity accurately, it is necessary to measure the actual front cavity depth instead of using nominal value. This paper explains the measurement of LS1P and LS2P standard microphone front cavity depth using optical depth measurement facilities at SNSU-BSN and its effect on determining microphone sensitivity. The measurement was obtained from 4 positions distributed over the diaphragm for each microphone. The front cavity depth measurement result for LS1P is 1,94 mm ± 0,01 mm and for LS2P is 0,48 mm ± 0,01 mm These results comply with IEC 61094 Measurement Microphones-Part 2: Primary Method for Pressure Calibration of Laboratory Standard Microphones by Reciprocity Technique as the results are within the permissible range.</p>
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Gustafson, Samantha J., Todd A. Ricketts, and Erin M. Picou. "Individual Differences Offer Insight Into Clinical Recommendations for Directional and Remote Microphone Technology Use in Children." Journal of Speech, Language, and Hearing Research 64, no. 2 (February 17, 2021): 635–50. http://dx.doi.org/10.1044/2020_jslhr-20-00281.
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Purpose This study sought to evaluate the effects of common hearing aid microphone technologies on speech recognition and listening effort, and to evaluate potential predictive factors related to microphone benefits for school-age children with hearing loss in a realistic listening situation. Method Children ( n = 17, ages 10–17 years) with bilateral, sensorineural hearing loss were fitted with hearing aids set to include three programs: omnidirectional, adaptive directional, and omnidirectional + remote microphone. Children completed a dual-task paradigm in a moderately reverberant room. The primary task included monosyllabic word recognition, with target speech presented at 60 dB A from 0° (front) or 180° (back) azimuth. The secondary task was a “go/no-go,” visual shape-recognition task. Multitalker babble noise created a +5 dB SNR. Children were evaluated in two speaker conditions (front, back) using all three hearing aid programs. The remote microphone transmitter remained at the front speaker throughout testing. Speech recognition performance was calculated from the primary task while listening effort was measured as response time during the secondary task. Results Speech presented from the back significantly increased listening effort and caused a reduction in speech perception when directional and remote microphones were used. Considerable variability was found in pattern of benefit across microphones and source location. Clinical measures did not predict benefit patterns with directional or remote microphones; however, child age and performance with omnidirectional microphones did. Conclusions When compared to a traditional omnidirectional setting, the directional and remote microphone configurations evaluated in this study have the potential to provide benefit for some children and increase difficulty for others when used in dynamic environments. A child's performance with omnidirectional hearing aids could be used to better inform clinical recommendations for these technologies.
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Mirabilii, Daniele, and Emanuël A. Habets. "Pseudospectrum-based methods for estimating the wind speed and direction based on closely spaced microphone signals." Journal of the Acoustical Society of America 153, no. 3_supplement (March 1, 2023): A142. http://dx.doi.org/10.1121/10.0018438.
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Acoustic array processing can be employed to measure the wind speed and direction based on microphone signals. Turbulent pressure fluctuations picked up by microphones are referred to as wind noise. According to Taylor's frozen turbulence hypothesis, turbulent eddies retain their shape while advecting at nearly the mean wind speed and in the wind direction. It follows that wind noise propagates accordingly across a microphone array when the inter-microphone distance is smaller than the turbulence wavelength. This property can be exploited to track the orientation of the turbulence advection, and hence to characterize the wind flow. We propose beamforming and signal subspace-based methods to estimate the wind speed and direction using a compact microphone array. In particular, the pseudospectrum of measured wind noise is computed against candidate pairs of wind speed and direction. The wind speed and direction estimates are then obtained as the maximizers of the pseudospectrum. In addition, we extend an existing time difference of arrival-based method originally derived for three microphones to an arbitrary number of microphones. We evaluate the estimation accuracy of the proposed methods separately for the wind speed and direction. Possible applications include highly integrable, portable, and inexpensive anemometers for smart sensors or action cameras.
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Zhang, Xinyi, Alessandro Braga, Arian Shamei, and Rachel Bouserhal. "A comparison of voice quality measures across in-ear, outer-ear, and standard microphone recordings." Journal of the Acoustical Society of America 156, no. 4_Supplement (October 1, 2024): A55. https://doi.org/10.1121/10.0035092.
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Voice quality provides significant insights into one’s health. Recent advancements in intra-aural devices have enabled the longitudinal monitoring of speech and its changes. These wearables can record speech from in-ear microphones (IEMs) and outer-ear microphones (OEMs). The laboratory gold standard of speech recording uses a high-sensitivity low-noise microphone placed in front of the mouth to capture signals accurately. Speech recorded inside an occluded ear canal differs considerably from this due to the bone-and-tissue conduction and the effect of ear occlusion. OEMs record speech transmitted solely through air like the standard microphones but are more influenced by indirect air conduction due to their positioning. This study compares voice quality measures across IEM, OEM, and the standard microphone (REF) using an open-access database. We employed linear mixed-effect modeling to analyze the effects of different microphone recordings on these metrics. Results indicate that while pitch control remains relatively intact, IEM and OEM exhibit varying deviations from REF. Overall, OEM resembles REF in jitter, whereas IEM is higher. IEM resembles REF in shimmer, while OEM is lower. For HNR, both IEM and OEM are higher than REF. Sex-based difference was also observed, and the correlation between microphone differences and fundamental frequency was explored.
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Catur Bhakti, Muhammad Agni, and Wandy Wandy. "Analisis Perbedaan Tekanan Suara di Mikrofon Internal dengan Mikrofon Eksternal pada Aplikasi Soundmeter berbasis Ponsel." Jurnal Teknologi 14, no. 2 (December 6, 2021): 114–21. http://dx.doi.org/10.34151/jurtek.v14i2.3532.
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Mobile phones and tablets installed with a soundmeter application can be used as an alternative for recording noise pollution to support Occupational Health and Safety (OHS). Mobile and tablet applications that have been calibrated with a soundmeter has the ability to record sound pressure which could be a solution for data recording. An external microphone can be attached to mobile phones and tablets to improve the quality of recorded data from sound sources. This research aims to analyze and determine the differences between the sound captured by internal and external microphones, and then compared them with results from a soundmeter device. This research was conducted using quantitative methods with primary data being analyzed descriptively. It was concluded from this research that there were significant increments in sound pressure using external microphones compared to internal microphones, with increment average using external microphone of 6.11 dBA on mobile phone, and 8.09 dbA on tablet.
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da Silva, Bruno, An Braeken, Kris Steenhaut, and Abdellah Touhafi. "Design Considerations When Accelerating an FPGA-Based Digital Microphone Array for Sound-Source Localization." Journal of Sensors 2017 (2017): 1–20. http://dx.doi.org/10.1155/2017/6782176.
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The use of microphone arrays for sound-source localization is a well-researched topic. The response of such sensor arrays is dependent on the quantity of microphones operating on the array. A higher number of microphones, however, increase the computational demand, making real-time response challenging. In this paper, we present a Filter-and-Sum based architecture and several acceleration techniques to provide accurate sound-source localization in real-time. Experiments demonstrate how an accurate sound-source localization is obtained in a couple of milliseconds, independently of the number of microphones. Finally, we also propose different strategies to further accelerate the sound-source localization while offering increased angular resolution.