[An implantable microphone for electronic hearing aids].

PubMed

Leysieffer, H; Müller, G; Zenner, H P

1997-10-01

Fully implantable hearing aids and cochlea implants of the future require an implantable microphone. A hermetically sealed implantable microphone based on the idea of a microphone implanted in the posterior wall of the auditory canal, as suggested by Ohno et al. in 1988, is presented. Through consistent technological and clinical design optimization, it was possible to achieve a membrane diameter of only 4.5 mm (as opposed to 8 mm in the Japanese system) and a significant volume reduction of nearly 50%. The microphone weights only 0.4 g. In spite of this miniaturization, the performance characteristics of the microphone equal those of the Japanese model or are superior. The sound-pressure transfer function shows a very small ripple and the bandwidth amounts to approximately 10 kHz. Because of its high tuning and high no-load resonance frequency, the microphone is mostly insensitive to post-operational changes to the loading mass on the microphone membrane initiated by the covering skin of the auditory canal. The sound-pressure transfer factor at 1000 Hz is approximately 1.5 mV/Pa. Using different manufacturing technologies, this value can be increased in the range of 6-8 dB with a corresponding reduction in bandwidth. Due to the small mass, the microphone is highly insensitive to environmental mechanical disturbances. The module is made of pure titanium and is hermetically sealed according to Mil-Std 883 D. Full metal encapsulation and additional internal electronic components protect the microphone well against environmental electromagnetic influences (EMC).

Design, modeling, and fabrication of crab-shape capacitive microphone using silicon-on-isolator wafer

NASA Astrophysics Data System (ADS)

Ganji, Bahram Azizollah; Sedaghat, Sedighe Babaei; Roncaglia, Alberto; Belsito, Luca; Ansari, Reza

2018-01-01

This paper presents design, modeling, and fabrication of a crab-shape microphone using silicon-on-isolator (SOI) wafer. SOI wafer is used to prevent the additional deposition of sacrificial and diaphragm layers. The holes have been made on diaphragm to prevent back plate etching. Dry etching is used for removing the sacrificial layer, because wet etching causes adhesion between the diaphragm and the back plate. Crab legs around the perforated diaphragm allow for improving the microphone performance and reducing the mechanical stiffness and air damping of the microphone. In this structure, the supply voltage is decreased due to the uniform deflection of the diaphragm due to the designed low-K (spring constant) structure. An analytical model of the structure for description of microphone behavior is presented. The proposed method for estimating the basic parameters of the microphone is based on the calculation of the spring constant using the energy method. The microphone is fabricated using only one mask to pattern the crab-shape diaphragm, resulting in a low-cost and easy fabrication process. The diaphragm size is 0.3 mm×0.3 mm, which is smaller than the conventional microelectromechanical systems capacitive microphone. The results show that the analytical equations have a good agreement with measurement results. The device has the pull-in voltage of 14.3 V, a resonant frequency of 90 kHz, an open-circuit sensitivity of 1.33 mV/Pa under bias voltage of 5 V. Comparing with previous works, this microphone has several advantages: SOI wafer decreases the fabrication process steps, the microphone is smaller than the previous works, and crab-shape diaphragm improves the microphone performances.

Phase Calibration of Microphones by Measurement in the Free-field

NASA Technical Reports Server (NTRS)

Shams, Qamar A.; Bartram, Scott M.; Humphreys, William M.; Zuckewar, Allan J.

2006-01-01

Over the past several years, significant effort has been expended at NASA Langley developing new Micro-Electro-Mechanical System (MEMS)-based microphone directional array instrumentation for high-frequency aeroacoustic measurements in wind tunnels. This new type of array construction solves two challenges which have limited the widespread use of large channel-count arrays, namely by providing a lower cost-per-channel and a simpler method for mounting microphones in wind tunnels and in field-deployable arrays. The current generation of array instrumentation is capable of extracting accurate noise source location and directivity on a variety of airframe components using sophisticated data reduction algorithms [1-2]. Commercially-available MEMS microphones are condenser-type devices and have some desirable characteristics when compared with conventional condenser-type microphones. The most important advantages of MEMS microphones are their size, price, and power consumption. However, the commercially-available units suffer from certain important shortcomings. Based on experiments with array prototypes, it was found that both the bandwidth and the sound pressure limit of the microphones should be increased significantly to improve the performance and flexibility of the microphone array [3]. It was also desired to modify the packaging to eliminate unwanted Helmholtz resonance s exhibited by the commercial devices. Thus, new requirements were defined as follows: Frequency response: 100 Hz to 100 KHz (+/-3dB) Upper sound pressure limit: Design 1: 130 dB SPL (THD less than 5%) Design 2: 150-160 dB SPL (THD less than 5%) Packaging: 3.73 x 6.13 x 1.3 mm can with laser-etched lid. In collaboration with Novusonic Acoustic Innovation, NASA modified a Knowles SiSonic MEMS design to meet these new requirements. Coupled with the design of the enhanced MEMS microphones was the development of a new calibration method for simultaneously obtaining the sensitivity and phase response of the devices over their entire broadband frequency range. Traditionally, electrostatic actuators (EA) have been used to characterize air-condenser microphones; however, MEMS microphones are not adaptable to the EA method due to their construction and very small diaphragm size [4]. Hence a substitution based, free-field method was developed to calibrate these microphones at frequencies up to 80 kHz. The technique relied on the use of a random, ultrasonic broadband centrifugal sound source located in a small anechoic chamber. The free-field sensitivity (voltage per unit sound pressure) was obtained using the procedure outlined in reference 4. Phase calibrations of the MEMS microphones were derived from cross spectral phase comparisons between the reference and test substitution microphones and an adjacent and invariant grazing-incidence 1/8-inch standard microphone. The free-field calibration procedure along with representative sensitivity and phase responses for the new high-frequency MEMS microphones are presented here.

Intelligibility of Telephone Speech for the Hearing Impaired When Various Microphones Are Used for Acoustic Coupling.

ERIC Educational Resources Information Center

Janota, Claus P.; Janota, Jeanette Olach

1991-01-01

Various candidate microphones were evaluated for acoustic coupling of hearing aids to a telephone receiver. Results from testing by 9 hearing-impaired adults found comparable listening performance with a pressure gradient microphone at a 10 decibel higher level of interfering noise than with a normal pressure-sensitive microphone. (Author/PB)

Methods for Room Acoustic Analysis and Synthesis using a Monopole-Dipole Microphone Array

NASA Technical Reports Server (NTRS)

Abel, J. S.; Begault, Durand R.; Null, Cynthia H. (Technical Monitor)

1998-01-01

In recent work, a microphone array consisting of an omnidirectional microphone and colocated dipole microphones having orthogonally aligned dipole axes was used to examine the directional nature of a room impulse response. The arrival of significant reflections was indicated by peaks in the power of the omnidirectional microphone response; reflection direction of arrival was revealed by comparing zero-lag crosscorrelations between the omnidirectional response and the dipole responses to the omnidirectional response power to estimate arrival direction cosines with respect to the dipole axes.

Outlier Detection for Sensor Systems (ODSS): A MATLAB Macro for Evaluating Microphone Sensor Data Quality.

PubMed

Vasta, Robert; Crandell, Ian; Millican, Anthony; House, Leanna; Smith, Eric

2017-10-13

Microphone sensor systems provide information that may be used for a variety of applications. Such systems generate large amounts of data. One concern is with microphone failure and unusual values that may be generated as part of the information collection process. This paper describes methods and a MATLAB graphical interface that provides rapid evaluation of microphone performance and identifies irregularities. The approach and interface are described. An application to a microphone array used in a wind tunnel is used to illustrate the methodology.

High-temperature microphone system. [for measuring pressure fluctuations in gases at high temperature

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J. (Inventor)

1979-01-01

Pressure fluctuations in air or other gases in an area of elevated temperature are measured using a condenser microphone located in the area of elevated temperature and electronics for processing changes in the microphone capacitance located outside the area the area and connected to the microphone by means of high-temperature cable assembly. The microphone includes apparatus for decreasing the undesirable change in microphone sensitivity at high temperatures. The high temperature cable assembly operates as a half-wavelength transmission line in an AM carrier system and maintains a large temperature gradient between the two ends of the cable assembly. The processing electronics utilizes a voltage controlled oscillator for automatic tuning thereby increasing the sensitivity of the measuring apparatus.

Quantifying Errors in Jet Noise Research Due to Microphone Support Reflection

NASA Technical Reports Server (NTRS)

Nallasamy, Nambi; Bridges, James

2002-01-01

The reflection coefficient of a microphone support structure used insist noise testing is documented through tests performed in the anechoic AeroAcoustic Propulsion Laboratory. The tests involve the acquisition of acoustic data from a microphone mounted in the support structure while noise is generated from a known broadband source. The ratio of reflected signal amplitude to the original signal amplitude is determined by performing an auto-correlation function on the data. The documentation of the reflection coefficients is one component of the validation of jet noise data acquired using the given microphone support structure. Finally. two forms of acoustic material were applied to the microphone support structure to determine their effectiveness in reducing reflections which give rise to bias errors in the microphone measurements.

The impact of the microphone position on the frequency analysis of snoring sounds.

PubMed

Herzog, Michael; Kühnel, Thomas; Bremert, Thomas; Herzog, Beatrice; Hosemann, Werner; Kaftan, Holger

2009-08-01

Frequency analysis of snoring sounds has been reported as a diagnostic tool to differentiate between different sources of snoring. Several studies have been published presenting diverging results of the frequency analyses of snoring sounds. Depending on the position of the used microphones, the results of the frequency analysis of snoring sounds vary. The present study investigated the influence of different microphone positions on the outcome of the frequency analysis of snoring sounds. Nocturnal snoring was recorded simultaneously at six positions (air-coupled: 30 cm middle, 100 cm middle, 30 cm lateral to both sides of the patients' head; body contact: neck and parasternal) in five patients. The used microphones had a flat frequency response and a similar frequency range (10/40 Hz-18 kHz). Frequency analysis was performed by fast Fourier transformation and frequency bands as well as peak intensities (Peaks 1-5) were detected. Air-coupled microphones presented a wider frequency range (60 Hz-10 kHz) compared to contact microphones. The contact microphone at cervical position presented a cut off at frequencies above 300 Hz, whereas the contact microphone at parasternal position revealed a cut off above 100 Hz. On an exemplary base, the study demonstrates that frequencies above 1,000 Hz do appear in complex snoring patterns, and it is emphasised that high frequencies are imported for the interpretation of snoring sounds with respect to the identification of the source of snoring. Contact microphones might be used in screening devices, but for a natural analysis of snoring sounds the use of air-coupled microphones is indispensable.

49 CFR 325.77 - Computation of open site requirements-nonstandard sites.

Code of Federal Regulations, 2010 CFR

2010-10-01

... microphone target point is other than 50 feet (15.2 m), the test site must be an open site within a radius... microphone target point. (b) Plan view diagrams of nonstandard test sites are shown in Figures 3 and 4... (18.3 m) distance between the microphone location point and the microphone target point. (See § 325.79...

49 CFR 325.77 - Computation of open site requirements-nonstandard sites.

Code of Federal Regulations, 2011 CFR

2011-10-01

... microphone target point is other than 50 feet (15.2 m), the test site must be an open site within a radius... microphone target point. (b) Plan view diagrams of nonstandard test sites are shown in Figures 3 and 4... (18.3 m) distance between the microphone location point and the microphone target point. (See § 325.79...

Method for extracting forward acoustic wave components from rotating microphone measurements in the inlets of turbofan engines

NASA Technical Reports Server (NTRS)

Cicon, D. E.; Sofrin, T. G.

1995-01-01

This report describes a procedure for enhancing the use of the basic rotating microphone system so as to determine the forward propagating mode components of the acoustic field in the inlet duct at the microphone plane in order to predict more accurate far-field radiation patterns. In addition, a modification was developed to obtain, from the same microphone readings, the forward acoustic modes generated at the fan face, which is generally some distance downstream of the microphone plane. Both these procedures employ computer-simulated calibrations of sound propagation in the inlet duct, based upon the current radiation code. These enhancement procedures were applied to previously obtained rotating microphone data for the 17-inch ADP fan. The forward mode components at the microphone plane were obtained and were used to compute corresponding far-field directivities. The second main task of the program involved finding the forward wave modes generated at the fan face in terms of the same total radial mode structure measured at the microphone plane. To obtain satisfactory results with the ADP geometry it was necessary to limit consideration to the propagating modes. Sensitivity studies were also conducted to establish guidelines for use in other fan configurations.

Sub-Surface Windscreen for Outdoor Measurement of Infrasound

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J. (Inventor); Shams, Qamar A. (Inventor)

2014-01-01

A windscreen is configured for measuring outdoor infrasonic sound. The windscreen includes a container and a microphone. The container defines a chamber. The microphone is disposed in the chamber and can be operatively supported by the floor. The microphone is configured for detecting infrasonic sound. The container is advantageously formed from material that exhibits an acoustic impedance of between 0 and approximately 3150 times the acoustic impedance of air. A reflector plate may be disposed in the container. The reflector plate operatively can support the microphone and provides a doubling effect of infrasonic pressure at the microphone.

MEMS microphone innovations towards high signal to noise ratios (Conference Presentation) (Plenary Presentation)

NASA Astrophysics Data System (ADS)

Dehé, Alfons

2017-06-01

After decades of research and more than ten years of successful production in very high volumes Silicon MEMS microphones are mature and unbeatable in form factor and robustness. Audio applications such as video, noise cancellation and speech recognition are key differentiators in smart phones. Microphones with low self-noise enable those functions. Backplate-free microphones enter the signal to noise ratios above 70dB(A). This talk will describe state of the art MEMS technology of Infineon Technologies. An outlook on future technologies such as the comb sensor microphone will be given.

Assessment of Operational Progress of NASA Langley Developed Windshield and Microphone for Infrasound

DTIC Science & Technology

2013-04-01

Assessment of Operational Progress of NASA Langley Developed Windshield and Microphone for Infrasound by W.C. Kirkpatrick Alberts, II...Windshield and Microphone for Infrasound W.C. Kirkpatrick Alberts, II, Stephen M. Tenney, and John M. Noble Sensors and Electron Devices Directorate...2013 4. TITLE AND SUBTITLE Assessment of Operational Progress of NASA Langley Developed Windshield and Microphone for Infrasound 5a. CONTRACT

Combined Use of Standard and Throat Microphones for Measurement of Acoustic Voice Parameters and Voice Categorization.

PubMed

Uloza, Virgilijus; Padervinskis, Evaldas; Uloziene, Ingrida; Saferis, Viktoras; Verikas, Antanas

2015-09-01

The aim of the present study was to evaluate the reliability of the measurements of acoustic voice parameters obtained simultaneously using oral and contact (throat) microphones and to investigate utility of combined use of these microphones for voice categorization. Voice samples of sustained vowel /a/ obtained from 157 subjects (105 healthy and 52 pathological voices) were recorded in a soundproof booth simultaneously through two microphones: oral AKG Perception 220 microphone (AKG Acoustics, Vienna, Austria) and contact (throat) Triumph PC microphone (Clearer Communications, Inc, Burnaby, Canada) placed on the lamina of thyroid cartilage. Acoustic voice signal data were measured for fundamental frequency, percent of jitter and shimmer, normalized noise energy, signal-to-noise ratio, and harmonic-to-noise ratio using Dr. Speech software (Tiger Electronics, Seattle, WA). The correlations of acoustic voice parameters in vocal performance were statistically significant and strong (r = 0.71-1.0) for the entire functional measurements obtained for the two microphones. When classifying into healthy-pathological voice classes, the oral-shimmer revealed the correct classification rate (CCR) of 75.2% and the throat-jitter revealed CCR of 70.7%. However, combination of both throat and oral microphones allowed identifying a set of three voice parameters: throat-signal-to-noise ratio, oral-shimmer, and oral-normalized noise energy, which provided the CCR of 80.3%. The measurements of acoustic voice parameters using a combination of oral and throat microphones showed to be reliable in clinical settings and demonstrated high CCRs when distinguishing the healthy and pathological voice patient groups. Our study validates the suitability of the throat microphone signal for the task of automatic voice analysis for the purpose of voice screening. Copyright © 2015 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

Report of the Research Committee on Microphones and Microphone Placement. American School Band Directors' Association, Research Committee Reports for the 18th Annual Convention, Pittsburgh, Pennsylvania, 1970.

ERIC Educational Resources Information Center

American School Band Directors Association, Newark, OH.

The guide, one in a series of committee reports relating to school band performance, organization, and equipment needs, examines the relationship between microphones and tape recordings. The guide is presented in nine sections. Section I identifies types of microphones (carbon, crystal and ceramic, dynamic, condenser, and ribbon). Section II…

Wind noise in hearing aids with directional and omnidirectional microphones: polar characteristics of behind-the-ear hearing aids.

PubMed

Chung, King; Mongeau, Luc; McKibben, Nicholas

2009-04-01

Wind noise can be a significant problem for hearing instrument users. This study examined the polar characteristics of flow noise at outputs of two behind-the-ear digital hearing aids, and a microphone mounted on the surface of a cylinder at flow velocities ranging from a gentle breeze (4.5 m/s) to a strong gale (22.5 m/s) . The hearing aids were programed in an anechoic chamber, and tested in a quiet wind tunnel for flow noise recordings. Flow noise levels were estimated by normalizing the overall gain of the hearing aids to 0 dB. The results indicated that the two hearing aids had similar flow noise characteristics: The noise level was generally the lowest when the microphone faced upstream, higher when the microphone faced downstream, and the highest for frontal and rearward incidence angles. Directional microphones often generated higher flow noise level than omnidirectional microphones but they could reduce far-field background noise, resulting in a lower ambient noise level than omnidirectional microphones. Data for the academic microphone- on-cylinder configuration suggested that both turbulence and flow impingement might have contributed to the generation of flow noise in the hearing aids. Clinical and engineering design applications are discussed.

Hybrid method for determining the parameters of condenser microphones from measured membrane velocities and numerical calculations.

PubMed

Barrera-Figueroa, Salvador; Rasmussen, Knud; Jacobsen, Finn

2009-10-01

Typically, numerical calculations of the pressure, free-field, and random-incidence response of a condenser microphone are carried out on the basis of an assumed displacement distribution of the diaphragm of the microphone; the conventional assumption is that the displacement follows a Bessel function. This assumption is probably valid at frequencies below the resonance frequency. However, at higher frequencies the movement of the membrane is heavily coupled with the damping of the air film between membrane and backplate and with resonances in the back chamber of the microphone. A solution to this problem is to measure the velocity distribution of the membrane by means of a non-contact method, such as laser vibrometry. The measured velocity distribution can be used together with a numerical formulation such as the boundary element method for estimating the microphone response and other parameters, e.g., the acoustic center. In this work, such a hybrid method is presented and examined. The velocity distributions of a number of condenser microphones have been determined using a laser vibrometer, and these measured velocity distributions have been used for estimating microphone responses and other parameters. The agreement with experimental data is generally good. The method can be used as an alternative for validating the parameters of the microphones determined by classical calibration techniques.

Challenges and Recent Developments in Hearing Aids: Part I. Speech Understanding in Noise, Microphone Technologies and Noise Reduction Algorithms

PubMed Central

Chung, King

2004-01-01

This review discusses the challenges in hearing aid design and fitting and the recent developments in advanced signal processing technologies to meet these challenges. The first part of the review discusses the basic concepts and the building blocks of digital signal processing algorithms, namely, the signal detection and analysis unit, the decision rules, and the time constants involved in the execution of the decision. In addition, mechanisms and the differences in the implementation of various strategies used to reduce the negative effects of noise are discussed. These technologies include the microphone technologies that take advantage of the spatial differences between speech and noise and the noise reduction algorithms that take advantage of the spectral difference and temporal separation between speech and noise. The specific technologies discussed in this paper include first-order directional microphones, adaptive directional microphones, second-order directional microphones, microphone matching algorithms, array microphones, multichannel adaptive noise reduction algorithms, and synchrony detection noise reduction algorithms. Verification data for these technologies, if available, are also summarized. PMID:15678225

The Effect of Microphone Type on Acoustical Measures of Synthesized Vowels.

PubMed

Kisenwether, Jessica Sofranko; Sataloff, Robert T

2015-09-01

The purpose of this study was to compare microphones of different directionality, transducer type, and cost, with attention to their effects on acoustical measurements of period perturbation, amplitude perturbation, and noise using synthesized sustained vowel samples. This was a repeated measures design. Synthesized sustained vowel stimuli (with known acoustic characteristics and systematic changes in jitter, shimmer, and noise-to-harmonics ratio) were recorded by a variety of dynamic and condenser microphones. Files were then analyzed for mean fundamental frequency (fo), fo standard deviation, absolute jitter, shimmer in dB, peak-to-peak amplitude variation, and noise-to-harmonics ratio. Acoustical measures following recording were compared with the synthesized, known acoustical measures before recording. Although informal analyses showed some differences among microphones, and analyses of variance showed that type of microphone is a significant predictor, t-tests revealed that none of the microphones generated different means compared with the generated acoustical measures. In this sample, microphone type, directionality, and cost did not have a significant effect on the validity of acoustic measures. Copyright © 2015 The Voice Foundation. Published by Elsevier Inc. All rights reserved.

ZnO thin film piezoelectric micromachined microphone with symmetric composite vibrating diaphragm

NASA Astrophysics Data System (ADS)

Li, Junhong; Wang, Chenghao; Ren, Wei; Ma, Jun

2017-05-01

Residual stress is an important factor affecting the sensitivity of piezoelectric micromachined microphone. A symmetric composite vibrating diaphragm was adopted in the micro electro mechanical systems piezoelectric microphone to decrease the residual stress and improve the sensitivity of microphone in this paper. The ZnO film was selected as piezoelectric materials of microphone for its higher piezoelectric coefficient d 31 and lower relative dielectric constant. The thickness optimization of piezoelectric film on square diaphragm is difficult to be fulfilled by analytic method. To optimize the thickness of ZnO films, the stress distribution in ZnO film was analyzed by finite element method and the average stress in different thickness of ZnO films was given. The ZnO films deposited using dc magnetron sputtering exhibits a densely packed structure with columnar crystallites preferentially oriented along (002) plane. The diaphragm of microphone fabricated by micromachining techniques is flat and no wrinkling at corners, and the sensitivity of microphone is higher than 1 mV Pa-1. These results indicate the diaphragm has lower residual stress.

The effect of microphone wind noise on the amplitude modulation of wind turbine noise and its mitigation.

PubMed

Kendrick, Paul; von Hünerbein, Sabine; Cox, Trevor J

2016-07-01

Microphone wind noise can corrupt outdoor recordings even when wind shields are used. When monitoring wind turbine noise, microphone wind noise is almost inevitable because measurements cannot be made in still conditions. The effect of microphone wind noise on two amplitude modulation (AM) metrics is quantified in a simulation, showing that even at low wind speeds of 2.5 m/s errors of over 4 dBA can result. As microphone wind noise is intermittent, a wind noise detection algorithm is used to automatically find uncorrupted sections of the recording, and so recover the true AM metrics to within ±2/±0.5 dBA.

Fiber optic microphone with large dynamic range based on bi-fiber Fabry-Perot cavity

NASA Astrophysics Data System (ADS)

Cheng, Jin; Lu, Dan-feng; Gao, Ran; Qi, Zhi-mei

2017-10-01

In this paper, we report a fiber optic microphone with a large dynamic range. The probe of microphone consists of bi-fiber Fabry-Perot cavity architecture. The wavelength of the working laser is about 1552.05nm. At this wavelength, the interference spectroscopies of these two fiber Fabry-Perot cavities have a quadrature shift. So the outputs of these two fiber Fabry-Perot sensors are orthogonal signal. By using orthogonal signal demodulation method, this microphone can output a signal of acoustic wave. Due to no relationship between output signal and the linear region on interference spectroscopy, the microphones have a large maximum acoustic pressure above 125dB.

50 years of progress in microphone arrays for speech processing

NASA Astrophysics Data System (ADS)

Elko, Gary W.; Frisk, George V.

2004-10-01

In the early 1980s, Jim Flanagan had a dream of covering the walls of a room with microphones. He occasionally referred to this concept as acoustic wallpaper. Being a new graduate in the field of acoustics and signal processing, it was fortunate that Bell Labs was looking for someone to investigate this area of microphone arrays for telecommunication. The job interview was exciting, with all of the big names in speech signal processing and acoustics sitting in the audience, many of whom were the authors of books and articles that were seminal contributions to the fields of acoustics and signal processing. If there ever was an opportunity of a lifetime, this was it. Fortunately, some of the work had already begun, and Sessler and West had already laid the groundwork for directional electret microphones. This talk will describe some of the very early work done at Bell Labs on microphone arrays and reflect on some of the many systems, from large 400-element arrays, to small two-microphone arrays. These microphone array systems were built under Jim Flanagan's leadership in an attempt to realize his vision of seamless hands-free speech communication between people and the communication of people with machines.

Multi-criteria anomaly detection in urban noise sensor networks.

PubMed

Dauwe, Samuel; Oldoni, Damiano; De Baets, Bernard; Van Renterghem, Timothy; Botteldooren, Dick; Dhoedt, Bart

2014-01-01

The growing concern of citizens about the quality of their living environment and the emergence of low-cost microphones and data acquisition systems triggered the deployment of numerous noise monitoring networks spread over large geographical areas. Due to the local character of noise pollution in an urban environment, a dense measurement network is needed in order to accurately assess the spatial and temporal variations. The use of consumer grade microphones in this context appears to be very cost-efficient compared to the use of measurement microphones. However, the lower reliability of these sensing units requires a strong quality control of the measured data. To automatically validate sensor (microphone) data, prior to their use in further processing, a multi-criteria measurement quality assessment model for detecting anomalies such as microphone breakdowns, drifts and critical outliers was developed. Each of the criteria results in a quality score between 0 and 1. An ordered weighted average (OWA) operator combines these individual scores into a global quality score. The model is validated on datasets acquired from a real-world, extensive noise monitoring network consisting of more than 50 microphones. Over a period of more than a year, the proposed approach successfully detected several microphone faults and anomalies.

Speech understanding in noise with an eyeglass hearing aid: asymmetric fitting and the head shadow benefit of anterior microphones.

PubMed

Mens, Lucas H M

2011-01-01

To test speech understanding in noise using array microphones integrated in an eyeglass device and to test if microphones placed anteriorly at the temple provide better directivity than above the pinna. Sentences were presented from the front and uncorrelated noise from 45, 135, 225 and 315°. Fifteen hearing impaired participants with a significant speech discrimination loss were included, as well as 5 normal hearing listeners. The device (Varibel) improved speech understanding in noise compared to most conventional directional devices with a directional benefit of 5.3 dB in the asymmetric fit mode, which was not significantly different from the bilateral fully directional mode (6.3 dB). Anterior microphones outperformed microphones at a conventional position above the pinna by 2.6 dB. By integrating microphones in an eyeglass frame, a long array can be used resulting in a higher directionality index and improved speech understanding in noise. An asymmetric fit did not significantly reduce performance and can be considered to increase acceptance and environmental awareness. Directional microphones at the temple seemed to profit more from the head shadow than above the pinna, better suppressing noise from behind the listener.

Use of a Parabolic Microphone to Detect Hidden Subjects in Search and Rescue.

PubMed

Bowditch, Nathaniel L; Searing, Stanley K; Thomas, Jeffrey A; Thompson, Peggy K; Tubis, Jacqueline N; Bowditch, Sylvia P

2018-03-01

This study compares a parabolic microphone to unaided hearing in detecting and comprehending hidden callers at ranges of 322 to 2510 m. Eight subjects were placed 322 to 2510 m away from a central listening point. The subjects were concealed, and their calling volume was calibrated. In random order, subjects were asked to call the name of a state for 5 minutes. Listeners with parabolic microphones and others with unaided hearing recorded the direction of the call (detection) and name of the state (comprehension). The parabolic microphone was superior to unaided hearing in both detecting subjects and comprehending their calls, with an effect size (Cohen's d) of 1.58 for detection and 1.55 for comprehension. For each of the 8 hidden subjects, there were 24 detection attempts with the parabolic microphone and 54 to 60 attempts by unaided listeners. At the longer distances (1529-2510 m), the parabolic microphone was better at detecting callers (83% vs 51%; P<0.00001 by χ 2 ) and comprehension (57% vs 12%; P<0.00001). At the shorter distances (322-1190 m), the parabolic microphone offered advantages in detection (100% vs 83%; P=0.000023) and comprehension (86% vs 51%; P<0.00001), although not as pronounced as at the longer distances. Use of a 66-cm (26-inch) parabolic microphone significantly improved detection and comprehension of hidden calling subjects at distances between 322 and 2510 m when compared with unaided hearing. This study supports the use of a parabolic microphone in search and rescue to locate responsive subjects in favorable weather and terrain. Copyright © 2017 The Authors. Published by Elsevier Inc. All rights reserved.

Cochlear microphonic broad tuning curves

NASA Astrophysics Data System (ADS)

Ayat, Mohammad; Teal, Paul D.; Searchfield, Grant D.; Razali, Najwani

2015-12-01

It is known that the cochlear microphonic voltage exhibits much broader tuning than does the basilar membrane motion. The most commonly used explanation for this is that when an electrode is inserted at a particular point inside the scala media, the microphonic potentials of neighbouring hair cells have different phases, leading to cancelation at the electrodes location. In situ recording of functioning outer hair cells (OHCs) for investigating this hypothesis is exceptionally difficult. Therefore, to investigate the discrepancy between the tuning curves of the basilar membrane and those of the cochlear microphonic, and the effect of phase cancellation of adjacent hair cells on the broadness of the cochlear microphonic tuning curves, we use an electromechanical model of the cochlea to devise an experiment. We explore the effect of adjacent hair cells (i.e., longitudinal phase cancellation) on the broadness of the cochlear microphonic tuning curves in different locations. The results of the experiment indicate that active longitudinal coupling (i.e., coupling with active adjacent outer hair cells) only slightly changes the broadness of the CM tuning curves. The results also demonstrate that there is a π phase difference between the potentials produced by the hair bundle and the soma near the place associated with the characteristic frequency based on place-frequency maps (i.e., the best place). We suggest that the transversal phase cancellation (caused by the phase difference between the hair bundle and the soma) plays a far more important role than longitudinal phase cancellation in the broadness of the cochlear microphonic tuning curves. Moreover, by increasing the modelled longitudinal resistance resulting the cochlear microphonic curves exhibiting sharper tuning. The results of the simulations suggest that the passive network of the organ of Corti determines the phase difference between the hair bundle and soma, and hence determines the sharpness of the cochlear microphonic tuning curves.

High-temperature fiber-optic lever microphone

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J.; Cuomo, Frank W.; Nguyen, Trung D.; Rizzi, Stephen A.; Clevenson, Sherman A.

1995-01-01

The design and construction of a fiber-optic lever microphone, capable of operating continuously at temperatures up to 538 C (1000 F) are described. The design is based on the theoretical sensitivities of each of the microphone system components, namely, a cartridge containing a stretched membrane, an optical fiber probe, and an optoelectronic amplifier. Laboratory calibrations include the pistonphone sensitivity and harmonic distortion at ambient temperature, and frequency response, background noise, and optical power transmission at both ambient and elevated temperatures. A field test in the Thermal Acoustic Fatigue Apparatus at Langley Research Center, in which the microphone was subjected to overall sound-pressure levels in the range of 130-160 dB and at temperatures from ambient to 538 C, revealed good agreement with a standard probe microphone.

A directional microphone array for acoustic studies of wind tunnel models

NASA Technical Reports Server (NTRS)

Soderman, P. T.; Noble, S. C.

1974-01-01

An end-fire microphone array that utilizes a digital time delay system has been designed and evaluated for measuring noise in wind tunnels. The directional response of both a four- and eight-element linear array of microphones has enabled substantial rejection of background noise and reverberations in the NASA Ames 40- by 80-foot wind tunnel. In addition, it is estimated that four- and eight-element arrays reject 6 and 9 dB, respectively, of microphone wind noise, as compared with a conventional omnidirectional microphone with nose cone. Array response to two types of jet engine models in the wind tunnel is presented. Comparisons of array response to loudspeakers in the wind tunnel and in free field are made.

Background noise in piezoresistive, electret condenser, and ceramic microphones.

PubMed

Zuckerwar, Allan J; Kuhn, Theodore R; Serbyn, Roman M

2003-06-01

Background noise studies have been extended from air condenser microphones to piezoresistive, electret condenser, and ceramic microphones. Theoretical models of the respective noise sources within each microphone are developed and are used to derive analytical expressions for the noise power spectral density for each type. Several additional noise sources for the piezoresistive and electret microphones, beyond what had previously been considered, were applied to the models and were found to contribute significantly to the total noise power spectral density. Experimental background noise measurements were taken using an upgraded acoustic isolation vessel and data acquisition system, and the results were compared to the theoretically obtained expressions. The models were found to yield power spectral densities consistent with the experimental results. The measurements reveal that the 1/f noise coefficient is strongly correlated with the diaphragm damping resistance, irrespective of the detection technology, i.e., air condenser, piezoresistive, etc. This conclusion has profound implications upon the expected 1/f noise component of micromachined (MEMS) microphones.

Microphone directionality, pre-emphasis filter, and wind noise in cochlear implants.

PubMed

Chung, King; McKibben, Nicholas

2011-10-01

Wind noise can be a nuisance or a debilitating masker for cochlear implant users in outdoor environments. Previous studies indicated that wind noise at the microphone/hearing aid output had high levels of low-frequency energy and the amount of noise generated is related to the microphone directionality. Currently, cochlear implants only offer either directional microphones or omnidirectional microphones for users at-large. As all cochlear implants utilize pre-emphasis filters to reduce low-frequency energy before the signal is encoded, effective wind noise reduction algorithms for hearing aids might not be applicable for cochlear implants. The purposes of this study were to investigate the effect of microphone directionality on speech recognition and perceived sound quality of cochlear implant users in wind noise and to derive effective wind noise reduction strategies for cochlear implants. A repeated-measure design was used to examine the effects of spectral and temporal masking created by wind noise recorded through directional and omnidirectional microphones and the effects of pre-emphasis filters on cochlear implant performance. A digital hearing aid was programmed to have linear amplification and relatively flat in-situ frequency responses for the directional and omnidirectional modes. The hearing aid output was then recorded from 0 to 360° at flow velocities of 4.5 and 13.5 m/sec in a quiet wind tunnel. Sixteen postlingually deafened adult cochlear implant listeners who reported to be able to communicate on the phone with friends and family without text messages participated in the study. Cochlear implant users listened to speech in wind noise recorded at locations that the directional and omnidirectional microphones yielded the lowest noise levels. Cochlear implant listeners repeated the sentences and rated the sound quality of the testing materials. Spectral and temporal characteristics of flow noise, as well as speech and/or noise characteristics before and after the pre-emphasis filter, were analyzed. Correlation coefficients between speech recognition scores and crest factors of wind noise before and after pre-emphasis filtering were also calculated. Listeners obtained higher scores using the omnidirectional than the directional microphone mode at 13.5 m/sec, but they obtained similar speech recognition scores for the two microphone modes at 4.5 m/sec. Higher correlation coefficients were obtained between speech recognition scores and crest factors of wind noise after pre-emphasis filtering rather than before filtering. Cochlear implant users would benefit from both directional and omnidirectional microphones to reduce far-field background noise and near-field wind noise. Automatic microphone switching algorithms can be more effective if the incoming signal were analyzed after pre-emphasis filters for microphone switching decisions. American Academy of Audiology.

Speech recognition for bilaterally asymmetric and symmetric hearing aid microphone modes in simulated classroom environments.

PubMed

Ricketts, Todd A; Picou, Erin M

2013-09-01

This study aimed to evaluate the potential utility of asymmetrical and symmetrical directional hearing aid fittings for school-age children in simulated classroom environments. This study also aimed to evaluate speech recognition performance of children with normal hearing in the same listening environments. Two groups of school-age children 11 to 17 years of age participated in this study. Twenty participants had normal hearing, and 29 participants had sensorineural hearing loss. Participants with hearing loss were fitted with behind-the-ear hearing aids with clinically appropriate venting and were tested in 3 hearing aid configurations: bilateral omnidirectional, bilateral directional, and asymmetrical directional microphones. Speech recognition testing was completed in each microphone configuration in 3 environments: Talker-Front, Talker-Back, and Question-Answer situations. During testing, the location of the speech signal changed, but participants were always seated in a noisy, moderately reverberant classroom-like room. For all conditions, results revealed expected effects of directional microphones on speech recognition performance. When the signal of interest was in front of the listener, bilateral directional microphone was best, and when the signal of interest was behind the listener, bilateral omnidirectional microphone was best. Performance with asymmetric directional microphones was between the 2 symmetrical conditions. The magnitudes of directional benefits and decrements were not significantly correlated. In comparison with their peers with normal hearing, children with hearing loss performed similarly to their peers with normal hearing when fitted with directional microphones and the speech was from the front. In contrast, children with normal hearing still outperformed children with hearing loss if the speech originated from behind, even when the children were fitted with the optimal hearing aid microphone mode for the situation. Bilateral directional microphones can be effective in improving speech recognition performance for children in the classroom, as long as child is facing the talker of interest. Bilateral directional microphones, however, can impair performance if the signal originates from behind a listener. However, these data suggest that the magnitude of decrement is not predictable from an individual's benefit. The results re-emphasize the importance of appropriate switching between microphone modes so children can take full advantage of directional benefits without being hurt by directional decrements. An asymmetric fitting limits decrements, but does not lead to maximum speech recognition scores when compared with the optimal symmetrical fitting. Therefore, the asymmetric mode may not be the best option as a default fitting for children in a classroom environment. While directional microphones improve performance for children with hearing loss, their performance in most conditions continues to be impaired relative to their normal-hearing peers, particularly when the signals of interest originate from behind or from an unpredictable location.

The Effects of Linear Microphone Array Changes on Computed Sound Exposure Level Footprints

NASA Technical Reports Server (NTRS)

Mueller, Arnold W.; Wilson, Mark R.

1997-01-01

Airport land planning commissions often are faced with determining how much area around an airport is affected by the sound exposure levels (SELS) associated with helicopter operations. This paper presents a study of the effects changing the size and composition of a microphone array has on the computed SEL contour (ground footprint) areas used by such commissions. Descent flight acoustic data measured by a fifteen microphone array were reprocessed for five different combinations of microphones within this array. This resulted in data for six different arrays for which SEL contours were computed. The fifteen microphone array was defined as the 'baseline' array since it contained the greatest amount of data. The computations used a newly developed technique, the Acoustic Re-propagation Technique (ART), which uses parts of the NASA noise prediction program ROTONET. After the areas of the SEL contours were calculated the differences between the areas were determined. The area differences for the six arrays are presented that show a five and a three microphone array (with spacing typical of that required by the FAA FAR Part 36 noise certification procedure) compare well with the fifteen microphone array. All data were obtained from a database resulting from a joint project conducted by NASA and U.S. Army researchers at Langley and Ames Research Centers. A brief description of the joint project test design, microphone array set-up, and data reduction methodology associated with the database are discussed.

[Value of the study of cochlear microphonic recordings in deep and severe deafness].

PubMed

Moatti, L; Busquet, D; Cotin, G

1983-01-01

A study was conducted to assess the contribution of cochlear microphonic potential recordings during electrophysiologic audiometry examinations. Amplitude of microphonic recordings were correlated with the degree of deafness, its etiology, and the prosthetic prognosis in 38 electrocochleographic examinations. Preliminary results are analyzed.

A low-cost acoustic permeameter

NASA Astrophysics Data System (ADS)

Drake, Stephen A.; Selker, John S.; Higgins, Chad W.

2017-04-01

Intrinsic permeability is an important parameter that regulates air exchange through porous media such as snow. Standard methods of measuring snow permeability are inconvenient to perform outdoors, are fraught with sampling errors, and require specialized equipment, while bringing intact samples back to the laboratory is also challenging. To address these issues, we designed, built, and tested a low-cost acoustic permeameter that allows computation of volume-averaged intrinsic permeability for a homogenous medium. In this paper, we validate acoustically derived permeability of homogenous, reticulated foam samples by comparison with results derived using a standard flow-through permeameter. Acoustic permeameter elements were designed for use in snow, but the measurement methods are not snow-specific. The electronic components - consisting of a signal generator, amplifier, speaker, microphone, and oscilloscope - are inexpensive and easily obtainable. The system is suitable for outdoor use when it is not precipitating, but the electrical components require protection from the elements in inclement weather. The permeameter can be operated with a microphone either internally mounted or buried a known depth in the medium. The calibration method depends on choice of microphone positioning. For an externally located microphone, calibration was based on a low-frequency approximation applied at 500 Hz that provided an estimate of both intrinsic permeability and tortuosity. The low-frequency approximation that we used is valid up to 2 kHz, but we chose 500 Hz because data reproducibility was maximized at this frequency. For an internally mounted microphone, calibration was based on attenuation at 50 Hz and returned only intrinsic permeability. We found that 50 Hz corresponded to a wavelength that minimized resonance frequencies in the acoustic tube and was also within the response limitations of the microphone. We used reticulated foam of known permeability (ranging from 2 × 10-7 to 3 × 10-9 m2) and estimated tortuosity of 1.05 to validate both methods. For the externally mounted microphone the mean normalized standard deviation was 6 % for permeability and 2 % for tortuosity. The mean relative error from known measurements was 17 % for permeability and 2 % for tortuosity. For the internally mounted microphone the mean normalized standard deviation for permeability was 10 % and the relative error was also 10 %. Permeability determination for an externally mounted microphone is less sensitive to environmental noise than is the internally mounted microphone and is therefore the recommended method. The approximation using the internally mounted microphone was developed as an alternative for circumstances in which placing the microphone in the medium was not feasible. Environmental noise degrades precision of both methods and is recognizable as increased scatter for replicate data points.

The benefits of remote microphone technology for adults with cochlear implants.

PubMed

Fitzpatrick, Elizabeth M; Séguin, Christiane; Schramm, David R; Armstrong, Shelly; Chénier, Josée

2009-10-01

Cochlear implantation has become a standard practice for adults with severe to profound hearing loss who demonstrate limited benefit from hearing aids. Despite the substantial auditory benefits provided by cochlear implants, many adults experience difficulty understanding speech in noisy environments and in other challenging listening conditions such as television. Remote microphone technology may provide some benefit in these situations; however, little is known about whether these systems are effective in improving speech understanding in difficult acoustic environments for this population. This study was undertaken with adult cochlear implant recipients to assess the potential benefits of remote microphone technology. The objectives were to examine the measurable and perceived benefit of remote microphone devices during television viewing and to assess the benefits of a frequency-modulated system for speech understanding in noise. Fifteen adult unilateral cochlear implant users were fit with remote microphone devices in a clinical environment. The study used a combination of direct measurements and patient perceptions to assess speech understanding with and without remote microphone technology. The direct measures involved a within-subject repeated-measures design. Direct measures of patients' speech understanding during television viewing were collected using their cochlear implant alone and with their implant device coupled to an assistive listening device. Questionnaires were administered to document patients' perceptions of benefits during the television-listening tasks. Speech recognition tests of open-set sentences in noise with and without remote microphone technology were also administered. Participants showed improved speech understanding for television listening when using remote microphone devices coupled to their cochlear implant compared with a cochlear implant alone. This benefit was documented both when listening to news and talk show recordings. Questionnaire results also showed statistically significant differences between listening with a cochlear implant alone and listening with a remote microphone device. Participants judged that remote microphone technology provided them with better comprehension, more confidence, and greater ease of listening. Use of a frequency-modulated system coupled to a cochlear implant also showed significant improvement over a cochlear implant alone for open-set sentence recognition in +10 and +5 dB signal to noise ratios. Benefits were measured during remote microphone use in focused-listening situations in a clinical setting, for both television viewing and speech understanding in noise in the audiometric sound suite. The results suggest that adult cochlear implant users should be counseled regarding the potential for enhanced speech understanding in difficult listening environments through the use of remote microphone technology.

Microphone Phenomena Observed with EFL Students.

ERIC Educational Resources Information Center

Wilcox, Wilma B.

This study investigated changes in the speech patterns of Japanese college students in an intensive English language course when using a microphone, focusing in part on possible links to "karaoke" activities common in Japan, in which participants sing along with music using a microphone. The researcher first observed several karaoke…

Noise Attenuation Performance Assessment of the Joint Helmet Mounted Cueing System (JHMCS)

DTIC Science & Technology

2010-08-01

Flash Drive (CFD) memory (Figure 9) and Sound Professionals SP-TFB-2 Miniature Binaural Microphones with the Sound Professionals SP-SPSB-1 Slim-line...flight noise. Sound Professionals binaural microphones were placed to record both internal and external sounds. One microphone was attached to the

A high-temperature wideband pressure transducer

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J.

1975-01-01

Progress in the development of a pressure transducer for measurement of the pressure fluctuations in the high temperature environment of a jet exhaust is reported. A condenser microphone carrier system was adapted to meet the specifications. A theoretical analysis is presented which describes the operation of the condenser microphone in terms of geometry, materials, and other physical properties. The analysis was used as the basis for design of a prototype high temperature microphone. The feasibility of connecting the microphone to a converter over a high temperature cable operating as a half-wavelength transmission line was also examined.

Infrasonic Stethoscope for Monitoring Physiological Processes

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J. (Inventor); Shams, Qamar A. (Inventor); Dimarcantonio, Albert L. (Inventor)

2018-01-01

An infrasonic stethoscope for monitoring physiological processes of a patient includes a microphone capable of detecting acoustic signals in the audible frequency bandwidth and in the infrasonic bandwidth (0.03 to 1000 Hertz), a body coupler attached to the body at a first opening in the microphone, a flexible tube attached to the body at a second opening in the microphone, and an earpiece attached to the flexible tube. The body coupler is capable of engagement with a patient to transmit sounds from the person, to the microphone and then to the earpiece.

Infrasonic Stethoscope for Monitoring Physiological Processes

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J. (Inventor); Shams, Qamar A. (Inventor); Dimarcantonio, Albert L. (Inventor)

2016-01-01

An infrasonic stethoscope for monitoring physiological processes of a patient includes a microphone capable of detecting acoustic signals in the audible frequency bandwidth and in the infrasonic bandwidth (0.03 to 1000 Hertz), a body coupler attached to the body at a first opening in the microphone, a flexible tube attached to the body at a second opening in the microphone, and an earpiece attached to the flexible tube. The body coupler is capable of engagement with a patient to transmit sounds from the person, to the microphone and then to the earpiece.

Noise Robust Speech Recognition Applied to Voice-Driven Wheelchair

NASA Astrophysics Data System (ADS)

Sasou, Akira; Kojima, Hiroaki

2009-12-01

Conventional voice-driven wheelchairs usually employ headset microphones that are capable of achieving sufficient recognition accuracy, even in the presence of surrounding noise. However, such interfaces require users to wear sensors such as a headset microphone, which can be an impediment, especially for the hand disabled. Conversely, it is also well known that the speech recognition accuracy drastically degrades when the microphone is placed far from the user. In this paper, we develop a noise robust speech recognition system for a voice-driven wheelchair. This system can achieve almost the same recognition accuracy as the headset microphone without wearing sensors. We verified the effectiveness of our system in experiments in different environments, and confirmed that our system can achieve almost the same recognition accuracy as the headset microphone without wearing sensors.

Mach-Zehnder interferometry method for acoustic shock wave measurements in air and broadband calibration of microphones.

PubMed

Yuldashev, Petr; Karzova, Maria; Khokhlova, Vera; Ollivier, Sébastien; Blanc-Benon, Philippe

2015-06-01

A Mach-Zehnder interferometer is used to measure spherically diverging N-waves in homogeneous air. An electrical spark source is used to generate high-amplitude (1800 Pa at 15 cm from the source) and short duration (50 μs) N-waves. Pressure waveforms are reconstructed from optical phase signals using an Abel-type inversion. It is shown that the interferometric method allows one to reach 0.4 μs of time resolution, which is 6 times better than the time resolution of a 1/8-in. condenser microphone (2.5 μs). Numerical modeling is used to validate the waveform reconstruction method. The waveform reconstruction method provides an error of less than 2% with respect to amplitude in the given experimental conditions. Optical measurement is used as a reference to calibrate a 1/8-in. condenser microphone. The frequency response function of the microphone is obtained by comparing the spectra of the waveforms resulting from optical and acoustical measurements. The optically measured pressure waveforms filtered with the microphone frequency response are in good agreement with the microphone output voltage. Therefore, an optical measurement method based on the Mach-Zehnder interferometer is a reliable tool to accurately characterize evolution of weak shock waves in air and to calibrate broadband acoustical microphones.

A three-microphone acoustic reflection technique using transmitted acoustic waves in the airway.

PubMed

Fujimoto, Yuki; Huang, Jyongsu; Fukunaga, Toshiharu; Kato, Ryo; Higashino, Mari; Shinomiya, Shohei; Kitadate, Shoko; Takahara, Yutaka; Yamaya, Atsuyo; Saito, Masatoshi; Kobayashi, Makoto; Kojima, Koji; Oikawa, Taku; Nakagawa, Ken; Tsuchihara, Katsuma; Iguchi, Masaharu; Takahashi, Masakatsu; Mizuno, Shiro; Osanai, Kazuhiro; Toga, Hirohisa

2013-10-15

The acoustic reflection technique noninvasively measures airway cross-sectional area vs. distance functions and uses a wave tube with a constant cross-sectional area to separate incidental and reflected waves introduced into the mouth or nostril. The accuracy of estimated cross-sectional areas gets worse in the deeper distances due to the nature of marching algorithms, i.e., errors of the estimated areas in the closer distances accumulate to those in the further distances. Here we present a new technique of acoustic reflection from measuring transmitted acoustic waves in the airway with three microphones and without employing a wave tube. Using miniaturized microphones mounted on a catheter, we estimated reflection coefficients among the microphones and separated incidental and reflected waves. A model study showed that the estimated cross-sectional area vs. distance function was coincident with the conventional two-microphone method, and it did not change with altered cross-sectional areas at the microphone position, although the estimated cross-sectional areas are relative values to that at the microphone position. The pharyngeal cross-sectional areas including retropalatal and retroglossal regions and the closing site during sleep was visualized in patients with obstructive sleep apnea. The method can be applicable to larger or smaller bronchi to evaluate the airspace and function in these localized airways.

Factors influencing individual variation in perceptual directional microphone benefit.

PubMed

Keidser, Gitte; Dillon, Harvey; Convery, Elizabeth; Mejia, Jorge

2013-01-01

Large variations in perceptual directional microphone benefit, which far exceed the variation expected from physical performance measures of directional microphones, have been reported in the literature. The cause for the individual variation has not been systematically investigated. To determine the factors that are responsible for the individual variation in reported perceptual directional benefit. A correlational study. Physical performance measures of the directional microphones obtained after they had been fitted to individuals, cognitive abilities of individuals, and measurement errors were related to perceptual directional benefit scores. Fifty-nine hearing-impaired adults with varied degrees of hearing loss participated in the study. All participants were bilaterally fitted with a Motion behind-the-ear device (500 M, 501 SX, or 501 P) from Siemens according to the National Acoustic Laboratories' non-linear prescription, version two (NAL-NL2). Using the Bamford-Kowal-Bench (BKB) sentences, the perceptual directional benefit was obtained as the difference in speech reception threshold measured in babble noise (SRTn) with the devices in directional (fixed hypercardioid) and in omnidirectional mode. The SRTn measurements were repeated three times with each microphone mode. Physical performance measures of the directional microphone included the angle of the microphone ports to loudspeaker axis, the frequency range dominated by amplified sound, the in situ signal-to-noise ratio (SNR), and the in situ three-dimensional, articulation-index weighted directivity index (3D AI-DI). The cognitive tests included auditory selective attention, speed of processing, and working memory. Intraparticipant variation on the repeated SRTn's and the interparticipant variation on the average SRTn were used to determine the effect of measurement error. A multiple regression analysis was used to determine the effect of other factors. Measurement errors explained 52% of the variation in perceptual directional microphone benefit (95% confidence interval [CI]: 34-78%), while another 37% of variation was explained primarily by the physical performance of the directional microphones after they were fitted to individuals. The most contributing factor was the in situ 3D AI-DI measured across the low frequencies. Repeated SRTn measurements are needed to obtain a reliable indication of the perceptual directional benefit in an individual. Further, to obtain optimum benefit from directional microphones, the effectiveness of the microphones should be maximized across the low frequencies. American Academy of Audiology.

Arrays of Miniature Microphones for Aeroacoustic Testing

NASA Technical Reports Server (NTRS)

Shams, Qamar A.; Humphreys, William M.; Sealey, Bradley S.; Bartram, Scott M.; Zuckewar, Allan J.; Comeaux, Toby; Adams, James K.

2007-01-01

A phased-array system comprised of custom-made and commercially available microelectromechanical system (MEMS) silicon microphones and custom ancillary hardware has been developed for use in aeroacoustic testing in hard-walled and acoustically treated wind tunnels. Recent advances in the areas of multi-channel signal processing and beam forming have driven the construction of phased arrays containing ever-greater numbers of microphones. Traditional obstacles to this trend have been posed by (1) the high costs of conventional condenser microphones, associated cabling, and support electronics and (2) the difficulty of mounting conventional microphones in the precise locations required for high-density arrays. The present development overcomes these obstacles. One of the hallmarks of the new system is a series of fabricated platforms on which multiple microphones can be mounted. These mounting platforms, consisting of flexible polyimide circuit-board material (see left side of figure), include all the necessary microphone power and signal interconnects. A single bus line connects all microphones to a common power supply, while the signal lines terminate in one or more data buses on the sides of the circuit board. To minimize cross talk between array channels, ground lines are interposed as shields between all the data bus signal lines. The MEMS microphones are electrically connected to the boards via solder pads that are built into the printed wiring. These flexible circuit boards share many characteristics with their traditional rigid counterparts, but can be manufactured much thinner, as small as 0.1 millimeter, and much lighter with boards weighing as much as 75 percent less than traditional rigid ones. For a typical hard-walled wind-tunnel installation, the flexible printed-circuit board is bonded to the tunnel wall and covered with a face sheet that contains precise cutouts for the microphones. Once the face sheet is mounted, a smooth surface is established over the entire array due to the flush mounting of all microphones (see right side of figure). The face sheet is made from a continuous glass-woven-fabric base impregnated with an epoxy resin binder. This material offers a combination of high mechanical strength and low dielectric loss, making it suitable for withstanding the harsh test section environment present in many wind tunnels, while at the same time protecting the underlying polyimide board. Customized signal-conditioning hardware consisting of line drivers and antialiasing filters are coupled with the array. The line drivers are constructed using low-supply-current, high-gain-bandwidth operational amplifiers designed to transmit the microphone signals several dozen feet from the array to external acquisition hardware. The anti-alias filters consist of individual Chebyshev low-pass filters (one for each microphone channel) housed on small printed-circuit boards mounted on one or more motherboards. The mother/daughter board design results in a modular system, which is easy to debug and service and which enables the filter characteristics to be changed by swapping daughter boards with ones containing different filter parameters. The filter outputs are passed to commercially- available acquisition hardware to digitize and store the conditioned microphone signals. Wind-tunnel testing of the new MEMS microphone polyimide mounting system shows that the array performance is comparable to that of traditional arrays, but with significantly less cost of construction.

The Benefit of Remote Microphones Using Four Wireless Protocols.

PubMed

Rodemerk, Krishna S; Galster, Jason A

2015-09-01

Many studies have reported the speech recognition benefits of a personal remote microphone system when used by adult listeners with hearing loss. The advance of wireless technology has allowed for many wireless audio transmission protocols. Some of these protocols interface with commercially available hearing aids. As a result, commercial remote microphone systems use a variety of different protocols for wireless audio transmission. It is not known how these systems compare, with regard to adult speech recognition in noise. The primary goal of this investigation was to determine the speech recognition benefits of four different commercially available remote microphone systems, each with a different wireless audio transmission protocol. A repeated-measures design was used in this study. Sixteen adults, ages 52 to 81 yr, with mild to severe sensorineural hearing loss participated in this study. Participants were fit with three different sets of bilateral hearing aids and four commercially available remote microphone systems (FM, 900 MHz, 2.4 GHz, and Bluetooth(®) paired with near-field magnetic induction). Speech recognition scores were measured by an adaptive version of the Hearing in Noise Test (HINT). The participants were seated both 6 and 12' away from the talker loudspeaker. Participants repeated HINT sentences with and without hearing aids and with four commercially available remote microphone systems in both seated positions with and without contributions from the hearing aid or environmental microphone (24 total conditions). The HINT SNR-50, or the signal-to-noise ratio required for correct repetition of 50% of the sentences, was recorded for all conditions. A one-way repeated measures analysis of variance was used to determine statistical significance of microphone condition. The results of this study revealed that use of the remote microphone systems statistically improved speech recognition in noise relative to unaided and hearing aid-only conditions across all four wireless transmission protocols at 6 and 12' away from the talker. Participants showed a significant improvement in speech recognition in noise when comparing four remote microphone systems with different wireless transmission methods to hearing aids alone. American Academy of Audiology.

Assessment of a directional microphone array for hearing-impaired listeners.

PubMed

Soede, W; Bilsen, F A; Berkhout, A J

1993-08-01

Hearing-impaired listeners often have great difficulty understanding speech in surroundings with background noise or reverberation. Based on array techniques, two microphone prototypes (broadside and endfire) have been developed with strongly directional characteristics [Soede et al., "Development of a new directional hearing instrument based on array technology," J. Acoust. Soc. Am. 94, 785-798 (1993)]. Physical measurements show that the arrays attenuate reverberant sound by 6 dB (free-field) and can improve the signal-to-noise ratio by 7 dB in a diffuse noise field (measured with a KEMAR manikin). For the clinical assessment of these microphones an experimental setup was made in a sound-insulated listening room with one loudspeaker in front of the listener simulating the partner in a discussion and eight loudspeakers placed on the edges of a cube producing a diffuse background noise. The hearing-impaired subject wearing his own (familiar) hearing aid is placed in the center of the cube. The speech-reception threshold in noise for simple Dutch sentences was determined with a normal single omnidirectional microphone and with one of the microphone arrays. The results of monaural listening tests with hearing impaired subjects show that in comparison with an omnidirectional hearing-aid microphone the broadside and endfire microphone array gives a mean improvement of the speech reception threshold in noise of 7.0 dB (26 subjects) and 6.8 dB (27 subjects), respectively. Binaural listening with two endfire microphone arrays gives a binaural improvement which is comparable to the binaural improvement obtained by listening with two normal ears or two conventional hearing aids.

49 CFR 325.33 - Site characteristics; highway operations.

Code of Federal Regulations, 2010 CFR

2010-10-01

... includes a portion of, a traveled lane of a public highway. A microphone target point shall be established on the centerline of the traveled lane of the highway, and a microphone location point shall be... microphone target point and on a line that is perpendicular to the centerline of the traveled lane of the...

Directional Microphone Hearing Aids in School Environments: Working toward Optimization

ERIC Educational Resources Information Center

Ricketts, Todd A.; Picou, Erin M.; Galster, Jason

2017-01-01

Purpose: The hearing aid microphone setting (omnidirectional or directional) can be selected manually or automatically. This study examined the percentage of time the microphone setting selected using each method was judged to provide the best signalto-noise ratio (SNR) for the talkers of interest in school environments. Method: A total of 26…

A high-temperature wideband pressure transducer

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J.

1976-01-01

The problem of operating a condenser microphone as a terminal element of a half wavelength transmission line was dealt with; the environment in which the microphone operates necessitates a 25 foot separation from its supporting electronics. A theoretical analysis of the microphone-cable system, substantiated by laboratory tests, provided criteria to optimize system gain.

MEMS Based Acoustic Array

NASA Technical Reports Server (NTRS)

Sheplak, Mark (Inventor); Nishida, Toshikaza (Inventor); Humphreys, William M. (Inventor); Arnold, David P. (Inventor)

2006-01-01

Embodiments of the present invention described and shown in the specification aid drawings include a combination responsive to an acoustic wave that can be utilized as a dynamic pressure sensor. In one embodiment of the present invention, the combination has a substrate having a first surface and an opposite second surface, a microphone positioned on the first surface of the substrate and having an input and a first output and a second output, wherein the input receives a biased voltage, and the microphone generates an output signal responsive to the acoustic wave between the first output and the second output. The combination further has an amplifier positioned on the first surface of the substrate and having a first input and a second input and an output, wherein the first input of the amplifier is electrically coupled to the first output of the microphone and the second input of the amplifier is electrically coupled to the second output of the microphone for receiving the output sinual from the microphone. The amplifier is spaced from the microphone with a separation smaller than 0.5 mm.

Technique for measurement of characteristic impedance and propagation constant for porous materials

NASA Astrophysics Data System (ADS)

Jung, Ki Won; Atchley, Anthony A.

2005-09-01

Knowledge of acoustic properties such as characteristic impedance and complex propagation constant is useful to characterize the acoustic behaviors of porous materials. Song and Bolton's four-microphone method [J. Acoust. Soc. Am. 107, 1131-1152 (2000)] is one of the most widely employed techniques. In this method two microphones are used to determine the complex pressure amplitudes for each side of a sample. Muehleisen and Beamer [J. Acoust. Soc. Am. 117, 536-544 (2005)] improved upon a four-microphone method by interchanging microphones to reduce errors due to uncertainties in microphone response. In this paper, a multiple microphone technique is investigated to reconstruct the pressure field inside an impedance tube. Measurements of the acoustic properties of a material having square cross-section pores is used to check the validity of the technique. The values of characteristic impedance and complex propagation constant extracted from the reconstruction agree well with predicted values. Furthermore, this technique is used in investigating the acoustic properties of reticulated vitreous carbon (RVC) in the range of 250-1100 Hz.

Design of UAV-Embedded Microphone Array System for Sound Source Localization in Outdoor Environments †

PubMed Central

Hoshiba, Kotaro; Washizaki, Kai; Wakabayashi, Mizuho; Ishiki, Takahiro; Bando, Yoshiaki; Gabriel, Daniel; Nakadai, Kazuhiro; Okuno, Hiroshi G.

2017-01-01

In search and rescue activities, unmanned aerial vehicles (UAV) should exploit sound information to compensate for poor visual information. This paper describes the design and implementation of a UAV-embedded microphone array system for sound source localization in outdoor environments. Four critical development problems included water-resistance of the microphone array, efficiency in assembling, reliability of wireless communication, and sufficiency of visualization tools for operators. To solve these problems, we developed a spherical microphone array system (SMAS) consisting of a microphone array, a stable wireless network communication system, and intuitive visualization tools. The performance of SMAS was evaluated with simulated data and a demonstration in the field. Results confirmed that the SMAS provides highly accurate localization, water resistance, prompt assembly, stable wireless communication, and intuitive information for observers and operators. PMID:29099790

Design of UAV-Embedded Microphone Array System for Sound Source Localization in Outdoor Environments.

PubMed

Hoshiba, Kotaro; Washizaki, Kai; Wakabayashi, Mizuho; Ishiki, Takahiro; Kumon, Makoto; Bando, Yoshiaki; Gabriel, Daniel; Nakadai, Kazuhiro; Okuno, Hiroshi G

2017-11-03

In search and rescue activities, unmanned aerial vehicles (UAV) should exploit sound information to compensate for poor visual information. This paper describes the design and implementation of a UAV-embedded microphone array system for sound source localization in outdoor environments. Four critical development problems included water-resistance of the microphone array, efficiency in assembling, reliability of wireless communication, and sufficiency of visualization tools for operators. To solve these problems, we developed a spherical microphone array system (SMAS) consisting of a microphone array, a stable wireless network communication system, and intuitive visualization tools. The performance of SMAS was evaluated with simulated data and a demonstration in the field. Results confirmed that the SMAS provides highly accurate localization, water resistance, prompt assembly, stable wireless communication, and intuitive information for observers and operators.

Improving the accuracy of smart devices to measure noise exposure.

PubMed

Roberts, Benjamin; Kardous, Chucri; Neitzel, Richard

2016-11-01

Occupational noise exposure is one of the most frequent hazards present in the workplace; up to 22 million workers have potentially hazardous noise exposures in the U.S. As a result, noise-induced hearing loss is one of the most common occupational injuries in the U.S. Workers in manufacturing, construction, and the military are at the highest risk for hearing loss. Despite the large number of people exposed to high levels of noise at work, many occupations have not been adequately evaluated for noise exposure. The objective of this experiment was to investigate whether or not iOS smartphones and other smart devices (Apple iPhones and iPods) could be used as reliable instruments to measure noise exposures. For this experiment three different types of microphones were tested with a single model of iPod and three generations of iPhones: the internal microphones on the device, a low-end lapel microphone, and a high-end lapel microphone marketed as being compliant with the International Electrotechnical Commission's (IEC) standard for a Class 2-microphone. All possible combinations of microphones and noise measurement applications were tested in a controlled environment using several different levels of pink noise ranging from 60-100 dBA. Results were compared to simultaneous measurements made using a Type 1 sound level measurement system. Analysis of variance and Tukey's honest significant difference (HSD) test were used to determine if the results differed by microphone or noise measurement application. Levels measured with external microphones combined with certain noise measurement applications did not differ significantly from levels measured with the Type 1 sound measurement system. Results showed that it may be possible to use iOS smartphones and smart devices, with specific combinations of measurement applications and calibrated external microphones, to collect reliable, occupational noise exposure data under certain conditions and within the limitations of the device. Further research is needed to determine how these devices compare to traditional noise dosimeter under real-world conditions.

Speech perception comparisons using an implanted and an external microphone in existing cochlear implant users.

PubMed

Jenkins, Herman A; Uhler, Kristin

2012-01-01

To compare the speech understanding abilities of cochlear implant listeners using 2 microphone technologies, the Otologics fully implantable Carina and the Cochlear Freedom microphones. Feasibility study using direct comparison of the 2 microphones, nonrandomized and nonblinded within case studies. Tertiary referral center hospital outpatient clinic. Four subjects with greater than 1 year of unilateral listening experience with the Freedom Cochlear Implant and a CNC word score higher than 40%. A Carina microphone coupled to a percutaneous plug was implanted on the ipsilateral side of the cochlear implant. Two months were allowed for healing before connecting to the Carina microphone. The percutaneous plug was connected to a body worn external processor with output leads inserted into the auxiliary port of the Freedom processor. Subjects were instructed to use each of the 2 microphones for half of their daily implant use. Aided pure tone thresholds, consonant-nucleus-consonant (CNC), Bamford-Kowel-Bench Speech in Noise test (BKN-SIN), and Abbreviated Profile of Hearing Aid Benefit. All subjects had sound perceptions using both microphones. The loudness and quality of the sound was judged to be poorer with the Carina in the first 2 subjects. The latter 2 demonstrated essential equivalence in the second two listeners, with the exception of the Abbreviated Profile of Hearing Aid Benefit reporting greater percentage of problems for the Carina in the background noise situation for subject 0011-003PP. CNC word scores were better with the Freedom than the Carina in all 4 subjects. The latter 2 showed improved speech perception abilities with the Carina, compared with the first 2. The BKB-SIN showed consistently better results with the Freedom in noise. Early observations indicate that it is potentially feasible to use the fully implanted Carina microphone with the Freedom Cochlear Implant. The authors would anticipate that outcomes would improve as more knowledge is gained in signal processing and with the fabrication of an integrated device.

Sound source tracking device for telematic spatial sound field reproduction

NASA Astrophysics Data System (ADS)

Cardenas, Bruno

This research describes an algorithm that localizes sound sources for use in telematic applications. The localization algorithm is based on amplitude differences between various channels of a microphone array of directional shotgun microphones. The amplitude differences will be used to locate multiple performers and reproduce their voices, which were recorded at close distance with lavalier microphones, spatially corrected using a loudspeaker rendering system. In order to track multiple sound sources in parallel the information gained from the lavalier microphones will be utilized to estimate the signal-to-noise ratio between each performer and the concurrent performers.

Micromachined diffraction based optical microphones and intensity probes with electrostatic force feedback

NASA Astrophysics Data System (ADS)

Bicen, Baris

Measuring acoustic pressure gradients is critical in many applications such as directional microphones for hearing aids and sound intensity probes. This measurement is especially challenging with decreasing microphone size, which reduces the sensitivity due to small spacing between the pressure ports. Novel, micromachined biomimetic microphone diaphragms are shown to provide high sensitivity to pressure gradients on one side of the diaphragm with low thermal mechanical noise. These structures have a dominant mode shape with see-saw like motion in the audio band, responding to pressure gradients as well as spurious higher order modes sensitive to pressure. In this dissertation, integration of a diffraction based optical detection method with these novel diaphragm structures to implement a low noise optical pressure gradient microphone is described and experimental characterization results are presented, showing 36 dBA noise level with 1mm port spacing, nearly an order of magnitude better than the current gradient microphones. The optical detection scheme also provides electrostatic actuation capability from both sides of the diaphragm separately which can be used for active force feedback. A 4-port electromechanical equivalent circuit model of this microphone with optical readout is developed to predict the overall response of the device to different acoustic and electrostatic excitations. The model includes the damping due to complex motion of air around the microphone diaphragm, and it calculates the detected optical signal on each side of the diaphragm as a combination of two separate dominant vibration modes. This equivalent circuit model is verified by experiments and used to predict the microphone response with different force feedback schemes. Single sided force feedback is used for active damping to improve the linearity and the frequency response of the microphone. Furthermore, it is shown that using two sided force feedback one can significantly suppress or enhance the desired vibration modes of the diaphragm. This approach provides an electronic means to tailor the directional response of the microphones, with significant implications in device performance for various applications. As an example, the use of this device as a particle velocity sensor for sound intensity and sound power measurements is investigated. Without force feedback, the gradient microphone provides accurate particle velocity measurement for frequencies below 2 kHz, after which the pressure response of the second order mode becomes significant. With two-sided force feedback, the calculations show that this upper frequency limit may be increased to 10 kHz. This improves the pressure residual intensity index by more than 15 dB in the 50 Hz--10 kHz range, matching the Class I requirements of IEC 1043 standards for intensity probes without any need for multiple spacers.

The Effect of Microphone Placement on Interaural Level Differences and Sound Localization Across the Horizontal Plane in Bilateral Cochlear Implant Users.

PubMed

Jones, Heath G; Kan, Alan; Litovsky, Ruth Y

2016-01-01

This study examined the effect of microphone placement on the interaural level differences (ILDs) available to bilateral cochlear implant (BiCI) users, and the subsequent effects on horizontal-plane sound localization. Virtual acoustic stimuli for sound localization testing were created individually for eight BiCI users by making acoustic transfer function measurements for microphones placed in the ear (ITE), behind the ear (BTE), and on the shoulders (SHD). The ILDs across source locations were calculated for each placement to analyze their effect on sound localization performance. Sound localization was tested using a repeated-measures, within-participant design for the three microphone placements. The ITE microphone placement provided significantly larger ILDs compared to BTE and SHD placements, which correlated with overall localization errors. However, differences in localization errors across the microphone conditions were small. The BTE microphones worn by many BiCI users in everyday life do not capture the full range of acoustic ILDs available, and also reduce the change in cue magnitudes for sound sources across the horizontal plane. Acute testing with an ITE placement reduced sound localization errors along the horizontal plane compared to the other placements in some patients. Larger improvements may be observed if patients had more experience with the new ILD cues provided by an ITE placement.

75 FR 25254 - Notice of Public Information Collection(s) Being Reviewed by the Federal Communications...

Federal Register 2010, 2011, 2012, 2013, 2014

2010-05-07

..., including the use of automated collection techniques or other forms of information technology, and (e) ways... Stations (Including Wireless Microphones). Form No.: N/A. Type of Review: Revision of a currently approved... wireless microphones and provide them a home in the core TV spectrum, where many wireless microphones are...

Novel Methods for Sensing Acoustical Emissions From the Knee for Wearable Joint Health Assessment.

PubMed

Teague, Caitlin N; Hersek, Sinan; Toreyin, Hakan; Millard-Stafford, Mindy L; Jones, Michael L; Kogler, Geza F; Sawka, Michael N; Inan, Omer T

2016-08-01

We present the framework for wearable joint rehabilitation assessment following musculoskeletal injury. We propose a multimodal sensing (i.e., contact based and airborne measurement of joint acoustic emission) system for at-home monitoring. We used three types of microphones-electret, MEMS, and piezoelectric film microphones-to obtain joint sounds in healthy collegiate athletes during unloaded flexion/extension, and we evaluated the robustness of each microphone's measurements via: 1) signal quality and 2) within-day consistency. First, air microphones acquired higher quality signals than contact microphones (signal-to-noise-and-interference ratio of 11.7 and 12.4 dB for electret and MEMS, respectively, versus 8.4 dB for piezoelectric). Furthermore, air microphones measured similar acoustic signatures on the skin and 5 cm off the skin (∼4.5× smaller amplitude). Second, the main acoustic event during repetitive motions occurred at consistent joint angles (intra-class correlation coefficient ICC(1, 1) = 0.94 and ICC(1, k) = 0.99). Additionally, we found that this angular location was similar between right and left legs, with asymmetry observed in only a few individuals. We recommend using air microphones for wearable joint sound sensing; for practical implementation of contact microphones in a wearable device, interface noise must be reduced. Importantly, we show that airborne signals can be measured consistently and that healthy left and right knees often produce a similar pattern in acoustic emissions. These proposed methods have the potential for enabling knee joint acoustics measurement outside the clinic/lab and permitting long-term monitoring of knee health for patients rehabilitating an acute knee joint injury.

Exploring the feasibility of smart phone microphone for measurement of acoustic voice parameters and voice pathology screening.

PubMed

Uloza, Virgilijus; Padervinskis, Evaldas; Vegiene, Aurelija; Pribuisiene, Ruta; Saferis, Viktoras; Vaiciukynas, Evaldas; Gelzinis, Adas; Verikas, Antanas

2015-11-01

The objective of this study is to evaluate the reliability of acoustic voice parameters obtained using smart phone (SP) microphones and investigate the utility of use of SP voice recordings for voice screening. Voice samples of sustained vowel/a/obtained from 118 subjects (34 normal and 84 pathological voices) were recorded simultaneously through two microphones: oral AKG Perception 220 microphone and SP Samsung Galaxy Note3 microphone. Acoustic voice signal data were measured for fundamental frequency, jitter and shimmer, normalized noise energy (NNE), signal to noise ratio and harmonic to noise ratio using Dr. Speech software. Discriminant analysis-based Correct Classification Rate (CCR) and Random Forest Classifier (RFC) based Equal Error Rate (EER) were used to evaluate the feasibility of acoustic voice parameters classifying normal and pathological voice classes. Lithuanian version of Glottal Function Index (LT_GFI) questionnaire was utilized for self-assessment of the severity of voice disorder. The correlations of acoustic voice parameters obtained with two types of microphones were statistically significant and strong (r = 0.73-1.0) for the entire measurements. When classifying into normal/pathological voice classes, the Oral-NNE revealed the CCR of 73.7% and the pair of SP-NNE and SP-shimmer parameters revealed CCR of 79.5%. However, fusion of the results obtained from SP voice recordings and GFI data provided the CCR of 84.60% and RFC revealed the EER of 7.9%, respectively. In conclusion, measurements of acoustic voice parameters using SP microphone were shown to be reliable in clinical settings demonstrating high CCR and low EER when distinguishing normal and pathological voice classes, and validated the suitability of the SP microphone signal for the task of automatic voice analysis and screening.

System for determining aerodynamic imbalance

NASA Technical Reports Server (NTRS)

Churchill, Gary B. (Inventor); Cheung, Benny K. (Inventor)

1994-01-01

A system is provided for determining tracking error in a propeller or rotor driven aircraft by determining differences in the aerodynamic loading on the propeller or rotor blades of the aircraft. The system includes a microphone disposed relative to the blades during the rotation thereof so as to receive separate pressure pulses produced by each of the blades during the passage thereof by the microphone. A low pass filter filters the output signal produced by the microphone, the low pass filter having an upper cut-off frequency set below the frequency at which the blades pass by the microphone. A sensor produces an output signal after each complete revolution of the blades, and a recording display device displays the outputs of the low pass filter and sensor so as to enable evaluation of the relative magnitudes of the pressure pulses produced by passage of the blades by the microphone during each complete revolution of the blades.

Localization of sound sources in a room with one microphone

NASA Astrophysics Data System (ADS)

Peić Tukuljac, Helena; Lissek, Hervé; Vandergheynst, Pierre

2017-08-01

Estimation of the location of sound sources is usually done using microphone arrays. Such settings provide an environment where we know the difference between the received signals among different microphones in the terms of phase or attenuation, which enables localization of the sound sources. In our solution we exploit the properties of the room transfer function in order to localize a sound source inside a room with only one microphone. The shape of the room and the position of the microphone are assumed to be known. The design guidelines and limitations of the sensing matrix are given. Implementation is based on the sparsity in the terms of voxels in a room that are occupied by a source. What is especially interesting about our solution is that we provide localization of the sound sources not only in the horizontal plane, but in the terms of the 3D coordinates inside the room.

Optimization of a fiber optic flexible disk microphone

NASA Astrophysics Data System (ADS)

Zhang, Gang; Yu, Benli; Wang, Hui; Liu, Fei; Peng, Jun; Wu, Xuqiang

2011-11-01

An optimized design of a fiber optic flexible disk microphone is presented and verified experimentally. The phase sensitivity of optical fiber microphone (both the ideal model with a simply supported disk (SSD) and the model with a clamped disk (CLD)) is analyzed by utilizing theory of plates and shells. The results show that the microphones have an optimum length of the sensing arm when inner radius of the fiber coils, radius and Poisson's radio of the flexible disk have been determined. Under a typical condition depicted in this paper, an optimum phase sensitivity for SSD model of 27.72 rad/Pa (-91.14 dB re 1 rad/μPa) and an optimum phase sensitivity for CLD model of 3.18 rad/Pa (-109.95 dB re 1 rad/μPa), can be achieved in theory. Several sample microphones are fabricated and tested. The experimental results are basically consistent with the theoretical analysis.

In vivo evaluation of mastication noise reduction for dual channel implantable microphone.

PubMed

Woo, SeongTak; Jung, EuiSung; Lim, HyungGyu; Lee, Jang Woo; Seong, Ki Woong; Won, Chul Ho; Kim, Myoung Nam; Cho, Jin Ho; Lee, Jyung Hyun

2014-01-01

Input for fully implantable hearing devices (FIHDs) is provided by an implantable microphone under the skin of the temporal bone. However, the implanted microphone can be affected when the FIHDs user chews. In this paper, a dual implantable microphone was designed that can filter out the noise from mastication. For the in vivo experiment, a fabricated microphone was implanted in a rabbit. Pure-tone sounds of 1 kHz through a standard speaker were applied to the rabbit, which was given food simultaneously. To evaluate noise reduction, the measured signals were processed using a MATLAB program based adaptive filter. To verify the proposed method, the correlation coefficients and signal to-noise ratio before and after signal processing were calculated. By comparing the results, signal-to-noise ratio and correlation coefficients are enhanced by 6.07dB and 0.529 respectively.

47 CFR 15.216 - Disclosure requirements for wireless microphones and other low power auxiliary stations capable...

Code of Federal Regulations, 2010 CFR

2010-10-01

... microphones and other low power auxiliary stations capable of operating in the core TV bands. 15.216 Section... wireless microphones and other low power auxiliary stations capable of operating in the core TV bands. (a... capable of operating in the core TV bands (channels 2-51, excluding channel 37) is subject to the...

Carbon granule probe microphone for leak detection. [recovery boilers

NASA Technical Reports Server (NTRS)

Parthasarathy, S. P. (Inventor)

1985-01-01

A microphone which is not subject to corrosion is provided by employing carbon granules to sense sound waves. The granules are packed into a ceramic tube and no diaphragm is used. A pair of electrodes is located in the tube adjacent the carbon granules and are coupled to a sensing circuit. Sound waves cause pressure changes on the carbon granules which results in a change in resistance in the electrical path between the electrodes. This change in resistance is detected by the sensing circuit. The microphone is suitable for use as a leak detection probe in recovery boilers, where it provides reliable operation without corrosion problems associated with conventional microphones.

Sound-field measurement with moving microphones

PubMed Central

Katzberg, Fabrice; Mazur, Radoslaw; Maass, Marco; Koch, Philipp; Mertins, Alfred

2017-01-01

Closed-room scenarios are characterized by reverberation, which decreases the performance of applications such as hands-free teleconferencing and multichannel sound reproduction. However, exact knowledge of the sound field inside a volume of interest enables the compensation of room effects and allows for a performance improvement within a wide range of applications. The sampling of sound fields involves the measurement of spatially dependent room impulse responses, where the Nyquist-Shannon sampling theorem applies in the temporal and spatial domains. The spatial measurement often requires a huge number of sampling points and entails other difficulties, such as the need for exact calibration of a large number of microphones. In this paper, a method for measuring sound fields using moving microphones is presented. The number of microphones is customizable, allowing for a tradeoff between hardware effort and measurement time. The goal is to reconstruct room impulse responses on a regular grid from data acquired with microphones between grid positions, in general. For this, the sound field at equidistant positions is related to the measurements taken along the microphone trajectories via spatial interpolation. The benefits of using perfect sequences for excitation, a multigrid recovery, and the prospects for reconstruction by compressed sensing are presented. PMID:28599533

The effect of bone conduction microphone placement on intensity and spectrum of transmitted speech items.

PubMed

Tran, Phuong K; Letowski, Tomasz R; McBride, Maranda E

2013-06-01

Speech signals can be converted into electrical audio signals using either conventional air conduction (AC) microphone or a contact bone conduction (BC) microphone. The goal of this study was to investigate the effects of the location of a BC microphone on the intensity and frequency spectrum of the recorded speech. Twelve locations, 11 on the talker's head and 1 on the collar bone, were investigated. The speech sounds were three vowels (/u/, /a/, /i/) and two consonants (/m/, /∫/). The sounds were produced by 12 talkers. Each sound was recorded simultaneously with two BC microphones and an AC microphone. Analyzed spectral data showed that the BC recordings made at the forehead of the talker were the most similar to the AC recordings, whereas the collar bone recordings were most different. Comparison of the spectral data with speech intelligibility data collected in another study revealed a strong negative relationship between BC speech intelligibility and the degree of deviation of the BC speech spectrum from the AC spectrum. In addition, the head locations that resulted in the highest speech intelligibility were associated with the lowest output signals among all tested locations. Implications of these findings for BC communication are discussed.

Modeling high signal-to-noise ratio in a novel silicon MEMS microphone with comb readout

NASA Astrophysics Data System (ADS)

Manz, Johannes; Dehe, Alfons; Schrag, Gabriele

2017-05-01

Strong competition within the consumer market urges the companies to constantly improve the quality of their devices. For silicon microphones excellent sound quality is the key feature in this respect which means that improving the signal-to-noise ratio (SNR), being strongly correlated with the sound quality is a major task to fulfill the growing demands of the market. MEMS microphones with conventional capacitive readout suffer from noise caused by viscous damping losses arising from perforations in the backplate [1]. Therefore, we conceived a novel microphone design based on capacitive read-out via comb structures, which is supposed to show a reduction in fluidic damping compared to conventional MEMS microphones. In order to evaluate the potential of the proposed design, we developed a fully energy-coupled, modular system-level model taking into account the mechanical motion, the slide film damping between the comb fingers, the acoustic impact of the package and the capacitive read-out. All submodels are physically based scaling with all relevant design parameters. We carried out noise analyses and due to the modular and physics-based character of the model, were able to discriminate the noise contributions of different parts of the microphone. This enables us to identify design variants of this concept which exhibit a SNR of up to 73 dB (A). This is superior to conventional and at least comparable to high-performance variants of the current state-of-the art MEMS microphones [2].

Ultra-low-noise preamplifier for condenser microphones.

PubMed

Starecki, Tomasz

2010-12-01

The paper presents the design of a low-noise preamplifier dedicated for condenser measurement microphones used in high sensitivity applications, in which amplifier noise is the main factor limiting sensitivity of the measurements. In measurement microphone preamplifiers, the dominant source of noise at lower frequencies is the bias resistance of the input stage. In the presented solution, resistors were connected to the input stage by means of switches. The switches are opened during measurements, which disconnects the resistors from the input stage and results in noise reduction. Closing the switches allows for fast charging of the microphone capacitance. At low frequencies the noise of the designed preamplifier is a few times lower in comparison to similar, commercially available instruments.

A four-element end-fire microphone array for acoustic measurements in wind tunnels

NASA Technical Reports Server (NTRS)

Soderman, P. T.; Noble, S. C.

1974-01-01

A prototype four-element end-fire microphone array was designed and built for evaluation as a directional acoustic receiver for use in large wind tunnels. The microphone signals were digitized, time delayed, summed, and reconverted to analog form in such a way as to create a directional response with the main lobe along the array axis. The measured array directivity agrees with theoretical predictions confirming the circuit design of the electronic control module. The array with 0.15 m (0.5 ft) microphone spacing rejected reverberations and background noise in the Ames 40- by 80-foot wind tunnel by 5 to 12 db for frequencies above 400 Hz.

Free-field Calibration of the Pressure Sensitivity of Microphones at Frequencies up to 80 kHz

NASA Technical Reports Server (NTRS)

Herring, G. C.; Zuckerwar, Allan J.; Elbing, Brian R.

2006-01-01

A free-field (FF) substitution method for calibrating the pressure sensitivity of microphones at frequencies up to 80 kHz is demonstrated with both grazing and normal incidence geometries. The substitution-based method, as opposed to a simultaneous method, avoids problems associated with the non-uniformity of the sound field and, as applied here, uses a 1/2 -inch air-condenser pressure microphone as a known reference. Best results were obtained with a centrifugal fan, which is used as a random, broadband sound source. A broadband source minimizes reflection-related interferences that often plague FF measurements. Calibrations were performed on 1/4-inch FF air-condenser, electret, and micro-electromechanical systems (MEMS) microphones in an anechoic chamber. The accuracy of this FF method is estimated by comparing the pressure sensitivity of an air-condenser microphone, as derived from the FF measurement, with that of an electrostatic actuator calibration and is typically 0.3 dB (95% confidence), over the range 2-80 kHz.

Contribution of crosstalk to the uncertainty of electrostatic actuator calibrations.

PubMed

Shams, Qamar A; Soto, Hector L; Zuckerwar, Allan J

2009-09-01

Crosstalk in electrostatic actuator calibrations is defined as the ratio of the microphone response to the actuator excitation voltage at a given frequency with the actuator polarization voltage turned off to the response, at the excitation frequency, with the polarization voltage turned on. It consequently contributes to the uncertainty of electrostatic actuator calibrations. Two sources of crosstalk are analyzed: the first attributed to the stray capacitance between the actuator electrode and the microphone backplate, and the second to the ground resistance appearing as a common element in the actuator excitation and microphone input loops. Measurements conducted on 1/4, 1/2, and 1 in. air condenser microphones reveal that the crosstalk has no frequency dependence up to the membrane resonance frequency and that the level of crosstalk lies at about -60 dB for all three microphones-conclusions that are consistent with theory. The measurements support the stray capacitance model. The contribution of crosstalk to the measurement standard uncertainty of an electrostatic actuator calibration is therewith 0.01 dB.

Unvoiced Speech Recognition Using Tissue-Conductive Acoustic Sensor

NASA Astrophysics Data System (ADS)

Heracleous, Panikos; Kaino, Tomomi; Saruwatari, Hiroshi; Shikano, Kiyohiro

2006-12-01

We present the use of stethoscope and silicon NAM (nonaudible murmur) microphones in automatic speech recognition. NAM microphones are special acoustic sensors, which are attached behind the talker's ear and can capture not only normal (audible) speech, but also very quietly uttered speech (nonaudible murmur). As a result, NAM microphones can be applied in automatic speech recognition systems when privacy is desired in human-machine communication. Moreover, NAM microphones show robustness against noise and they might be used in special systems (speech recognition, speech transform, etc.) for sound-impaired people. Using adaptation techniques and a small amount of training data, we achieved for a 20 k dictation task a[InlineEquation not available: see fulltext.] word accuracy for nonaudible murmur recognition in a clean environment. In this paper, we also investigate nonaudible murmur recognition in noisy environments and the effect of the Lombard reflex on nonaudible murmur recognition. We also propose three methods to integrate audible speech and nonaudible murmur recognition using a stethoscope NAM microphone with very promising results.

Spatial acoustic radiation of respiratory sounds for sleep evaluation.

PubMed

Shabtai, Noam R; Zigel, Yaniv

2017-09-01

Body posture has an effect on sleeping quality and breathing disorders and therefore it is important to be recognized for the completion of the sleep evaluation process. Since humans have a directional acoustic radiation pattern, it is hypothesized that microphone arrays can be used to recognize different body postures, which is highly practical for sleep evaluation applications that already measure respiratory sounds using distant microphones. Furthermore, body posture may have an effect on distant microphone measurement; hence, the measurement can be compensated if the body posture is correctly recognized. A spherical harmonics decomposition approach to the spatial acoustic radiation is presented, assuming an array of eight microphones in a medium-sized audiology booth. The spatial sampling and reconstruction of the radiation pattern is discussed, and a final setup for the microphone array is recommended. A case study is shown using recorded segments of snoring and breathing sounds of three human subjects in three body postures in a silent but not anechoic audiology booth.

Radiation impedance of condenser microphones and their diffuse-field responses.

PubMed

Barrera-Figueroa, Salvador; Rasmussen, Knud; Jacobsen, Finn

2010-04-01

The relation between the diffuse-field response and the radiation impedance of a microphone has been investigated. Such a relation can be derived from classical theory. The practical measurement of the radiation impedance requires (a) measuring the volume velocity of the membrane of the microphone and (b) measuring the pressure on the membrane of the microphone. The first measurement is carried out by means of laser vibrometry. The second measurement cannot be implemented in practice. However, the pressure on the membrane can be calculated numerically by means of the boundary element method. In this way, a hybrid estimate of the radiation impedance is obtained. The resulting estimate of the diffuse-field response is compared with experimental estimates of the diffuse-field response determined using reciprocity and the random-incidence method. The different estimates are in good agreement at frequencies below the resonance frequency of the microphone. Although the method may not be of great practical utility, it provides a useful validation of the estimates obtained by other means.

An analytical-numerical method for determining the mechanical response of a condenser microphone

PubMed Central

Homentcovschi, Dorel; Miles, Ronald N.

2011-01-01

The paper is based on determining the reaction pressure on the diaphragm of a condenser microphone by integrating numerically the frequency domain Stokes system describing the velocity and the pressure in the air domain beneath the diaphragm. Afterwards, the membrane displacement can be obtained analytically or numerically. The method is general and can be applied to any geometry of the backplate holes, slits, and backchamber. As examples, the method is applied to the Bruel & Kjaer (B&K) 4134 1/2-inch microphone determining the mechanical sensitivity and the mechano-thermal noise for a domain of frequencies and also the displacement field of the membrane for two specified frequencies. These elements compare well with the measured values published in the literature. Also a new design, completely micromachined (including the backvolume) of the B&K micro-electro-mechanical systems (MEM) 1/4-inch measurement microphone is proposed. It is shown that its mechanical performances are very similar to those of the B&K MEMS measurement microphone. PMID:22225026

An analytical-numerical method for determining the mechanical response of a condenser microphone.

PubMed

Homentcovschi, Dorel; Miles, Ronald N

2011-12-01

The paper is based on determining the reaction pressure on the diaphragm of a condenser microphone by integrating numerically the frequency domain Stokes system describing the velocity and the pressure in the air domain beneath the diaphragm. Afterwards, the membrane displacement can be obtained analytically or numerically. The method is general and can be applied to any geometry of the backplate holes, slits, and backchamber. As examples, the method is applied to the Bruel & Kjaer (B&K) 4134 1/2-inch microphone determining the mechanical sensitivity and the mechano-thermal noise for a domain of frequencies and also the displacement field of the membrane for two specified frequencies. These elements compare well with the measured values published in the literature. Also a new design, completely micromachined (including the backvolume) of the B&K micro-electro-mechanical systems (MEM) 1/4-inch measurement microphone is proposed. It is shown that its mechanical performances are very similar to those of the B&K MEMS measurement microphone. © 2011 Acoustical Society of America

Adaptive Wiener filtering for improved acquisition of distortion product otoacoustic emissions.

PubMed

Ozdamar, O; Delgado, R E; Rahman, S; Lopez, C

1998-01-01

An innovative acoustic noise canceling method using adaptive Wiener filtering (AWF) was developed for improved acquisition of distortion product otoacoustic emissions (DPOAEs). The system used one microphone placed in the test ear for the primary signal. Noise reference signals were obtained from three different sources: (a) pre-stimulus response from the test ear microphone, (b) post-stimulus response from a microphone placed near the head of the subject and (c) post-stimulus response obtained from a microphone placed in the subject's nontest ear. In order to improve spectral estimation, block averaging of a different number of single sweep responses was used. DPOAE data were obtained from 11 ears of healthy newborns in a well-baby nursery of a hospital under typical noise conditions. Simultaneously obtained recordings from all three microphones were digitized, stored and processed off-line to evaluate the effects of AWF with respect to DPOAE detection and signal-to-noise ratio (SNR) improvement. Results show that compared to standard DPOAE processing, AWF improved signal detection and improved SNR.

A transmission-line model of back-cavity dynamics for in-plane pressure-differential microphones.

PubMed

Kim, Donghwan; Kuntzman, Michael L; Hall, Neal A

2014-11-01

Pressure-differential microphones inspired by the hearing mechanism of a special parasitoid fly have been described previously. The designs employ a beam structure that rotates about two pivots over an enclosed back volume. The back volume is only partially enclosed due to open slits around the perimeter of the beam. The open slits enable incoming sound waves to affect the pressure profile in the microphone's back volume. The goal of this work is to study the net moment applied to pressure-differential microphones by an incoming sound wave, which in-turn requires modeling the acoustic pressure distribution within the back volume. A lumped-element distributed transmission-line model of the back volume is introduced for this purpose. It is discovered that the net applied moment follows a low-pass filter behavior such that, at frequencies below a corner frequency depending on geometrical parameters of the design, the applied moment is unaffected by the open slits. This is in contrast to the high-pass filter behavior introduced by barometric pressure vents in conventional omnidirectional microphones. The model accurately predicts observed curvature in the frequency response of a prototype pressure-differential microphone 2 mm × 1 mm × 0.5 mm in size and employing piezoelectric readout.

Vacuum-isolation vessel and method for measurement of thermal noise in microphones

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J. (Inventor); Ngo, Kim Chi T. (Inventor)

1992-01-01

The vacuum isolation vessel and method in accordance with the present invention are used to accurately measure thermal noise in microphones. The apparatus and method could be used in a microphone calibration facility or any facility used for testing microphones. Thermal noise is measured to determine the minimum detectable sound pressure by the microphone. Conventional isolation apparatus and methods have been unable to provide an acoustically quiet and substantially vibration free environment for accurately measuring thermal noise. In the present invention, an isolation vessel assembly comprises a vacuum sealed outer vessel, a vacuum sealed inner vessel, and an interior suspension assembly coupled between the outer and inner vessels for suspending the inner vessel within the outer vessel. A noise measurement system records thermal noise data from the isolation vessel assembly. A vacuum system creates a vacuum between an internal surface of the outer vessel and an external surface of the inner vessel. The present invention thus provides an acoustically quiet environment due to the vacuum created between the inner and outer vessels and a substantially vibration free environment due to the suspension assembly suspending the inner vessel within the outer vessel. The thermal noise in the microphone, effectively isolated according to the invention, can be accurately measured.

Monitor, a vibrotactile aid for environmental perception: a field evaluation by four people with severe hearing and vision impairment.

PubMed

Ranjbar, Parivash; Stenström, Ingeborg

2013-01-01

Monitor is a portable vibrotactile aid to improve the ability of people with severe hearing impairment or deafblindness to detect, identify, and recognize the direction of sound-producing events. It transforms and adapts sounds to the frequency sensitivity range of the skin. The aid was evaluated in the field. Four females (44-54 years) with Usher Syndrome I (three with tunnel vision and one with only light perception) tested the aid at home and in traffic in three different field studies: without Monitor, with Monitor with an omnidirectional microphone, and with Monitor with a directional microphone. The tests were video-documented, and the two field studies with Monitor were initiated after five weeks of training. The detection scores with omnidirectional and directional microphones were 100% for three participants and above 57% for one, both in their home and traffic environments. In the home environment the identification scores with the omnidirectional microphone were 70%-97% and 58%-95% with the directional microphone. The corresponding values in traffic were 29%-100% and 65%-100%, respectively. Their direction perception was improved to some extent by both microphones. Monitor improved the ability of people with deafblindness to detect, identify, and recognize the direction of events producing sounds.

Recording high quality speech during tagged cine-MRI studies using a fiber optic microphone.

PubMed

NessAiver, Moriel S; Stone, Maureen; Parthasarathy, Vijay; Kahana, Yuvi; Paritsky, Alexander; Paritsky, Alex

2006-01-01

To investigate the feasibility of obtaining high quality speech recordings during cine imaging of tongue movement using a fiber optic microphone. A Complementary Spatial Modulation of Magnetization (C-SPAMM) tagged cine sequence triggered by an electrocardiogram (ECG) simulator was used to image a volunteer while speaking the syllable pairs /a/-/u/, /i/-/u/, and the words "golly" and "Tamil" in sync with the imaging sequence. A noise-canceling, optical microphone was fastened approximately 1-2 inches above the mouth of the volunteer. The microphone was attached via optical fiber to a laptop computer, where the speech was sampled at 44.1 kHz. A reference recording of gradient activity with no speech was subtracted from target recordings. Good quality speech was discernible above the background gradient sound using the fiber optic microphone without reference subtraction. The audio waveform of gradient activity was extremely stable and reproducible. Subtraction of the reference gradient recording further reduced gradient noise by roughly 21 dB, resulting in exceptionally high quality speech waveforms. It is possible to obtain high quality speech recordings using an optical microphone even during exceptionally loud cine imaging sequences. This opens up the possibility of more elaborate MRI studies of speech including spectral analysis of the speech signal in all types of MRI.

Performance Analysis of a Cost-Effective Electret Condenser Microphone Directional Array

NASA Technical Reports Server (NTRS)

Humphreys, William M., Jr.; Gerhold, Carl H.; Zuckerwar, Allan J.; Herring, Gregory C.; Bartram, Scott M.

2003-01-01

Microphone directional array technology continues to be a critical part of the overall instrumentation suite for experimental aeroacoustics. Unfortunately, high sensor cost remains one of the limiting factors in the construction of very high-density arrays (i.e., arrays containing several hundred channels or more) which could be used to implement advanced beamforming algorithms. In an effort to reduce the implementation cost of such arrays, the authors have undertaken a systematic performance analysis of a prototype 35-microphone array populated with commercial electret condenser microphones. An ensemble of microphones coupling commercially available electret cartridges with passive signal conditioning circuitry was fabricated for use with the Langley Large Aperture Directional Array (LADA). A performance analysis consisting of three phases was then performed: (1) characterize the acoustic response of the microphones via laboratory testing and calibration, (2) evaluate the beamforming capability of the electret-based LADA using a series of independently controlled point sources in an anechoic environment, and (3) demonstrate the utility of an electret-based directional array in a real-world application, in this case a cold flow jet operating at high subsonic velocities. The results of the investigation revealed a microphone frequency response suitable for directional array use over a range of 250 Hz - 40 kHz, a successful beamforming evaluation using the electret-populated LADA to measure simple point sources at frequencies up to 20 kHz, and a successful demonstration using the array to measure noise generated by the cold flow jet. This paper presents an overview of the tests conducted along with sample data obtained from those tests.

Wireless microphone communication system telephonics P/N 484D000-1

NASA Technical Reports Server (NTRS)

1980-01-01

The wireless microphone is a lightweight, portable, wireless voice communications device for use by the crew of the space shuttle orbiter. The wireless microphone allows the crew to have normal hands-free voice communication while they are performing various mission activities. The unit is designed to transmit at 455 or 500 kilohertz and employs narrow band FM modulation. Two orthogonally placed antennas are used to insure good reception at the receiver.

Dynamically Reconfigurable Microphone Arrays

DTIC Science & Technology

2011-05-01

from a number of different positions. In the second tests, the 2 wireless microphones were combined with a rigid binaural array on top of the b21r...Static + 2 Wireless Using only a standard computer sound card, a robot is limited to binaural inputs. Even when using wireless microphones, the audio...34 in HRI, Arlington, VA, 2007, pp. 113-120. [6] M. Heckmann, T. Rodemann, F. Joublin, C. Goerick, and B. Scholling, "Auditory Inspired Binaural

PVDF-Based Piezoelectric Microphone for Sound Detection Inside the Cochlea: Toward Totally Implantable Cochlear Implants.

PubMed

Park, Steve; Guan, Xiying; Kim, Youngwan; Creighton, Francis Pete X; Wei, Eric; Kymissis, Ioannis John; Nakajima, Hideko Heidi; Olson, Elizabeth S

2018-01-01

We report the fabrication and characterization of a prototype polyvinylidene fluoride polymer-based implantable microphone for detecting sound inside gerbil and human cochleae. With the current configuration and amplification, the signal-to-noise ratios were sufficiently high for normally occurring sound pressures and frequencies (ear canal pressures >50-60 dB SPL and 0.1-10 kHz), though 10 to 20 dB poorer than for some hearing aid microphones. These results demonstrate the feasibility of the prototype devices as implantable microphones for the development of totally implantable cochlear implants. For patients, this will improve sound reception by utilizing the outer ear and will improve the use of cochlear implants.

PVDF-Based Piezoelectric Microphone for Sound Detection Inside the Cochlea: Toward Totally Implantable Cochlear Implants

PubMed Central

Guan, Xiying; Kim, Youngwan; Creighton, Francis (Pete) X.; Wei, Eric; Kymissis, Ioannis(John); Nakajima, Hideko Heidi; Olson, Elizabeth S.

2018-01-01

We report the fabrication and characterization of a prototype polyvinylidene fluoride polymer-based implantable microphone for detecting sound inside gerbil and human cochleae. With the current configuration and amplification, the signal-to-noise ratios were sufficiently high for normally occurring sound pressures and frequencies (ear canal pressures >50–60 dB SPL and 0.1–10 kHz), though 10 to 20 dB poorer than for some hearing aid microphones. These results demonstrate the feasibility of the prototype devices as implantable microphones for the development of totally implantable cochlear implants. For patients, this will improve sound reception by utilizing the outer ear and will improve the use of cochlear implants. PMID:29732950

Characterization of condenser microphones under different environmental conditions for accurate speed of sound measurements with acoustic resonators.

PubMed

Guianvarc'h, Cécile; Gavioso, Roberto M; Benedetto, Giuliana; Pitre, Laurent; Bruneau, Michel

2009-07-01

Condenser microphones are more commonly used and have been extensively modeled and characterized in air at ambient temperature and static pressure. However, several applications of interest for metrology and physical acoustics require to use these transducers in significantly different environmental conditions. Particularly, the extremely accurate determination of the speed of sound in monoatomic gases, which is pursued for a determination of the Boltzmann constant k by an acoustic method, entails the use of condenser microphones mounted within a spherical cavity, over a wide range of static pressures, at the temperature of the triple point of water (273.16 K). To further increase the accuracy achievable in this application, the microphone frequency response and its acoustic input impedance need to be precisely determined over the same static pressure and temperature range. Few previous works examined the influence of static pressure, temperature, and gas composition on the microphone's sensitivity. In this work, the results of relative calibrations of 1/4 in. condenser microphones obtained using an electrostatic actuator technique are presented. The calibrations are performed in pure helium and argon gas at temperatures near 273 K and in the pressure range between 10 and 600 kPa. These experimental results are compared with the predictions of a realistic model available in the literature, finding a remarkable good agreement. The model provides an estimate of the acoustic impedance of 1/4 in. condenser microphones as a function of frequency and static pressure and is used to calculate the corresponding frequency perturbations induced on the normal modes of a spherical cavity when this is filled with helium or argon gas.

Uncovering Spatial Variation in Acoustic Environments Using Sound Mapping.

PubMed

Job, Jacob R; Myers, Kyle; Naghshineh, Koorosh; Gill, Sharon A

2016-01-01

Animals select and use habitats based on environmental features relevant to their ecology and behavior. For animals that use acoustic communication, the sound environment itself may be a critical feature, yet acoustic characteristics are not commonly measured when describing habitats and as a result, how habitats vary acoustically over space and time is poorly known. Such considerations are timely, given worldwide increases in anthropogenic noise combined with rapidly accumulating evidence that noise hampers the ability of animals to detect and interpret natural sounds. Here, we used microphone arrays to record the sound environment in three terrestrial habitats (forest, prairie, and urban) under ambient conditions and during experimental noise introductions. We mapped sound pressure levels (SPLs) over spatial scales relevant to diverse taxa to explore spatial variation in acoustic habitats and to evaluate the number of microphones needed within arrays to capture this variation under both ambient and noisy conditions. Even at small spatial scales and over relatively short time spans, SPLs varied considerably, especially in forest and urban habitats, suggesting that quantifying and mapping acoustic features could improve habitat descriptions. Subset maps based on input from 4, 8, 12 and 16 microphones differed slightly (< 2 dBA/pixel) from those based on full arrays of 24 microphones under ambient conditions across habitats. Map differences were more pronounced with noise introductions, particularly in forests; maps made from only 4-microphones differed more (> 4 dBA/pixel) from full maps than the remaining subset maps, but maps with input from eight microphones resulted in smaller differences. Thus, acoustic environments varied over small spatial scales and variation could be mapped with input from 4-8 microphones. Mapping sound in different environments will improve understanding of acoustic environments and allow us to explore the influence of spatial variation in sound on animal ecology and behavior.

The capture and recreation of 3D auditory scenes

NASA Astrophysics Data System (ADS)

Li, Zhiyun

The main goal of this research is to develop the theory and implement practical tools (in both software and hardware) for the capture and recreation of 3D auditory scenes. Our research is expected to have applications in virtual reality, telepresence, film, music, video games, auditory user interfaces, and sound-based surveillance. The first part of our research is concerned with sound capture via a spherical microphone array. The advantage of this array is that it can be steered into any 3D directions digitally with the same beampattern. We develop design methodologies to achieve flexible microphone layouts, optimal beampattern approximation and robustness constraint. We also design novel hemispherical and circular microphone array layouts for more spatially constrained auditory scenes. Using the captured audio, we then propose a unified and simple approach for recreating them by exploring the reciprocity principle that is satisfied between the two processes. Our approach makes the system easy to build, and practical. Using this approach, we can capture the 3D sound field by a spherical microphone array and recreate it using a spherical loudspeaker array, and ensure that the recreated sound field matches the recorded field up to a high order of spherical harmonics. For some regular or semi-regular microphone layouts, we design an efficient parallel implementation of the multi-directional spherical beamformer by using the rotational symmetries of the beampattern and of the spherical microphone array. This can be implemented in either software or hardware and easily adapted for other regular or semi-regular layouts of microphones. In addition, we extend this approach for headphone-based system. Design examples and simulation results are presented to verify our algorithms. Prototypes are built and tested in real-world auditory scenes.

Traversing Microphone Track Installed in NASA Lewis' Aero-Acoustic Propulsion Laboratory Dome

NASA Technical Reports Server (NTRS)

Bauman, Steven W.; Perusek, Gail P.

1999-01-01

The Aero-Acoustic Propulsion Laboratory is an acoustically treated, 65-ft-tall dome located at the NASA Lewis Research Center. Inside this laboratory is the Nozzle Acoustic Test Rig (NATR), which is used in support of Advanced Subsonics Technology (AST) and High Speed Research (HSR) to test engine exhaust nozzles for thrust and acoustic performance under simulated takeoff conditions. Acoustic measurements had been gathered by a far-field array of microphones located along the dome wall and 10-ft above the floor. Recently, it became desirable to collect acoustic data for engine certifications (as specified by the Federal Aviation Administration (FAA)) that would simulate the noise of an aircraft taking off as heard from an offset ground location. Since nozzles for the High-Speed Civil Transport have straight sides that cause their noise signature to vary radially, an additional plane of acoustic measurement was required. Desired was an arched array of 24 microphones, equally spaced from the nozzle and each other, in a 25 off-vertical plane. The various research requirements made this a challenging task. The microphones needed to be aimed at the nozzle accurately and held firmly in place during testing, but it was also essential that they be easily and routinely lowered to the floor for calibration and servicing. Once serviced, the microphones would have to be returned to their previous location near the ceiling. In addition, there could be no structure could between the microphones and the nozzle, and any structure near the microphones would have to be designed to minimize noise reflections. After many concepts were considered, a single arched truss structure was selected that would be permanently affixed to the dome ceiling and to one end of the dome floor.

Bruel and Kjaer 4944 Microphone Grid Frequency Response Function System Identification

NASA Technical Reports Server (NTRS)

Bennett, Reginald; Lee, Erik

2010-01-01

Br el & Kjaer (B&K) 4944B pressure field microphone was judiciously selected to measure acoustic environments, 400Hz 50kHz, in close proximity of the nozzle during multiple firings of solid propellant rocket motors. It is well known that protective grids can affect the frequency response of microphones. B&K recommends operation of the B&K 4944B without a protective grid when recording measurements above 10 to 15 kHz.

Accuracy of an acoustic location system for monitoring the position of duetting songbirds in tropical forest

PubMed Central

Mennill, Daniel J.; Burt, John M.; Fristrup, Kurt M.; Vehrencamp, Sandra L.

2008-01-01

A field test was conducted on the accuracy of an eight-microphone acoustic location system designed to triangulate the position of duetting rufous-and-white wrens (Thryothorus rufalbus) in Costa Rica’s humid evergreen forest. Eight microphones were set up in the breeding territories of twenty pairs of wrens, with an average inter-microphone distance of 75.2±2.6 m. The array of microphones was used to record antiphonal duets broadcast through stereo loudspeakers. The positions of the loudspeakers were then estimated by evaluating the delay with which the eight microphones recorded the broadcast sounds. Position estimates were compared to coordinates surveyed with a global-positioning system (GPS). The acoustic location system estimated the position of loudspeakers with an error of 2.82±0.26 m and calculated the distance between the “male” and “female” loudspeakers with an error of 2.12±0.42 m. Given the large range of distances between duetting birds, this relatively low level of error demonstrates that the acoustic location system is a useful tool for studying avian duets. Location error was influenced partly by the difficulties inherent in collecting high accuracy GPS coordinates of microphone positions underneath a lush tropical canopy, and partly by the complicating influence of irregular topography and thick vegetation on sound transmission. PMID:16708941

Issues concerning international comparison of free-field calibrations of acoustical standards

NASA Astrophysics Data System (ADS)

Nedzelnitsky, Victor

2002-11-01

Primary free-field calibrations of laboratory standard microphones by the reciprocity method establish these microphones as reference standard devices for calibrating working standard microphones, other measuring microphones, and practical instruments such as sound level meters and personal sound exposure meters (noise dosimeters). These primary, secondary, and other calibrations are indispensable to the support of regulatory requirements, standards, and product characterization and quality control procedures important for industry, commerce, health, and safety. International Electrotechnical Commission (IEC) Technical Committee 29 Electroacoustics produces international documentary standards, including standards for primary and secondary free-field calibration and measurement procedures and their critically important application to practical instruments. This paper addresses some issues concerning calibrations, standards activities, and the international key comparison of primary free-field calibrations of IEC-type LS2 laboratory standard microphones that is being planned by the Consultative Committee for Acoustics, Ultrasound, and Vibration (CCAUV) of the International Committee for Weights and Measures (CIPM). This comparison will include free-field calibrations by the reciprocity method at participating major national metrology laboratories throughout the world.

Real time aircraft fly-over noise discrimination

NASA Astrophysics Data System (ADS)

Genescà, M.; Romeu, J.; Pàmies, T.; Sánchez, A.

2009-06-01

A method for measuring aircraft noise time history with automatic elimination of simultaneous urban noise is presented in this paper. A 3 m-long 12-microphone sparse array has been proven to give good performance in a wide range of urban placements. Nowadays, urban placements have to be avoided because their background noise has a great influence on the measurements made by sound level meters or single microphones. Because of the small device size and low number of microphones (that make it so easy to set up), the resolution of the device is not high enough to provide a clean aircraft noise time history by only applying frequency domain beamforming to the spatial cross-correlations of the microphones' signals. Therefore, a new step to the processing algorithm has been added to eliminate this handicap.

Extreme Low Frequency Acoustic Measurement System

NASA Technical Reports Server (NTRS)

Shams, Qamar A. (Inventor); Zuckerwar, Allan J. (Inventor)

2017-01-01

The present invention is an extremely low frequency (ELF) microphone and acoustic measurement system capable of infrasound detection in a portable and easily deployable form factor. In one embodiment of the invention, an extremely low frequency electret microphone comprises a membrane, a backplate, and a backchamber. The backchamber is sealed to allow substantially no air exchange between the backchamber and outside the microphone. Compliance of the membrane may be less than ambient air compliance. The backplate may define a plurality of holes and a slot may be defined between an outer diameter of the backplate and an inner wall of the microphone. The locations and sizes of the holes, the size of the slot, and the volume of the backchamber may be selected such that membrane motion is substantially critically damped.

Lumped-parameters equivalent circuit for condenser microphones modeling.

PubMed

Esteves, Josué; Rufer, Libor; Ekeom, Didace; Basrour, Skandar

2017-10-01

This work presents a lumped parameters equivalent model of condenser microphone based on analogies between acoustic, mechanical, fluidic, and electrical domains. Parameters of the model were determined mainly through analytical relations and/or finite element method (FEM) simulations. Special attention was paid to the air gap modeling and to the use of proper boundary condition. Corresponding lumped-parameters were obtained as results of FEM simulations. Because of its simplicity, the model allows a fast simulation and is readily usable for microphone design. This work shows the validation of the equivalent circuit on three real cases of capacitive microphones, including both traditional and Micro-Electro-Mechanical Systems structures. In all cases, it has been demonstrated that the sensitivity and other related data obtained from the equivalent circuit are in very good agreement with available measurement data.

Extreme low frequency acoustic measurement system

NASA Technical Reports Server (NTRS)

Shams, Qamar A. (Inventor); Zuckerwar, Allan J. (Inventor)

2013-01-01

The present invention is an extremely low frequency (ELF) microphone and acoustic measurement system capable of infrasound detection in a portable and easily deployable form factor. In one embodiment of the invention, an extremely low frequency electret microphone comprises a membrane, a backplate, and a backchamber. The backchamber is sealed to allow substantially no air exchange between the backchamber and outside the microphone. Compliance of the membrane may be less than ambient air compliance. The backplate may define a plurality of holes and a slot may be defined between an outer diameter of the backplate and an inner wall of the microphone. The locations and sizes of the holes, the size of the slot, and the volume of the backchamber may be selected such that membrane motion is substantially critically damped.

Wake Vortex Detection: Phased Microphone vs. Linear Infrasonic Array

NASA Technical Reports Server (NTRS)

Shams, Qamar A.; Zuckerwar, Allan J.; Sullivan, Nicholas T.; Knight, Howard K.

2014-01-01

Sensor technologies can make a significant impact on the detection of aircraft-generated vortices in an air space of interest, typically in the approach or departure corridor. Current state-of-the art sensor technologies do not provide three-dimensional measurements needed for an operational system or even for wake vortex modeling to advance the understanding of vortex behavior. Most wake vortex sensor systems used today have been developed only for research applications and lack the reliability needed for continuous operation. The main challenges for the development of an operational sensor system are reliability, all-weather operation, and spatial coverage. Such a sensor has been sought for a period of last forty years. Acoustic sensors were first proposed and tested by National Oceanic and Atmospheric Administration (NOAA) early in 1970s for tracking wake vortices but these acoustic sensors suffered from high levels of ambient noise. Over a period of the last fifteen years, there has been renewed interest in studying noise generated by aircraft wake vortices, both numerically and experimentally. The German Aerospace Center (DLR) was the first to propose the application of a phased microphone array for the investigation of the noise sources of wake vortices. The concept was first demonstrated at Berlins Airport Schoenefeld in 2000. A second test was conducted in Tarbes, France, in 2002, where phased microphone arrays were applied to study the wake vortex noise of an Airbus 340. Similarly, microphone phased arrays and other opto-acoustic microphones were evaluated in a field test at the Denver International Airport in 2003. For the Tarbes and Denver tests, the wake trajectories of phased microphone arrays and lidar were compared as these were installed side by side. Due to a built-in pressure equalization vent these microphones were not suitable for capturing acoustic noise below 20 Hz. Our group at NASA Langley Research Center developed and installed an infrasonic array at the Newport News-Williamsburg International Airport early in the year 2013. A pattern of pressure burst, high-coherence intervals, and diminishing-coherence intervals was observed for all takeoff and landing events without exception. The results of a phased microphone vs. linear infrasonic array comparison will be presented.

High temperature fiber optic microphone having a pressure-sensing reflective membrane under tensile stress

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J. (Inventor); Cuomo, Frank W. (Inventor); Robbins, William E. (Inventor); Hopson, Purnell, Jr. (Inventor)

1992-01-01

A fiber optic microphone is provided for measuring fluctuating pressures. An optical fiber probe having at least one transmitting fiber for transmitting light to a pressure-sensing membrane and at least one receiving fiber for receiving light reflected from a stretched membrane is provided. The pressure-sensing membrane may be stretched for high frequency response. Further, a reflecting surface of the pressure-sensing membrane may have dimensions which substantially correspond to dimensions of a cross section of the optical fiber probe. Further, the fiber optic microphone can be made of materials for use in high temperature environments, for example greater than 1000 F. A fiber optic probe is also provided with a backplate for damping membrane motion. The backplate further provides a means for on-line calibration of the microphone.

Fiber optic microphone having a pressure sensing reflective membrane and a voltage source for calibration purpose

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J. (Inventor); Cuomo, Frank W. (Inventor); Robbins, William E. (Inventor)

1993-01-01

A fiber optic microphone is provided for measuring fluctuating pressures. An optical fiber probe having at least one transmitting fiber for transmitting light to a pressure-sensing membrane and at least one receiving fiber for receiving light reflected from a stretched membrane is provided. The pressure-sensing membrane may be stretched for high frequency response. Further, a reflecting surface of the pressure-sensing membrane may have dimensions which substantially correspond to dimensions of a cross section of the optical fiber probe. Further, the fiber optic microphone can be made of materials for use in high temperature environments, for example greater than 1000 F. A fiber optic probe is also provided with a back plate for damping membrane motion. The back plate further provides a means for on-line calibration of the microphone.

A Background Noise Reduction Technique Using Adaptive Noise Cancellation for Microphone Arrays

NASA Technical Reports Server (NTRS)

Spalt, Taylor B.; Fuller, Christopher R.; Brooks, Thomas F.; Humphreys, William M., Jr.; Brooks, Thomas F.

2011-01-01

Background noise in wind tunnel environments poses a challenge to acoustic measurements due to possible low or negative Signal to Noise Ratios (SNRs) present in the testing environment. This paper overviews the application of time domain Adaptive Noise Cancellation (ANC) to microphone array signals with an intended application of background noise reduction in wind tunnels. An experiment was conducted to simulate background noise from a wind tunnel circuit measured by an out-of-flow microphone array in the tunnel test section. A reference microphone was used to acquire a background noise signal which interfered with the desired primary noise source signal at the array. The technique s efficacy was investigated using frequency spectra from the array microphones, array beamforming of the point source region, and subsequent deconvolution using the Deconvolution Approach for the Mapping of Acoustic Sources (DAMAS) algorithm. Comparisons were made with the conventional techniques for improving SNR of spectral and Cross-Spectral Matrix subtraction. The method was seen to recover the primary signal level in SNRs as low as -29 dB and outperform the conventional methods. A second processing approach using the center array microphone as the noise reference was investigated for more general applicability of the ANC technique. It outperformed the conventional methods at the -29 dB SNR but yielded less accurate results when coherence over the array dropped. This approach could possibly improve conventional testing methodology but must be investigated further under more realistic testing conditions.

The Measurement of Unsteady Surface Pressure Using a Remote Microphone Probe.

PubMed

Guan, Yaoyi; Berntsen, Carl R; Bilka, Michael J; Morris, Scott C

2016-12-03

Microphones are widely applied to measure pressure fluctuations at the walls of solid bodies immersed in turbulent flows. Turbulent motions with various characteristic length scales can result in pressure fluctuations over a wide frequency range. This property of turbulence requires sensing devices to have sufficient sensitivity over a wide range of frequencies. Furthermore, the small characteristic length scales of turbulent structures require small sensing areas and the ability to place the sensors in very close proximity to each other. The complex geometries of the solid bodies, often including large surface curvatures or discontinuities, require the probe to have the ability to be set up in very limited spaces. The development of a remote microphone probe, which is inexpensive, consistent, and repeatable, is described in the present communication. It allows for the measurement of pressure fluctuations with high spatial resolution and dynamic response over a wide range of frequencies. The probe is small enough to be placed within the interior of typical wind tunnel models. The remote microphone probe includes a small, rigid, and hollow tube that penetrates the model surface to form the sensing area. This tube is connected to a standard microphone, at some distance away from the surface, using a "T" junction. An experimental method is introduced to determine the dynamic response of the remote microphone probe. In addition, an analytical method for determining the dynamic response is described. The analytical method can be applied in the design stage to determine the dimensions and properties of the RMP components.

An experimental investigation of flow-induced oscillations of the Bruel and Kjaer in-flow microphone

NASA Technical Reports Server (NTRS)

Fields, Richard S., Jr.

1995-01-01

One source contributing to wind tunnel background noise is microphone self-noise. An experiment was conducted to investigate the flow-induced acoustic oscillations of Bruel & Kjaer (B&K) in-flow microphones. The results strongly suggest the B&K microphone cavity behaves more like an open cavity. Their cavity acoustic oscillations are likely caused by strong interactions between the cavity shear layer and the cavity trailing edge. But the results also suggest that cavity shear layer oscillations could be coupled with cavity acoustic resonance to generate tones. Detailed flow velocity measurements over the cavity screen have shown inflection points in the mean velocity profiles and high disturbance and spectral intensities in the vicinity of the cavity trailing edge. These results are the evidence for strong interactions between cavity shear layer oscillations and the cavity trailing edge. They also suggest that beside acoustic signals, the microphone inside the cavity has likely recorded hydrodynamic pressure oscillations, too. The results also suggest that the forebody shape does not have a direct effect on cavity oscillations. For the FITE (Flow Induced Tone Eliminator) microphone, it is probably the forebody length and the resulting boundary layer turbulence that have made it work. Turbulence might have thickened the boundary layer at the separation point, weakened the shear layer vortices, or lifted them to miss impinging on the cavity trailing edge. In addition, the study shows that the cavity screen can modulate the oscillation frequency but not the cavity acoustic oscillation mechanisms.

The Semicircular Canal Microphonic

NASA Technical Reports Server (NTRS)

Rabbitt, R. D.; Boyle, R.; Highstein, S. M.; Dalton, Bonnie P. (Technical Monitor)

2002-01-01

Present experiments were designed to quantify the alternating current (AC) component of the semicircular canal microphonic for angular motion stimulation as a function of stimulus frequency and amplitude. The oyster toadfish, Opsanus tau, was used as the experimental model. Calibrated mechanical indentation of the horizontal canal duct was used as a stimulus to generate hair-cell and afferent responses reproducing those present during head rotation. Sensitivity to polarization of the endolymph DC voltage re: perilymph was also investigated. Modulation of endolymph voltage was recorded using conventional glass electrodes and lock-in amplification over the frequency range 0.2-80 Hz. Access to the endolymph for inserting voltage recording and current passing electrodes was obtained by sectioning the anterior canal at its apex and isolating the cut ends in air. For sinusoidal stimulation below approx.10 Hz, the horizontal semicircular canal AC microphonic was nearly independent of stimulus frequency and equal to approximately 4 microV per micron indent (equivalent to approx. 1 microV per deg/s). A saturating nonlinearity decreasing the microphonic gain was present for stimuli exceeding approx.3 micron indent (approx. 12 deg/s angular velocity). The phase was not sensitive to the saturating nonlinearity. The microphonic exhibited a resonance near 30Hz consistent with basolateral current hair cell resonance observed previously in voltage-clamp records from semicircular canal hair cells. The magnitude and phase of the microphonic exhibited sensitivity to endolymphatic polarization consistent with electro-chemical reversal of hair cell transduction currents.

Separation of Main and Tail Rotor Noise Sources from Ground-Based Acoustic Measurements Using Time-Domain De-Dopplerization

NASA Technical Reports Server (NTRS)

Greenwood, Eric II; Schmitz, Fredric H.

2009-01-01

A new method of separating the contributions of helicopter main and tail rotor noise sources is presented, making use of ground-based acoustic measurements. The method employs time-domain de-Dopplerization to transform the acoustic pressure time-history data collected from an array of ground-based microphones to the equivalent time-history signals observed by an array of virtual inflight microphones traveling with the helicopter. The now-stationary signals observed by the virtual microphones are then periodically averaged with the main and tail rotor once per revolution triggers. The averaging process suppresses noise which is not periodic with the respective rotor, allowing for the separation of main and tail rotor pressure time-histories. The averaged measurements are then interpolated across the range of directivity angles captured by the microphone array in order to generate separate acoustic hemispheres for the main and tail rotor noise sources. The new method is successfully applied to ground-based microphone measurements of a Bell 206B3 helicopter and demonstrates the strong directivity characteristics of harmonic noise radiation from both the main and tail rotors of that helicopter.

Assessment of ground effects on the propagation of aircraft noise: The T-38A flight experiment

NASA Technical Reports Server (NTRS)

Willshire, W. L., Jr.

1980-01-01

A flight experiment was conducted to investigate air to ground propagation of sound at gazing angles of incidence. A turbojet powered airplane was flown at altitudes ranging from 10 to 160 m over a 20-microphone array positioned over grass and concrete. The dependence of ground effects on frequency, incidence angle, and slant range was determined using two analysis methods. In one method, a microphone close to the flight path is compared to down range microphones. In the other method, comparisons are made between two microphones which were equidistant from the flight path but positioned over the two surfaces. In both methods, source directivity angle was the criterion by which portions of the microphone signals were compared. The ground effects were largest in the frequency range of 200 to 400 Hz and were found to be dependent on incidence angle and slant range. Ground effects measured for angles of incidence greater than 10 deg to 15 deg were near zero. Measured attenuation increased with increasing slant range for slant ranges less than 750 m. Theoretical predictions were found to be in good agreement with the major details of the measured results.

Calibration of the pressure sensitivity of microphones by a free-field method at frequencies up to 80 khz.

PubMed

Zuckerwar, Allan J; Herring, G C; Elbing, Brian R

2006-01-01

A free-field (FF) substitution method for calibrating the pressure sensitivity of microphones at frequencies up to 80 kHz is demonstrated with both grazing and normal-incidence geometries. The substitution-based method, as opposed to a simultaneous method, avoids problems associated with the nonuniformity of the sound field and, as applied here, uses a 1/4-in. air-condenser pressure microphone as a known reference. Best results were obtained with a centrifugal fan, which is used as a random, broadband sound source. A broadband source minimizes reflection-related interferences that can plague FF measurements. Calibrations were performed on 1/4-in. FF air-condenser, electret, and microelectromechanical systems (MEMS) microphones in an anechoic chamber. The uncertainty of this FF method is estimated by comparing the pressure sensitivity of an air-condenser FF microphone, as derived from the FF measurement, with that of an electrostatic actuator calibration. The root-mean-square difference is found to be +/- 0.3 dB over the range 1-80 kHz, and the combined standard uncertainty of the FF method, including other significant contributions, is +/- 0.41 dB.

Evaluation of a scale-model experiment to investigate long-range acoustic propagation

NASA Technical Reports Server (NTRS)

Parrott, Tony L.; Mcaninch, Gerry L.; Carlberg, Ingrid A.

1987-01-01

Tests were conducted to evaluate the feasibility of using a scale-model experiment situated in an anechoic facility to investigate long-range sound propagation over ground terrain. For a nominal scale factor of 100:1, attenuations along a linear array of six microphones colinear with a continuous-wave type of sound source were measured over a wavelength range from 10 to 160 for a nominal test frequency of 10 kHz. Most tests were made for a hard model surface (plywood), but limited tests were also made for a soft model surface (plywood with felt). For grazing-incidence propagation over the hard surface, measured and predicted attenuation trends were consistent for microphone locations out to between 40 and 80 wavelengths. Beyond 80 wavelengths, significant variability was observed that was caused by disturbances in the propagation medium. Also, there was evidence of extraneous propagation-path contributions to data irregularities at more remote microphones. Sensitivity studies for the hard-surface and microphone indicated a 2.5 dB change in the relative excess attenuation for a systematic error in source and microphone elevations on the order of 1 mm. For the soft-surface model, no comparable sensitivity was found.

Advanced flow noise reducing acoustic sensor arrays

NASA Astrophysics Data System (ADS)

Fine, Kevin; Drzymkowski, Mark; Cleckler, Jay

2009-05-01

SARA, Inc. has developed microphone arrays that are as effective at reducing flow noise as foam windscreens and sufficiently rugged for tough battlefield environments. These flow noise reducing (FNR) sensors have a metal body and are flat and conformally mounted so they can be attached to the roofs of land vehicles and are resistant to scrapes from branches. Flow noise at low Mach numbers is created by turbulent eddies moving with the fluid flow and inducing pressure variations on microphones. Our FNR sensors average the pressure over the diameter (~20 cm) of their apertures, reducing the noise created by all but the very largest eddies. This is in contrast to the acoustic wave which has negligible variation over the aperture at the frequencies of interest (f less or equal than 400 Hz). We have also post-processed the signals to further reduce the flow noise. Two microphones separated along the flow direction exhibit highly correlated noise. The time shift of the correlation corresponds to the time for the eddies in the flow to travel between the microphones. We have created linear microphone arrays parallel to the flow and have reduced flow noise as much as 10 to 15 dB by subtracting time-shifted signals.

Talker Localization Based on Interference between Transmitted and Reflected Audible Sound

NASA Astrophysics Data System (ADS)

Nakayama, Masato; Nakasako, Noboru; Shinohara, Toshihiro; Uebo, Tetsuji

In many engineering fields, distance to targets is very important. General distance measurement method uses a time delay between transmitted and reflected waves, but it is difficult to estimate the short distance. On the other hand, the method using phase interference to measure the short distance has been known in the field of microwave radar. Therefore, we have proposed the distance estimation method based on interference between transmitted and reflected audible sound, which can measure the distance between microphone and target with one microphone and one loudspeaker. In this paper, we propose talker localization method based on distance estimation using phase interference. We expand the distance estimation method using phase interference into two microphones (microphone array) in order to estimate talker position. The proposed method can estimate talker position by measuring the distance and direction between target and microphone array. In addition, talker's speech is regarded as a noise in the proposed method. Therefore, we also propose combination of the proposed method and CSP (Cross-power Spectrum Phase analysis) method which is one of the DOA (Direction Of Arrival) estimation methods. We evaluated the performance of talker localization in real environments. The experimental result shows the effectiveness of the proposed method.

The effect of multi-channel wide dynamic range compression, noise reduction, and the directional microphone on horizontal localization performance in hearing aid wearers.

PubMed

Keidser, Gitte; Rohrseitz, Kristin; Dillon, Harvey; Hamacher, Volkmar; Carter, Lyndal; Rass, Uwe; Convery, Elizabeth

2006-10-01

This study examined the effect that signal processing strategies used in modern hearing aids, such as multi-channel WDRC, noise reduction, and directional microphones have on interaural difference cues and horizontal localization performance relative to linear, time-invariant amplification. Twelve participants were bilaterally fitted with BTE devices. Horizontal localization testing using a 360 degrees loudspeaker array and broadband pulsed pink noise was performed two weeks, and two months, post-fitting. The effect of noise reduction was measured with a constant noise present at 80 degrees azimuth. Data were analysed independently in the left/right and front/back dimension and showed that of the three signal processing strategies, directional microphones had the most significant effect on horizontal localization performance and over time. Specifically, a cardioid microphone could decrease front/back errors over time, whereas left/right errors increased when different microphones were fitted to left and right ears. Front/back confusions were generally prominent. Objective measurements of interaural differences on KEMAR explained significant shifts in left/right errors. In conclusion, there is scope for improving the sense of localization in hearing aid users.

Noise measurements of turboprop airplanes at different overflight elevations

NASA Technical Reports Server (NTRS)

Mueller, K.

1990-01-01

In order to establish criteria for the regulation of propfan aircraft engine noise emissions, measurement tests of overhead flights of a METRO-3 and a FOKKER-50 aircraft were performed. The decibel levels captured by the ground car microphone are tabulated according to the height of the microphone from the ground as the recording vehicle followed the aircraft through the test flight patterns. Microphone heights of 1.5 and 10 meters from the ground are recorded and correlated to the flight altitudes of the aircraft, which ranged from 5182-6401 meters.

Micromachined microphone array on a chip for turbulent boundary layer measurements

NASA Astrophysics Data System (ADS)

Krause, Joshua Steven

A surface micromachined microphone array on a single chip has been successfully designed, fabricated, characterized, and tested for aeroacoustic purposes. The microphone was designed to have venting through the diaphragm, 64 elements (8x8) on the chip, and used a capacitive transduction scheme. The microphone was fabricated using the MEMSCAP PolyMUMPs process (a foundry polysilicon surface micromachining process) along with facilities at Tufts Micro and Nano Fabrication Facility (TMNF) where a Parylene-C passivation layer deposition and release of the microstructures were performed. The devices are packaged with low profile interconnects, presenting a maximum of 100 mum of surface topology. The design of an individual microphone was completed through the use of a lumped element model (LEM) to determine the theoretical performance of the microphone. Off-chip electronics were created to allow the microphone array outputs to be redirected to one of two channels, allowing dynamic reconfiguration of the effective transducer shape in software and provide 80 dB off isolation. The characterization was completed through the use of laser Doppler vibrometry (LDV), acoustic plane wave tube and free-field calibration, and electrical noise floor testing in a Faraday cage. Measured microphone sensitivity is 0.15 mV/Pa for an individual microphone and 8.7 mV/Pa for the entire array, in close agreement with model predictions. The microphones and electronics operate over the 200--40 000 Hz band. The dynamic range extends from 60 dB SPL in a 1 Hz band to greater than 150 dB SPL. Element variability was +/-0.05 mV/Pa in sensitivity with an array yield of 95%. Wind tunnel testing at flow rates of up to 205.8 m/s indicates that the devices continue to operate in flow without damage, and can be successfully reconfigured on the fly. Care has been taken to systematically remove contaminating signals (acoustic, vibration, and noise floor) from the wind tunnel data to determine actual turbulent pressure fluctuations beneath the turbulent boundary layer to an uncertainty level of 1 dB. Analysis of measured boundary layer pressure spectra at six flow rates from 34.3 m/s to 205.8 m/s indicate single point wall spectral measurements in close agreement to the empirical models of Goody, Chase-Howe, and Efimtsov above Mach 0.4. The MEMS data more closely resembles the magnitude of the Efimtsov model at higher frequencies (25% higher above 3 kHz for the Mach 0.6 case); however, the shape of the spectral model is closer to the model of Goody (50% lower for the Mach 0.6 case for all frequencies). The Chase-Howe model does fall directly on the MEMS data starting at 6 kHz, but has a sharper slope and does not resemble the data at below 6 kHz.

49 CFR 325.7 - Allowable noise levels.

Code of Federal Regulations, 2010 CFR

2010-10-01

... distance between the microphone location point and the microphone target point is— 31 ft ( 9.5m) or more... is based on motor carrier noise emission requirements specified in 40 CFR 202.20 and 40 CFR 202.21...

Characterizing acoustic shocks in high-performance jet aircraft flyover noise.

PubMed

Reichman, Brent O; Gee, Kent L; Neilsen, Tracianne B; Downing, J Micah; James, Michael M; Wall, Alan T; McInerny, Sally Anne

2018-03-01

Acoustic shocks have been previously documented in high-amplitude jet noise, including both the near and far fields of military jet aircraft. However, previous investigations into the nature and formation of shocks have historically concentrated on stationary, ground run-up measurements, and previous attempts to connect full-scale ground run-up and flyover measurements have omitted the effect of nonlinear propagation. This paper shows evidence for nonlinear propagation and the presence of acoustic shocks in acoustical measurements of F-35 flyover operations. Pressure waveforms, derivatives, and statistics indicate nonlinear propagation, and the resulting shock formation is significant at high engine powers. Variations due to microphone size, microphone height, and sampling rate are considered, and recommendations for future measurements are made. Metrics indicating nonlinear propagation are shown to be influenced by changes in sampling rate and microphone size, and exhibit less variation due to microphone height.

Electronic filters, hearing aids and methods

NASA Technical Reports Server (NTRS)

Engebretson, A. Maynard (Inventor)

1995-01-01

An electronic filter for an electroacoustic system. The system has a microphone for generating an electrical output from external sounds and an electrically driven transducer for emitting sound. Some of the sound emitted by the transducer returns to the microphone means to add a feedback contribution to its electrical output. The electronic filter includes a first circuit for electronic processing of the electrical output of the microphone to produce a first signal. An adaptive filter, interconnected with the first circuit, performs electronic processing of the first signal to produce an adaptive output to the first circuit to substantially offset the feedback contribution in the electrical output of the microphone, and the adaptive filter includes means for adapting only in response to polarities of signals supplied to and from the first circuit. Other electronic filters for hearing aids, public address systems and other electroacoustic systems, as well as such systems and methods of operating them are also disclosed.

Electronic filters, hearing aids and methods

NASA Technical Reports Server (NTRS)

Engebretson, A. Maynard (Inventor); O'Connell, Michael P. (Inventor); Zheng, Baohua (Inventor)

1991-01-01

An electronic filter for an electroacoustic system. The system has a microphone for generating an electrical output from external sounds and an electrically driven transducer for emitting sound. Some of the sound emitted by the transducer returns to the microphone means to add a feedback contribution to its electical output. The electronic filter includes a first circuit for electronic processing of the electrical output of the microphone to produce a filtered signal. An adaptive filter, interconnected with the first circuit, performs electronic processing of the filtered signal to produce an adaptive output to the first circuit to substantially offset the feedback contribution in the electrical output of the microphone, and the adaptive filter includes means for adapting only in response to polarities of signals supplied to and from the first circuit. Other electronic filters for hearing aids, public address systems and other electroacoustic systems, as well as such systems, and methods of operating them are also disclosed.

Implementation of the CMOS MEMS Condenser Microphone with Corrugated Metal Diaphragm and Silicon Back-Plate

PubMed Central

Huang, Chien-Hsin; Lee, Chien-Hsing; Hsieh, Tsung-Min; Tsao, Li-Chi; Wu, Shaoyi; Liou, Jhyy-Cheng; Wang, Ming-Yi; Chen, Li-Che; Yip, Ming-Chuen; Fang, Weileun

2011-01-01

This study reports a CMOS-MEMS condenser microphone implemented using the standard thin film stacking of 0.35 μm UMC CMOS 3.3/5.0 V logic process, and followed by post-CMOS micromachining steps without introducing any special materials. The corrugated diaphragm for the microphone is designed and implemented using the metal layer to reduce the influence of thin film residual stresses. Moreover, a silicon substrate is employed to increase the stiffness of the back-plate. Measurements show the sensitivity of microphone is −42 ± 3 dBV/Pa at 1 kHz (the reference sound-level is 94 dB) under 6 V pumping voltage, the frequency response is 100 Hz–10 kHz, and the S/N ratio >55 dB. It also has low power consumption of less than 200 μA, and low distortion of less than 1% (referred to 100 dB). PMID:22163953

Deconvolution Methods and Systems for the Mapping of Acoustic Sources from Phased Microphone Arrays

NASA Technical Reports Server (NTRS)

Humphreys, Jr., William M. (Inventor); Brooks, Thomas F. (Inventor)

2012-01-01

Mapping coherent/incoherent acoustic sources as determined from a phased microphone array. A linear configuration of equations and unknowns are formed by accounting for a reciprocal influence of one or more cross-beamforming characteristics thereof at varying grid locations among the plurality of grid locations. An equation derived from the linear configuration of equations and unknowns can then be iteratively determined. The equation can be attained by the solution requirement of a constraint equivalent to the physical assumption that the coherent sources have only in phase coherence. The size of the problem may then be reduced using zoning methods. An optimized noise source distribution is then generated over an identified aeroacoustic source region associated with a phased microphone array (microphones arranged in an optimized grid pattern including a plurality of grid locations) in order to compile an output presentation thereof, thereby removing beamforming characteristics from the resulting output presentation.

Non-Intrusive Sensor for In-Situ Measurement of Recession Rate of Ablative and Eroding Materials

NASA Technical Reports Server (NTRS)

Papadopoulos, George (Inventor); Tiliakos, Nicholas (Inventor); Thomson, Clint (Inventor); Benel, Gabriel (Inventor)

2014-01-01

A non-intrusive sensor for in-situ measurement of recession rate of heat shield ablatives. An ultrasonic wave source is carried in the housing. A microphone is also carried in the housing, for collecting the reflected ultrasonic waves from an interface surface of the ablative material. A time phasing control circuit is also included for time-phasing the ultrasonic wave source so that the waves reflected from the interface surface of the ablative material focus on the microphone, to maximize the acoustic pressure detected by the microphone and to mitigate acoustic velocity variation effects through the material through a de-coupling process that involves a software algorithm. A software circuit for computing the location off of which the ultrasonic waves scattered to focus back at the microphone is also included, so that the recession rate of the heat shield ablative may be monitored in real-time through the scan-focus approach.

Deconvolution methods and systems for the mapping of acoustic sources from phased microphone arrays

NASA Technical Reports Server (NTRS)

Brooks, Thomas F. (Inventor); Humphreys, Jr., William M. (Inventor)

2010-01-01

A method and system for mapping acoustic sources determined from a phased microphone array. A plurality of microphones are arranged in an optimized grid pattern including a plurality of grid locations thereof. A linear configuration of N equations and N unknowns can be formed by accounting for a reciprocal influence of one or more beamforming characteristics thereof at varying grid locations among the plurality of grid locations. A full-rank equation derived from the linear configuration of N equations and N unknowns can then be iteratively determined. A full-rank can be attained by the solution requirement of the positivity constraint equivalent to the physical assumption of statically independent noise sources at each N location. An optimized noise source distribution is then generated over an identified aeroacoustic source region associated with the phased microphone array in order to compile an output presentation thereof, thereby removing the beamforming characteristics from the resulting output presentation.

Design and development of second order MEMS sound pressure gradient sensor

NASA Astrophysics Data System (ADS)

Albahri, Shehab

The design and development of a second order MEMS sound pressure gradient sensor is presented in this dissertation. Inspired by the directional hearing ability of the parasitoid fly, Ormia ochracea, a novel first order directional microphone that mimics the mechanical structure of the fly's ears and detects the sound pressure gradient has been developed. While the first order directional microphones can be very beneficial in a large number of applications, there is great potential for remarkable improvements in performance through the use of second order systems. The second order directional microphone is able to provide a theoretical improvement in Sound to Noise ratio (SNR) of 9.5dB, compared to the first-order system that has its maximum SNR of 6dB. Although second order microphone is more sensitive to sound angle of incidence, the nature of the design and fabrication process imposes different factors that could lead to deterioration in its performance. The first Ormia ochracea second order directional microphone was designed in 2004 and fabricated in 2006 at Binghamton University. The results of the tested parts indicate that the Ormia ochracea second order directional microphone performs mostly as an Omni directional microphone. In this work, the previous design is reexamined and analyzed to explain the unexpected results. A more sophisticated tool implementing a finite element package ANSYS is used to examine the previous design response. This new tool is used to study different factors that used to be ignored in the previous design, mainly; response mismatch and fabrication uncertainty. A continuous model using Hamilton's principle is introduced to verify the results using the new method. Both models agree well, and propose a new way for optimizing the second order directional microphone using geometrical manipulation. In this work we also introduce a new fabrication process flow to increase the fabrication yield. The newly suggested method uses the shell layered analysis method in ANSYS. The developed models simulate the fabricated chips at different stages; with the stress at each layer is introduced using thermal loading. The results indicate a new fabrication process flow to increase the rigidity of the composite layers, and countering the deformation caused by the high stress in the thermal oxide layer.

Wake acoustic analysis and image decomposition via beamforming of microphone signal projections on wavelet subspaces

DOT National Transportation Integrated Search

2006-05-08

This paper describes the integration of wavelet analysis and time-domain beamforming : of microphone array output signals for analyzing the acoustic emissions from airplane : generated wake vortices. This integrated process provides visual and quanti...

Microphone Detects Boiler-Tube Leaks

NASA Technical Reports Server (NTRS)

Parthasarathy, S. P.

1985-01-01

Unit simple, sensitive, rugged, and reliable. Diaphragmless microphone detects leaks from small boiler tubes. Porous plug retains carbon granules in tube while allowing pressure changes to penetrate to granules. Has greater life expectancy than previous controllers and used in variety of hot corrosive atmospheres.

Adapter for mounting a microphone flush with the external surface of the skin of a pressurized aircraft

NASA Technical Reports Server (NTRS)

Cohn, R. B. (Inventor)

1983-01-01

A mounting device for securing a microphone pick up head flush with respect to the external surfaces of the skin of an aircraft for detecting shock waves passing thereover is described. The mount includes a sleeve mounted internally of the aircraft for capturing and supporting an electronics package having the microphone pick up head attached thereto in a manner such that the head is flush with the external surface of the aircraft skin and a pressure seal is established between the internal and external surfaces of the aircraft skin.

Acoustic Source Localization in Aircraft Interiors Using Microphone Array Technologies

NASA Technical Reports Server (NTRS)

Sklanka, Bernard J.; Tuss, Joel R.; Buehrle, Ralph D.; Klos, Jacob; Williams, Earl G.; Valdivia, Nicolas

2006-01-01

Using three microphone array configurations at two aircraft body stations on a Boeing 777-300ER flight test, the acoustic radiation characteristics of the sidewall and outboard floor system are investigated by experimental measurement. Analysis of the experimental data is performed using sound intensity calculations for closely spaced microphones, PATCH Inverse Boundary Element Nearfield Acoustic Holography, and Spherical Nearfield Acoustic Holography. Each method is compared assessing strengths and weaknesses, evaluating source identification capability for both broadband and narrowband sources, evaluating sources during transient and steady-state conditions, and quantifying field reconstruction continuity using multiple array positions.

Design and Use of Microphone Directional Arrays for Aeroacoustic Measurements

NASA Technical Reports Server (NTRS)

Humphreys, William M., Jr.; Brooks, Thomas F.; Hunter, William W., Jr.; Meadows, Kristine R.

1998-01-01

An overview of the development of two microphone directional arrays for aeroacoustic testing is presented. These arrays were specifically developed to measure airframe noise in the NASA Langley Quiet Flow Facility. A large aperture directional array using 35 flush-mounted microphones was constructed to obtain high resolution noise localization maps around airframe models. This array possesses a maximum diagonal aperture size of 34 inches. A unique logarithmic spiral layout design was chosen for the targeted frequency range of 2-30 kHz. Complementing the large array is a small aperture directional array, constructed to obtain spectra and directivity information from regions on the model. This array, possessing 33 microphones with a maximum diagonal aperture size of 7.76 inches, is easily moved about the model in elevation and azimuth. Custom microphone shading algorithms have been developed to provide a frequency- and position-invariant sensing area from 10-40 kHz with an overall targeted frequency range for the array of 5-60 kHz. Both arrays are employed in acoustic measurements of a 6 percent of full scale airframe model consisting of a main element NACA 632-215 wing section with a 30 percent chord half-span flap. Representative data obtained from these measurements is presented, along with details of the array calibration and data post-processing procedures.

Motorcycle detection and counting using stereo camera, IR camera, and microphone array

NASA Astrophysics Data System (ADS)

Ling, Bo; Gibson, David R. P.; Middleton, Dan

2013-03-01

Detection, classification, and characterization are the key to enhancing motorcycle safety, motorcycle operations and motorcycle travel estimation. Average motorcycle fatalities per Vehicle Mile Traveled (VMT) are currently estimated at 30 times those of auto fatalities. Although it has been an active research area for many years, motorcycle detection still remains a challenging task. Working with FHWA, we have developed a hybrid motorcycle detection and counting system using a suite of sensors including stereo camera, thermal IR camera and unidirectional microphone array. The IR thermal camera can capture the unique thermal signatures associated with the motorcycle's exhaust pipes that often show bright elongated blobs in IR images. The stereo camera in the system is used to detect the motorcyclist who can be easily windowed out in the stereo disparity map. If the motorcyclist is detected through his or her 3D body recognition, motorcycle is detected. Microphones are used to detect motorcycles that often produce low frequency acoustic signals. All three microphones in the microphone array are placed in strategic locations on the sensor platform to minimize the interferences of background noises from sources such as rain and wind. Field test results show that this hybrid motorcycle detection and counting system has an excellent performance.

Evaluation of Methods for In-Situ Calibration of Field-Deployable Microphone Phased Arrays

NASA Technical Reports Server (NTRS)

Humphreys, William M.; Lockard, David P.; Khorrami, Mehdi R.; Culliton, William G.; McSwain, Robert G.

2017-01-01

Current field-deployable microphone phased arrays for aeroacoustic flight testing require the placement of hundreds of individual sensors over a large area. Depending on the duration of the test campaign, the microphones may be required to stay deployed at the testing site for weeks or even months. This presents a challenge in regards to tracking the response (i.e., sensitivity) of the individual sensors as a function of time in order to evaluate the health of the array. To address this challenge, two different methods for in-situ tracking of microphone responses are described. The first relies on the use of an aerial sound source attached as a payload on a hovering small Unmanned Aerial System (sUAS) vehicle. The second relies on the use of individually excited ground-based sound sources strategically placed throughout the array pattern. Testing of the two methods was performed in microphone array deployments conducted at Fort A.P. Hill in 2015 and at Edwards Air Force Base in 2016. The results indicate that the drift in individual sensor responses can be tracked reasonably well using both methods. Thus, in-situ response tracking methods are useful as a diagnostic tool for monitoring the health of a phased array during long duration deployments.

Numerical calculation of listener-specific head-related transfer functions and sound localization: Microphone model and mesh discretization

PubMed Central

Ziegelwanger, Harald; Majdak, Piotr; Kreuzer, Wolfgang

2015-01-01

Head-related transfer functions (HRTFs) can be numerically calculated by applying the boundary element method on the geometry of a listener’s head and pinnae. The calculation results are defined by geometrical, numerical, and acoustical parameters like the microphone used in acoustic measurements. The scope of this study was to estimate requirements on the size and position of the microphone model and on the discretization of the boundary geometry as triangular polygon mesh for accurate sound localization. The evaluation involved the analysis of localization errors predicted by a sagittal-plane localization model, the comparison of equivalent head radii estimated by a time-of-arrival model, and the analysis of actual localization errors obtained in a sound-localization experiment. While the average edge length (AEL) of the mesh had a negligible effect on localization performance in the lateral dimension, the localization performance in sagittal planes, however, degraded for larger AELs with the geometrical error as dominant factor. A microphone position at an arbitrary position at the entrance of the ear canal, a microphone size of 1 mm radius, and a mesh with 1 mm AEL yielded a localization performance similar to or better than observed with acoustically measured HRTFs. PMID:26233020

Final report on key comparison CCAUV.A-K5: pressure calibration of laboratory standard microphones in the frequency range 2 Hz to 10 kHz

NASA Astrophysics Data System (ADS)

Avison, Janine; Barham, Richard

2014-01-01

This document and the accompanying spreadsheets constitute the final report for key comparison CCAUV.A-K5 on the pressure calibration of laboratory standard microphones in the frequency range from 2 Hz to 10 kHz. Twelve national measurement institutes took part in the key comparison and the National Physical Laboratory piloted the project. Two laboratory standard microphones IEC type LS1P were circulated to the participants and results in the form of regular calibration certificates were collected throughout the project. One of the microphones was subsequently deemed to have compromised stability for the purpose of deriving a reference value. Consequently the key comparison reference value (KCRV) has been made based on the weighted mean results for sensitivity level and for sensitivity phase from just one of the microphones. Corresponding degrees of equivalence (DoEs) have also been calculated and are presented. Main text. To reach the main text of this paper, click on Final Report. Note that this text is that which appears in Appendix B of the BIPM key comparison database kcdb.bipm.org/. The final report has been peer-reviewed and approved for publication by the CCAUV, according to the provisions of the CIPM Mutual Recognition Arrangement (CIPM MRA).

Single and Multiple Microphone Noise Reduction Strategies in Cochlear Implants

PubMed Central

Azimi, Behnam; Hu, Yi; Friedland, David R.

2012-01-01

To restore hearing sensation, cochlear implants deliver electrical pulses to the auditory nerve by relying on sophisticated signal processing algorithms that convert acoustic inputs to electrical stimuli. Although individuals fitted with cochlear implants perform well in quiet, in the presence of background noise, the speech intelligibility of cochlear implant listeners is more susceptible to background noise than that of normal hearing listeners. Traditionally, to increase performance in noise, single-microphone noise reduction strategies have been used. More recently, a number of approaches have suggested that speech intelligibility in noise can be improved further by making use of two or more microphones, instead. Processing strategies based on multiple microphones can better exploit the spatial diversity of speech and noise because such strategies rely mostly on spatial information about the relative position of competing sound sources. In this article, we identify and elucidate the most significant theoretical aspects that underpin single- and multi-microphone noise reduction strategies for cochlear implants. More analytically, we focus on strategies of both types that have been shown to be promising for use in current-generation implant devices. We present data from past and more recent studies, and furthermore we outline the direction that future research in the area of noise reduction for cochlear implants could follow. PMID:22923425

Objective analysis of ambisonics for hearing aid applications: Effect of listener's head, room reverberation, and directional microphones.

PubMed

Oreinos, Chris; Buchholz, Jörg M

2015-06-01

Recently, an increased interest has been demonstrated in evaluating hearing aids (HAs) inside controlled, but at the same time, realistic sound environments. A promising candidate that employs loudspeakers for realizing such sound environments is the listener-centered method of higher-order ambisonics (HOA). Although the accuracy of HOA has been widely studied, it remains unclear to what extent the results can be generalized when (1) a listener wearing HAs that may feature multi-microphone directional algorithms is considered inside the reconstructed sound field and (2) reverberant scenes are recorded and reconstructed. For the purpose of objectively validating HOA for listening tests involving HAs, a framework was developed to simulate the entire path of sounds presented in a modeled room, recorded by a HOA microphone array, decoded to a loudspeaker array, and finally received at the ears and HA microphones of a dummy listener fitted with HAs. Reproduction errors at the ear signals and at the output of a cardioid HA microphone were analyzed for different anechoic and reverberant scenes. It was found that the diffuse reverberation reduces the considered time-averaged HOA reconstruction errors which, depending on the considered application, suggests that reverberation can increase the usable frequency range of a HOA system.

Optimum sensor placement for microphone arrays

NASA Astrophysics Data System (ADS)

Rabinkin, Daniel V.

Microphone arrays can be used for high-quality sound pickup in reverberant and noisy environments. Sound capture using conventional single microphone methods suffers severe degradation under these conditions. The beamforming capabilities of microphone array systems allow highly directional sound capture, providing enhanced signal-to-noise ratio (SNR) when compared to single microphone performance. The overall performance of an array system is governed by its ability to locate and track sound sources and its ability to capture sound from desired spatial volumes. These abilities are strongly affected by the spatial placement of microphone sensors. A method is needed to optimize placement for a specified number of sensors in a given acoustical environment. The objective of the optimization is to obtain the greatest average system SNR for sound capture in the region of interest. A two-step sound source location method is presented. In the first step, time delay of arrival (TDOA) estimates for select microphone pairs are determined using a modified version of the Omologo-Svaizer cross-power spectrum phase expression. In the second step, the TDOA estimates are used in a least-mean-squares gradient descent search algorithm to obtain a location estimate. Statistics for TDOA estimate error as a function of microphone pair/sound source geometry and acoustic environment are gathered from a set of experiments. These statistics are used to model position estimation accuracy for a given array geometry. The effectiveness of sound source capture is also dependent on array geometry and the acoustical environment. Simple beamforming and time delay compensation (TDC) methods provide spatial selectivity but suffer performance degradation in reverberant environments. Matched filter array (MFA) processing can mitigate the effects of reverberation. The shape and gain advantage of the capture region for these techniques is described and shown to be highly influenced by the placement of array sensors. A procedure is developed to evaluate a given array configuration based on the above-mentioned metrics. Constrained placement optimizations are performed that maximize SNR for both TDC and MFA capture methods. Results are compared for various acoustic environments and various enclosure sizes. General guidelines are presented for placement strategy and bandwidth dependence, as they relate to reverberation levels, ambient noise, and enclosure geometry. An overall performance function is described based on these metrics. Performance of the microphone array system is also constrained by the design limitations of the supporting hardware. Two newly developed hardware architectures are presented that support the described algorithms. A low- cost 8-channel system with off-the-shelf componentry was designed and its performance evaluated. A massively parallel 512-channel custom-built system is in development-its capabilities and the rationale for its design are described.

Investigation of laser Doppler anemometry in developing a velocity-based measurement technique

NASA Astrophysics Data System (ADS)

Jung, Ki Won

2009-12-01

Acoustic properties, such as the characteristic impedance and the complex propagation constant, of porous materials have been traditionally characterized based on pressure-based measurement techniques using microphones. Although the microphone techniques have evolved since their introduction, the most general form of the microphone technique employs two microphones in characterizing the acoustic field for one continuous medium. The shortcomings of determining the acoustic field based on only two microphones can be overcome by using numerous microphones. However, the use of a number of microphones requires a careful and intricate calibration procedure. This dissertation uses laser Doppler anemometry (LDA) to establish a new measurement technique which can resolve issues that microphone techniques have: First, it is based on a single sensor, thus the calibration is unnecessary when only overall ratio of the acoustic field is required for the characterization of a system. This includes the measurements of the characteristic impedance and the complex propagation constant of a system. Second, it can handle multiple positional measurements without calibrating the signal at each position. Third, it can measure three dimensional components of velocity even in a system with a complex geometry. Fourth, it has a flexible adaptability which is not restricted to a certain type of apparatus only if the apparatus is transparent. LDA is known to possess several disadvantages, such as the requirement of a transparent apparatus, high cost, and necessity of seeding particles. The technique based on LDA combined with a curvefitting algorithm is validated through measurements on three systems. First, the complex propagation constant of the air is measured in a rigidly terminated cylindrical pipe which has very low dissipation. Second, the radiation impedance of an open-ended pipe is measured. These two parameters can be characterized by the ratio of acoustic field measured at multiple locations. Third, the power dissipated in a variable RLC load is measured. The three experiments validate the LDA technique proposed. The utility of the LDA method is then extended to the measurement of the complex propagation constant of the air inside a 100 ppi reticulated vitreous carbon (RVC) sample. Compared to measurements in the available studies, the measurement with the 100 ppi RVC sample supports the LDA technique in that it can achieve a low uncertainty in the determined quantity. This dissertation concludes with using the LDA technique for modal decomposition of the plane wave mode and the (1,1) mode that are driven simultaneously. This modal decomposition suggests that the LDA technique surpasses microphone-based techniques, because they are unable to determine the acoustic field based on an acoustic model with unconfined propagation constants for each modal component.

Speech Understanding and Sound Source Localization by Cochlear Implant Listeners Using a Pinna-Effect Imitating Microphone and an Adaptive Beamformer.

PubMed

Dorman, Michael F; Natale, Sarah; Loiselle, Louise

2018-03-01

Sentence understanding scores for patients with cochlear implants (CIs) when tested in quiet are relatively high. However, sentence understanding scores for patients with CIs plummet with the addition of noise. To assess, for patients with CIs (MED-EL), (1) the value to speech understanding of two new, noise-reducing microphone settings and (2) the effect of the microphone settings on sound source localization. Single-subject, repeated measures design. For tests of speech understanding, repeated measures on (1) number of CIs (one, two), (2) microphone type (omni, natural, adaptive beamformer), and (3) type of noise (restaurant, cocktail party). For sound source localization, repeated measures on type of signal (low-pass [LP], high-pass [HP], broadband noise). Ten listeners, ranging in age from 48 to 83 yr (mean = 57 yr), participated in this prospective study. Speech understanding was assessed in two noise environments using monaural and bilateral CIs fit with three microphone types. Sound source localization was assessed using three microphone types. In Experiment 1, sentence understanding scores (in terms of percent words correct) were obtained in quiet and in noise. For each patient, noise was first added to the signal to drive performance off of the ceiling in the bilateral CI-omni microphone condition. The other conditions were then administered at that signal-to-noise ratio in quasi-random order. In Experiment 2, sound source localization accuracy was assessed for three signal types using a 13-loudspeaker array over a 180° arc. The dependent measure was root-mean-score error. Both the natural and adaptive microphone settings significantly improved speech understanding in the two noise environments. The magnitude of the improvement varied between 16 and 19 percentage points for tests conducted in the restaurant environment and between 19 and 36 percentage points for tests conducted in the cocktail party environment. In the restaurant and cocktail party environments, both the natural and adaptive settings, when implemented on a single CI, allowed scores that were as good as, or better, than scores in the bilateral omni test condition. Sound source localization accuracy was unaltered by either the natural or adaptive settings for LP, HP, or wideband noise stimuli. The data support the use of the natural microphone setting as a default setting. The natural setting (1) provides better speech understanding in noise than the omni setting, (2) does not impair sound source localization, and (3) retains low-frequency sensitivity to signals from the rear. Moreover, bilateral CIs equipped with adaptive beamforming technology can engender speech understanding scores in noise that fall only a little short of scores for a single CI in quiet. American Academy of Audiology

Guidelines for Selecting Microphones for Human Voice Production Research

ERIC Educational Resources Information Center

Svec, Jan G.; Granqvist, Svante

2010-01-01

Purpose: This tutorial addresses fundamental characteristics of microphones (frequency response, frequency range, dynamic range, and directionality), which are important for accurate measurements of voice and speech. Method: Technical and voice literature was reviewed and analyzed. The following recommendations on desirable microphone…

A Retrospective Multicentre Cohort Review of Patient Characteristics and Surgical Aspects versus the Long-Term Outcomes for Recipients of a Fully Implantable Active Middle Ear Implant

PubMed Central

Lefebvre, Philippe P.; Gisbert, Javier; Cuda, Domenico; Tringali, Stéphane; Deveze, Arnaud

2017-01-01

Objective To summarise treatment outcomes compared to surgical and patient variables for a multicentre recipient cohort using a fully implantable active middle ear implant for hearing impairment. To describe the authors' preferred surgical technique to determine microphone placement. Study Design Multicentre retrospective, observational survey. Setting Five tertiary referral centres. Patients Carina recipients (66 ears, 62 subjects) using the current Cochlear® Carina® System or the legacy device, the Otologics® Fully Implantable Middle Ear, with a T2 transducer. Methods Patient file review and routine clinical review. Patient outcomes assessed were satisfaction, daily use and feedback reports at the first fitting and ≥12 months after implantation. Descriptive and statistical analysis of correlations of variables and their influence on outcomes was performed. Independently reported preferred methods for microphone placement are collectively summarised. Results The average implant experience was 3.5 years. Satisfaction increased significantly over time (p < 0.05). No correlation with covariates examined was observed. Feedback significantly decreased over time, showing a significant correlation with microphone location, primary motivation, gender, age at implantation, and contralateral hearing aid use (p < 0.05). Patient satisfaction was inversely correlated with reports of system feedback (p < 0.05). The implantable microphone was most commonly on the posterior inferior mastoid line, in 42/66 (65%) cases, correlating with less likelihood for feedback and consistent with author surgical preference. Conclusion Carina recipients in this study present as satisfied consistent daily users with very few reports of persistent feedback. As microphone location is an influencing factor, a careful surgical consideration of microphone placement is required. The authors prefer a posterior inferior mastoid line position whenever possible. PMID:28052264

A microacoustic analysis including viscosity and thermal conductivity to model the effect of the protective cap on the acoustic response of a MEMS microphone

PubMed Central

Homentcovschi, D.; Miles, R. N.; Loeppert, P. V.; Zuckerwar, A. J.

2013-01-01

An analysis is presented of the effect of the protective cover on the acoustic response of a miniature silicon microphone. The microphone diaphragm is contained within a small rectangular enclosure and the sound enters through a small hole in the enclosure's top surface. A numerical model is presented to predict the variation in the sound field with position within the enclosure. An objective of this study is to determine up to which frequency the pressure distribution remains sufficiently uniform so that a pressure calibration can be made in free space. The secondary motivation for this effort is to facilitate microphone design by providing a means of predicting how the placement of the microphone diaphragm in the package affects the sensitivity and frequency response. While the size of the package is typically small relative to the wavelength of the sounds of interest, because the dimensions of the package are on the order of the thickness of the viscous boundary layer, viscosity can significantly affect the distribution of sound pressure around the diaphragm. In addition to the need to consider viscous effects, it is shown here that one must also carefully account for thermal conductivity to properly represent energy dissipation at the system's primary acoustic resonance frequency. The sound field is calculated using a solution of the linearized system consisting of continuity equation, Navier-Stokes equations, the state equation and the energy equation using a finite element approach. The predicted spatial variation of both the amplitude and phase of the sound pressure is shown over the range of audible frequencies. Excellent agreement is shown between the predicted and measured effects of the package on the microphone's sensitivity. PMID:24701031

Noise-Canceling Helmet Audio System

NASA Technical Reports Server (NTRS)

Seibert, Marc A.; Culotta, Anthony J.

2007-01-01

A prototype helmet audio system has been developed to improve voice communication for the wearer in a noisy environment. The system was originally intended to be used in a space suit, wherein noise generated by airflow of the spacesuit life-support system can make it difficult for remote listeners to understand the astronaut s speech and can interfere with the astronaut s attempt to issue vocal commands to a voice-controlled robot. The system could be adapted to terrestrial use in helmets of protective suits that are typically worn in noisy settings: examples include biohazard, fire, rescue, and diving suits. The system (see figure) includes an array of microphones and small loudspeakers mounted at fixed positions in a helmet, amplifiers and signal-routing circuitry, and a commercial digital signal processor (DSP). Notwithstanding the fixed positions of the microphones and loudspeakers, the system can accommodate itself to any normal motion of the wearer s head within the helmet. The system operates in conjunction with a radio transceiver. An audio signal arriving via the transceiver intended to be heard by the wearer is adjusted in volume and otherwise conditioned and sent to the loudspeakers. The wearer s speech is collected by the microphones, the outputs of which are logically combined (phased) so as to form a microphone- array directional sensitivity pattern that discriminates in favor of sounds coming from vicinity of the wearer s mouth and against sounds coming from elsewhere. In the DSP, digitized samples of the microphone outputs are processed to filter out airflow noise and to eliminate feedback from the loudspeakers to the microphones. The resulting conditioned version of the wearer s speech signal is sent to the transceiver.

Feasible pickup from intact ossicular chain with floating piezoelectric microphone.

PubMed

Kang, Hou-Yong; Na, Gao; Chi, Fang-Lu; Jin, Kai; Pan, Tie-Zheng; Gao, Zhen

2012-02-22

Many microphones have been developed to meet with the implantable requirement of totally implantable cochlear implant (TICI). However, a biocompatible one without destroying the intactness of the ossicular chain still remains under investigation. Such an implantable floating piezoelectric microphone (FPM) has been manufactured and shows an efficient electroacoustic performance in vitro test at our lab. We examined whether it pick up sensitively from the intact ossicular chain and postulated whether it be an optimal implantable one. Animal controlled experiment: five adult cats (eight ears) were sacrificed as the model to test the electroacoustic performance of the FPM. Three groups were studied: (1) the experiment group (on malleus): the FPM glued onto the handle of the malleus of the intact ossicular chains; (2) negative control group (in vivo): the FPM only hung into the tympanic cavity; (3) positive control group (Hy-M30): a HiFi commercial microphone placed close to the site of the experiment ear. The testing speaker played pure tones orderly ranged from 0.25 to 8.0 kHz. The FPM inside the ear and the HiFi microphone simultaneously picked up acoustic vibration which recorded as .wav files to analyze. The FPM transformed acoustic vibration sensitively and flatly as did the in vitro test across the frequencies above 2.0 kHz, whereas inefficiently below 1.0 kHz for its overloading mass. Although the HiFi microphone presented more efficiently than the FPM did, there was no significant difference at 3.0 kHz and 8.0 kHz. It is feasible to develop such an implantable FPM for future TICIs and TIHAs system on condition that the improvement of Micro Electromechanical System and piezoelectric ceramic material technology would be applied to reduce its weight and minimize its size.

Field-Deployable Acoustic Digital Systems for Noise Measurement

NASA Technical Reports Server (NTRS)

Shams, Qamar A.; Wright, Kenneth D.; Lunsford, Charles B.; Smith, Charlie D.

2000-01-01

Langley Research Center (LaRC) has for years been a leader in field acoustic array measurement technique. Two field-deployable digital measurement systems have been developed to support acoustic research programs at LaRC. For several years, LaRC has used the Digital Acoustic Measurement System (DAMS) for measuring the acoustic noise levels from rotorcraft and tiltrotor aircraft. Recently, a second system called Remote Acquisition and Storage System (RASS) was developed and deployed for the first time in the field along with DAMS system for the Community Noise Flight Test using the NASA LaRC-757 aircraft during April, 2000. The test was performed at Airborne Airport in Wilmington, OH to validate predicted noise reduction benefits from alternative operational procedures. The test matrix was composed of various combinations of altitude, cutback power, and aircraft weight. The DAMS digitizes the acoustic inputs at the microphone site and can be located up to 2000 feet from the van which houses the acquisition, storage and analysis equipment. Digitized data from up to 10 microphones is recorded on a Jaz disk and is analyzed post-test by microcomputer system. The RASS digitizes and stores acoustic inputs at the microphone site that can be located up to three miles from the base station and can compose a 3 mile by 3 mile array of microphones. 16-bit digitized data from the microphones is stored on removable Jaz disk and is transferred through a high speed array to a very large high speed permanent storage device. Up to 30 microphones can be utilized in the array. System control and monitoring is accomplished via Radio Frequency (RF) link. This paper will present a detailed description of both systems, along with acoustic data analysis from both systems.

77 FR 64446 - Wireless Microphones Proceeding

Federal Register 2010, 2011, 2012, 2013, 2014

2012-10-22

... FEDERAL COMMUNICATIONS COMMISSION 47 CFR Parts 15, 74, and 90 [WT Docket Nos. 08-166, 08-167, ET Docket No. 10-24; DA 12-1570] Wireless Microphones Proceeding AGENCY: Federal Communications Commission.... [ssquf] Federal Communications Commission's Web site: http://www.fcc.gov/cgb/ecfs2/ . Follow the...

A New Kind of Laser Microphone Using High Sensitivity Pulsed Laser Vibrometer

NASA Technical Reports Server (NTRS)

Wang, Chen-Chia; Trivedi, Sudhir; Jin, Feng; Swaminathan, V.; Prasad, Narasimha S.

2008-01-01

We demonstrate experimentally a new kind of laser microphone using a highly sensitive pulsed laser vibrometer. By using the photo-electromotive-force (photo-EMF) sensors, we present data indicating the real-time detection of surface displacements as small as 4 pm.

Hydrogel microphones for stealthy underwater listening

PubMed Central

Gao, Yang; Song, Jingfeng; Li, Shumin; Elowsky, Christian; Zhou, You; Ducharme, Stephen; Chen, Yong Mei; Zhou, Qin; Tan, Li

2016-01-01

Exploring the abundant resources in the ocean requires underwater acoustic detectors with a high-sensitivity reception of low-frequency sound from greater distances and zero reflections. Here we address both challenges by integrating an easily deformable network of metal nanoparticles in a hydrogel matrix for use as a cavity-free microphone. Since metal nanoparticles can be densely implanted as inclusions, and can even be arranged in coherent arrays, this microphone can detect static loads and air breezes from different angles, as well as underwater acoustic signals from 20 Hz to 3 kHz at amplitudes as low as 4 Pa. Unlike dielectric capacitors or cavity-based microphones that respond to stimuli by deforming the device in thickness directions, this hydrogel device responds with a transient modulation of electric double layers, resulting in an extraordinary sensitivity (217 nF kPa−1 or 24 μC N−1 at a bias of 1.0 V) without using any signal amplification tools. PMID:27554792

A two-dimensional model of a directional microphone: calculation of the normal force and moment on the diaphragm.

PubMed

Homentcovschi, Dorel; Aubrey, Matthew J; Miles, Ronald N

2006-02-01

It has been shown that the parasitoid fly Ormia Ochracea exhibits exceptional sound localization ability achieved through the mechanical coupling of its eardrums [R. N. Miles et al., J. Acoust. Soc. Am. 98, 3059-3070 (1995)]. Based on this biological system a new directional microphone has been designed, having as a basic element a special diaphragm undergoing a rocking motion. This paper considers a 2D model of the microphone in which the diaphragm is considered as a 2D plate having slits on the sides. The slits lead to a backing volume limited by an infinite rigid wall parallel to the diaphragm in its neutral position. The reflection and diffraction of an incoming plane wave by this system are studied to determine the resultant force and resultant moment of pressure upon the diaphragm. The results show that such a microphone will be driven better in the case of narrow slits and deep cavities.

Analysis of the cochlear microphonic to a low-frequency tone embedded in filtered noise

PubMed Central

Chertoff, Mark E.; Earl, Brian R.; Diaz, Francisco J.; Sorensen, Janna L.

2012-01-01

The cochlear microphonic was recorded in response to a 733 Hz tone embedded in noise that was high-pass filtered at 25 different frequencies. The amplitude of the cochlear microphonic increased as the high-pass cutoff frequency of the noise increased. The amplitude growth for a 60 dB SPL tone was steeper and saturated sooner than that of an 80 dB SPL tone. The growth for both signal levels, however, was not entirely cumulative with plateaus occurring at about 4 and 7 mm from the apex. A phenomenological model of the electrical potential in the cochlea that included a hair cell probability function and spiral geometry of the cochlea could account for both the slope of the growth functions and the plateau regions. This suggests that with high-pass-filtered noise, the cochlear microphonic recorded at the round window comes from the electric field generated at the source directed towards the electrode and not down the longitudinal axis of the cochlea. PMID:23145616

Leak locating microphone, method and system for locating fluid leaks in pipes

DOEpatents

Kupperman, David S.; Spevak, Lev

1994-01-01

A leak detecting microphone inserted directly into fluid within a pipe includes a housing having a first end being inserted within the pipe and a second opposed end extending outside the pipe. A diaphragm is mounted within the first housing end and an acoustic transducer is coupled to the diaphragm for converting acoustical signals to electrical signals. A plurality of apertures are provided in the housing first end, the apertures located both above and below the diaphragm, whereby to equalize fluid pressure on either side of the diaphragm. A leak locating system and method are provided for locating fluid leaks within a pipe. A first microphone is installed within fluid in the pipe at a first selected location and sound is detected at the first location. A second microphone is installed within fluid in the pipe at a second selected location and sound is detected at the second location. A cross-correlation is identified between the detected sound at the first and second locations for identifying a leak location.

A fast signal subspace approach for the determination of absolute levels from phased microphone array measurements

NASA Astrophysics Data System (ADS)

Sarradj, Ennes

2010-04-01

Phased microphone arrays are used in a variety of applications for the estimation of acoustic source location and spectra. The popular conventional delay-and-sum beamforming methods used with such arrays suffer from inaccurate estimations of absolute source levels and in some cases also from low resolution. Deconvolution approaches such as DAMAS have better performance, but require high computational effort. A fast beamforming method is proposed that can be used in conjunction with a phased microphone array in applications with focus on the correct quantitative estimation of acoustic source spectra. This method bases on an eigenvalue decomposition of the cross spectral matrix of microphone signals and uses the eigenvalues from the signal subspace to estimate absolute source levels. The theoretical basis of the method is discussed together with an assessment of the quality of the estimation. Experimental tests using a loudspeaker setup and an airfoil trailing edge noise setup in an aeroacoustic wind tunnel show that the proposed method is robust and leads to reliable quantitative results.

Probe Microphone Measurements: 20 Years of Progress

PubMed Central

Mueller, H. Gustav

2001-01-01

Probe-microphone testing was conducted in the laboratory as early as the 1940s (e.g., the classic work of Wiener and Ross, reported in 1946), however, it was not until the late 1970s that a “dispenser friendly” system was available for testing hearing aids in the real ear. In this case, the term “dispenser friendly,” is used somewhat loosely. The 1970s equipment that I'm referring to was first described in a paper that was presented by Earl Harford, Ph.D. in September of 1979 at the International Ear Clinics' Symposium in Minneapolis. At this meeting, Earl reported on his clinical experiences of testing hearing aids in the real ear using a miniature (by 1979 standards) Knowles microphone. The microphone was coupled to an interfacing impedance matching system (developed by David Preves, Ph.D., who at the time worked at Starkey Laboratories) which could be used with existing hearing aid analyzer systems (see Harford, 1980 for review of this early work). Unlike today's probe tube microphone systems, this early method of clinical real-ear measurement involved putting the entire microphone (about 4mm by 5mm by 2mm) in the ear canal down by the eardrum of the patient. If you think cerumen is a problem with probe-mic measurements today, you should have seen the condition of this microphone after a day's work! While this early instrumentation was a bit cumbersome, we quickly learned the advantages that probe-microphone measures provided in the fitting of hearing aids. We frequently ran into calibration and equalization problems, not to mention a yelp or two from the patient, but the resulting information was worth the trouble. Help soon arrived. In the early 1980s, the first computerized probe-tube microphone system, the Rastronics CCI-10 (developed in Denmark by Steen Rasmussen), entered the U.S. market (Nielsen and Rasmussen, 1984). This system had a silicone tube attached to the microphone (the transmission of sound through this tube was part of the calibration process), which eliminated the need to place the microphone itself in the ear canal. By early 1985, three or four different manufactures had introduced this new type of computerized probe-microphone equipment, and this hearing aid verification procedure became part of the standard protocol for many audiology clinics. At his time, the POGO (Prescription Of Gain and Output) and Libby 1/3 prescriptive fitting methods were at the peak of their popularity, and a revised NAL (National Acoustic Laboratories) procedure was just being introduced. All three of these methods were based on functional gain, but insertion gain easily could be substituted, and therefore, manufacturers included calculation of these prescriptive targets as part of the probe-microphone equipment software. Audiologists, frustrated with the tedious and unreliable functional gain procedure they had been using, soon developed a fascination with matching real-ear results to prescriptive targets on a computer monitor. In some ways, not a lot has changed since those early days of probe-microphone measurements. Most people who use this equipment simply run a gain curve for a couple inputs and see if it's close to prescriptive target—something that could be accomplished using the equipment from 1985. Contrary to the predictions of many, probe-mic measures have not become the “standard hearing aid verification procedure.” (Mueller and Strouse, 1995). There also has been little or no increase in the use of this equipment in recent years. In 1998, I reported on a survey that was conducted by The Hearing Journal regarding the use of probe-microphone measures (Mueller, 1998). We first looked at what percent of people dispensing hearing aids own (or have immediate access to) probe-microphone equipment. Our results showed that 23% of hearing instrument specialists and 75% of audiologists have this equipment. Among audiologists, ownership varied among work settings: 91% for hospitals/clinics, 73% for audiologists working for physicians, and 69% for audiologists in private practice. But more importantly, and a bit puzzling, was the finding that showed that nearly one half of the people who fit hearing aids and have access to this equipment, seldom or never use it. I doubt that the use rate of probe-microphone equipment has changed much in the last three years, and if anything, I suspect it has gone down. Why do I say that? As programmable hearing aids have become the standard fitting in many clinics, it is tempting to become enamoured with the simulated gain curves on the fitting screen, somehow believing that this is what really is happening in the real ear. Additionally, some dispensers have been told that you can't do reliable probe-mic testing with modern hearing aids—this of course is not true, and we'll address this issue in the Frequently Asked Questions portion of this paper. The infrequent use of probe-mic testing among dispensers is discouraging, and let's hope that probe-mic equipment does not suffer the fate of the rowing machine stored in your garage. A lot has changed over the years with the equipment itself, and there are also expanded clinical applications and procedures. We have new manufacturers, procedures, acronyms and noises. We have test procedures that allow us to accurately predict the output of a hearing aid in an infant's ear. We now have digital hearing aids, which provide us the opportunity to conduct real-ear measures of the effects of digital noise reduction, speech enhancement, adaptive feedback, expansion, and all the other features. Directional microphone hearing aids have grown in popularity and what better way to assess the real-ear directivity than with probe-mic measures? The array of assistive listening devices has expanded, and so has the role of the real-ear assessment of these products. And finally, with today's PC -based systems, we can program our hearing aids and simultaneously observe the resulting real-ear effects on the same fitting screen, or even conduct an automated target fitting using earcanal monitoring of the output. There have been a lot of changes, and we'll talk about all of them in this issue of Trends. PMID:25425897

Optimization of Acoustic Pressure Measurements for Impedance Eduction

NASA Technical Reports Server (NTRS)

Jones, M. G.; Watson, W. R.; Nark, D. M.

2007-01-01

As noise constraints become increasingly stringent, there is continued emphasis on the development of improved acoustic liner concepts to reduce the amount of fan noise radiated to communities surrounding airports. As a result, multiple analytical prediction tools and experimental rigs have been developed by industry and academia to support liner evaluation. NASA Langley has also placed considerable effort in this area over the last three decades. More recently, a finite element code (Q3D) based on a quasi-3D implementation of the convected Helmholtz equation has been combined with measured data acquired in the Langley Grazing Incidence Tube (GIT) to reduce liner impedance in the presence of grazing flow. A new Curved Duct Test Rig (CDTR) has also been developed to allow evaluation of liners in the presence of grazing flow and controlled, higher-order modes, with straight and curved waveguides. Upgraded versions of each of these two test rigs are expected to begin operation by early 2008. The Grazing Flow Impedance Tube (GFIT) will replace the GIT, and additional capabilities will be incorporated into the CDTR. The current investigation uses the Q3D finite element code to evaluate some of the key capabilities of these two test rigs. First, the Q3D code is used to evaluate the microphone distribution designed for the GFIT. Liners ranging in length from 51 to 610 mm are investigated to determine whether acceptable impedance eduction can be achieved with microphones placed on the wall opposite the liner. This analysis indicates the best results are achieved for liner lengths of at least 203 mm. Next, the effects of moving this GFIT microphone array to the wall adjacent to the liner are evaluated, and acceptable results are achieved if the microphones are placed off the centerline. Finally, the code is used to investigate potential microphone placements in the CDTR rigid wall adjacent to the wall containing an acoustic liner, to determine if sufficient fidelity can be achieved with 32 microphones available for this purpose. Initial results indicate 32 microphones can provide acceptable measurements to support impedance eduction with this test rig.

Effect of occlusion, directionality and age on horizontal localization

NASA Astrophysics Data System (ADS)

Alworth, Lynzee Nicole

Localization acuity of a given listener is dependent upon the ability discriminate between interaural time and level disparities. Interaural time differences are encoded by low frequency information whereas interaural level differences are encoded by high frequency information. Much research has examined effects of hearing aid microphone technologies and occlusion separately and prior studies have not evaluated age as a factor in localization acuity. Open-fit hearing instruments provide new earmold technologies and varying microphone capabilities; however, these instruments have yet to be evaluated with regard to horizontal localization acuity. Thus, the purpose of this study is to examine the effects of microphone configuration, type of dome in open-fit hearing instruments, and age on the horizontal localization ability of a given listener. Thirty adults participated in this study and were grouped based upon hearing sensitivity and age (young normal hearing, >50 years normal hearing, >50 hearing impaired). Each normal hearing participant completed one localization experiment (unaided/unamplified) where they listened to the stimulus "Baseball" and selected the point of origin. Hearing impaired listeners were fit with the same two receiver-in-the-ear hearing aids and same dome types, thus controlling for microphone technologies, type of dome, and fitting between trials. Hearing impaired listeners completed a total of 7 localization experiments (unaided/unamplified; open dome: omnidirectional, adaptive directional, fixed directional; micromold: omnidirectional, adaptive directional, fixed directional). Overall, results of this study indicate that age significantly affects horizontal localization ability as younger adult listeners with normal hearing made significantly fewer localization errors than older adult listeners with normal hearing. Also, results revealed a significant difference in performance between dome type; however, upon further examination was not significant. Therefore, results examining type of dome should be viewed with caution. Results examining microphone configuration and microphone configuration by dome type were not significant. Moreover, results evaluating performance relative to unaided (unamplified) were not significant. Taken together, these results suggest open-fit hearing instruments, regardless of microphone or dome type, do not degrade horizontal localization acuity within a given listener relative to their 'older aged' normal hearing counterparts in quiet environments.

Test-retest reliability of probe-microphone verification in children fitted with open and closed hearing aid tips.

PubMed

Kim, Hannah; Ricketts, Todd A

2013-01-01

To investigate the test-retest reliability of real-ear aided response (REAR) measures in open and closed hearing aid fittings in children using appropriate probe-microphone calibration techniques (stored equalization for open fittings and concurrent equalization for closed fittings). Probe-microphone measurements were completed for two mini-behind-the-ear (BTE) hearing aids which were coupled to the ear using open and closed eartips via thin (0.9 mm) tubing. Before probe-microphone testing, the gain of each of the test hearing aids was programmed using an artificial ear simulator (IEC 711) and a Knowles Electronic Manikin for Acoustic Research to match the National Acoustic Laboratories-Non-Linear, version 1 targets for one of two separate hearing loss configurations using an Audioscan Verifit. No further adjustments were made, and the same amplifier gain was used within each hearing aid across both eartip configurations and all participants. Probe-microphone testing included real-ear occluded response (REOR) and REAR measures using the Verifit's standard speech signal (the carrot passage) presented at 65 dB sound pressure level (SPL). Two repeated probe-microphone measures were made for each participant with the probe-tube and hearing aid removed and repositioned between each trial in order to assess intrasubject measurement variability. These procedures were repeated using both open and closed domes. Thirty-two children, ages ranging from 4 to 14 yr. The test-retest standard deviations for open and closed measures did not exceed 4 dB at any frequency. There was also no significant difference between the open (stored equalization) and closed (concurrent equalization) methods. Reliability was particularly similar in the high frequencies and was also quite similar to that reported in previous research. There was no correlation between reliability and age, suggesting high reliability across all ages evaluated. The findings from this study suggest that reliable probe-microphone measurements are obtainable on children 4 yr and older for both traditional unvented and open-canal hearing aid fittings. These data suggest that clinicians should not avoid fitting open technology to children as young as 4 y because of concerns regarding the reliability of verification techniques. American Academy of Audiology.

Benefits of the Fiber Optic versus the Electret Microphone in Voice Amplification

ERIC Educational Resources Information Center

Kyriakou, Kyriaki; Fisher, Helene R.

2013-01-01

Background: Voice disorders that result in reduced loudness may cause difficulty in communicating, socializing and participating in occupational activities. Amplification is often recommended in order to facilitate functional communication, reduce vocal load and avoid developing maladaptive compensatory behaviours. The most common microphone used…

Launcher Dynamic Data Acquisition

DTIC Science & Technology

2012-07-31

K PR Pressure PR Pressure PR Accelerometer PR Accelerometer PR Accelerometer PR Pressure PR Pressure IEPE Microphone IEPE ...transducers, displacement potentiometers, or Integrated Electronics Piezoelectric ( IEPE ) microphones and accelerometers. The characteristics of these...Engineering Units HCl hydrogen chloride HVAC heating ventilation and cooling Hz hertz IEC International Electrotechnical Commission IEPE

Functional hearing in the classroom: assistive listening devices for students with hearing impairment in a mainstream school setting.

PubMed

Zanin, Julien; Rance, Gary

2016-12-01

To assess the benefit of assistive listening devices (ALDs) for students with hearing impairment in mainstream schools. Speech recognition (CNC words) in background noise was assessed in a typical classroom. Participants underwent testing using four device configurations: (1) HA(s)/CI(s) alone, (2) soundfield amplification, (3) remote microphone (Roger Pen) on desk and (4) remote microphone at the loudspeaker. A sub-group of students subsequently underwent a 2-week classroom trial of each ALD. Degree of improvement from baseline [HA(s)/CI(s)] alone was assessed using teacher and student Listening Inventory for Education-Revised (LIFE-R) questionnaires. In all, 20 students, aged 12.5-18.9 years, underwent speech recognition assessment. In total, 10 of these participated in the classroom trial. Hearing loss ranged from mild-to-profound levels. Performance in each ALD configuration was higher than for HAs/CIs alone (p < 0.001). Teacher and student LIFE-R results indicated significant improvement in listening/communication when using the remote microphone in conjunction with HAs/CIs (p < 0.05). There was no difference between the soundfield system and the baseline measurement (p > 0.05). Speech recognition improvements were demonstrated with the implementation of both remote microphones and soundfield systems. Both students and teachers reported functional hearing advantages in the classroom when using the remote microphone in concert with their standard hearing devices.

AMELIA CESTOL Test: Acoustic Characteristics of Circulation Control Wing with Leading-and Trailing-Edge Slot Blowing

NASA Technical Reports Server (NTRS)

Horne, Clifton; Burnside, Nathan J.

2013-01-01

Aeroacoustic measurements of the 11 % scale full-span AMELIA CESTOL model with leading- and trailing-edge slot blowing circulation control (CCW) wing were obtained during a recent test in the Arnold Engineering Development Center 40- by 80-Ft. Wind Tunnel at NASA Ames Research Center, Sound levels and spectra were acquired with seven in-flow microphones and a 48-element phased microphone array for a variety of vehicle configurations, CCW slot flow rates, and forward speeds, Corrections to the measurements and processing are in progress, however the data from selected configurations presented in this report confirm good measurement quality and dynamic range over the test conditions, Array beamform maps at 40 kts tunnel speed show that the trailing edge flap source is dominant for most frequencies at flap angles of 0deg and 60deg, The overall sound level for the 60deg flap was similar to the 0deg flap for most slot blowing rates forward of 90deg incidence, but was louder by up to 6 dB for downstream angles, At 100 kts, the in-flow microphone levels were louder than the sensor self-noise for the higher blowing rates, while passive and active background noise suppression methods for the microphone array revealed source levels as much as 20 dB lower than observed with the in-flow microphones,

A Simple Laser Microphone for Classroom Demonstration

ERIC Educational Resources Information Center

Moses, James M.; Trout, K. P.

2006-01-01

Communication through the modulation of electromagnetic radiation has become a foundational technique in modern technology. In this paper we discuss a modern day method of eavesdropping based upon the modulation of laser light reflected from a window pane. A simple and affordable classroom demonstration of a "laser microphone" is…

14 CFR 29.1457 - Cockpit voice recorders.

Code of Federal Regulations, 2010 CFR

2010-01-01

... second pilot stations and voice communications of other crewmembers on the flight deck when directed to those stations; or (2) By installing a continually energized or voice-actuated lip microphone at the first and second pilot stations. The microphone specified in this paragraph must be so located and, if...

Active noise canceling system for mechanically cooled germanium radiation detectors

DOEpatents

Nelson, Karl Einar; Burks, Morgan T

2014-04-22

A microphonics noise cancellation system and method for improving the energy resolution for mechanically cooled high-purity Germanium (HPGe) detector systems. A classical adaptive noise canceling digital processing system using an adaptive predictor is used in an MCA to attenuate the microphonics noise source making the system more deployable.

Effect of Three Classroom Listening Conditions on Speech Intelligibility

ERIC Educational Resources Information Center

Ross, Mark; Giolas, Thomas G.

1971-01-01

Speech discrimination scores for 13 deaf children were obtained in a classroom under: usual listening condition (hearing aid or not), binaural listening situation using auditory trainer/FM receiver with wireless microphone transmitter turned off, and binaural condition with inputs from auditory trainer/FM receiver and wireless microphone/FM…

47 CFR 73.14 - AM broadcast definitions.

Code of Federal Regulations, 2011 CFR

2011-10-01

.... The term “field strength” is synonymous with the term “field intensity” as contained elsewhere in this... output of a microphone or combination of microphones placed so as to convey the intensity, time, and location of sounds originated predominately to the listener's left (or right) of the center of the...

Analysis of STS-3 Get Away Special (GAS) flight data and vibration specification for gas payloads

NASA Technical Reports Server (NTRS)

Talapatra, D. C.

1983-01-01

During the Space Transportation System (STS)-3 mission, a Get Away Special (GAS) canister was flown. In order to determine the flight environment for GAS payloads, triaxial accelerometers and a microphone were installed inside the GAS canister. Data from these accelerometers and the microphone were analyzed. The microphone data is presented as overall sound pressure level (SPL) and one-third octave band time history plots. And the accelerometer data is provided in the forms of instantaneous time history, RMS time history and power spectral density plots. Also based on this flight data, vibration test specification for GAS payloads was developed and the recommended specification is presented here.

Impedance measurement using a two-microphone, random-excitation method

NASA Technical Reports Server (NTRS)

Seybert, A. F.; Parrott, T. L.

1978-01-01

The feasibility of using a two-microphone, random-excitation technique for the measurement of acoustic impedance was studied. Equations were developed, including the effect of mean flow, which show that acoustic impedance is related to the pressure ratio and phase difference between two points in a duct carrying plane waves only. The impedances of a honeycomb ceramic specimen and a Helmholtz resonator were measured and compared with impedances obtained using the conventional standing-wave method. Agreement between the two methods was generally good. A sensitivity analysis was performed to pinpoint possible error sources and recommendations were made for future study. The two-microphone approach evaluated in this study appears to have some advantages over other impedance measuring techniques.

Reproducibility of Dual-Microphone Voice Range Profile Equipment

ERIC Educational Resources Information Center

Printz, Trine; Pedersen, Ellen Raben; Juhl, Peter; Nielsen, Troels; Grøntved, Ågot Møller; Godballe, Christian

2017-01-01

Purpose: The aim of this study was to add further knowledge about the usefulness of the Voice Range Profile (VRP) assessment in clinical settings and research by analyzing VRP dual-microphone equipment precision, reliability, and room effect. Method: Test-retest studies were conducted in an anechoic chamber and an office: (a) comparing sound…

High Precision Method of Measuring the Velocity of Sound With Simple Apparatus

ERIC Educational Resources Information Center

Hansen, Russell C.

1975-01-01

A capacitor is discharged to generate an acoustical pulse and start a digital counter which is stopped by a microphone. Differences are accurately measured by positioning the sensing microphone on an optical bench. Results are discussed and extended experiments are suggested. The sources of some components are given. (GH)

40 CFR 201.26 - Procedures for the measurement on receiving property of retarder and car coupling noise.

Code of Federal Regulations, 2011 CFR

2011-07-01

... receiving property of retarder and car coupling noise. 201.26 Section 201.26 Protection of Environment ENVIRONMENTAL PROTECTION AGENCY (CONTINUED) NOISE ABATEMENT PROGRAMS NOISE EMISSION STANDARDS FOR TRANSPORTATION... receiving property of retarder and car coupling noise. (a) Retarders—(1) Microphone. The microphone must be...

Distance Learning Plan for the Defense Finance and Accounting Service (DFAS): A Study for the DBMU

DTIC Science & Technology

1994-09-01

according to the standard (H.261) motion video compression algorithm.24 n Schaphorst, Richard, notes presented at TELECON XIII, San Jose , California, 10...include automatic microphone mixing systems with one microphone for every two student seats, a large screen interactive computer display and the Socrates

78 FR 45272 - Certain Silicon Microphone Packages and Products Containing Same Institution of Investigation...

Federal Register 2010, 2011, 2012, 2013, 2014

2013-07-26

... INTERNATIONAL TRADE COMMISSION [Investigation No. 337-TA-888] Certain Silicon Microphone Packages.... International Trade Commission. ACTION: Notice. SUMMARY: Notice is hereby given that a complaint was filed with the U.S. International Trade Commission on June 21, 2013, under section 337 of the Tariff Act of 1930...

The $19.95 Solution to Large Group Telephone Interviews with Special Speakers.

ERIC Educational Resources Information Center

Robinson, George H.

1998-01-01

Describes an inexpensive solution for holding large-group telephone interviews, listing the equipment needed (record control, telephone, phone line with modular jack, portable amplifier with microphone-level input jack, audio cable with jack and plug compatible with the microphone input jack on the amplifier) and providing directions for setup.…

Phonocardiography with a smartphone

NASA Astrophysics Data System (ADS)

Thoms, Lars-Jochen; Colicchia, Giuseppe; Girwidz, Raimund

2017-03-01

When a stethoscope is placed on the chest over the heart, sounds coming from the heart can be directly heard. These sound vibrations can be captured through a microphone and the electrical signals from the transducer can be processed and plotted in a phonocardiogram. Students can easily use a microphone and smartphone to capture and analyse characteristic heart sounds.

System Measures Thermal Noise In A Microphone

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J.; Ngo, Kim Chi T.

1994-01-01

Vacuum provides acoustic isolation from environment. System for measuring thermal noise of microphone and its preamplifier eliminates some sources of error found in older systems. Includes isolation vessel and exterior suspension, acting together, enables measurement of thermal noise under realistic conditions while providing superior vibrational and accoustical isolation. System yields more accurate measurements of thermal noise.

46 CFR 121.510 - Recommended emergency broadcast instructions.

Code of Federal Regulations, 2010 CFR

2010-10-01

... Immediate Danger to Life or Property. (4) Say: “THIS IS (INSERT VESSEL'S NAME), (INSERT VESSEL'S NAME), (INSERT VESSEL'S NAME), (INSERT VESSEL'S CALL SIGN), OVER.” (5) Release the microphone button briefly and... 16 VHF and 2182 kHz on SSB are for emergency and calling purposes only.) (3) Press microphone button...

Phonocardiography with a Smartphone

ERIC Educational Resources Information Center

Thoms, Lars-Jochen; Colicchia, Giuseppe; Girwidz, Raimund

2017-01-01

When a stethoscope is placed on the chest over the heart, sounds coming from the heart can be directly heard. These sound vibrations can be captured through a microphone and the electrical signals from the transducer can be processed and plotted in a phonocardiogram. Students can easily use a microphone and smartphone to capture and analyse…

40 CFR 205.174 - Remedial orders.

Code of Federal Regulations, 2010 CFR

2010-07-01

... power (maximum rated RPM) is developed; and (B) Response characteristics such that, when closing RPM is...-state accuracy of within ±10% at 20 km/h (12.4 mph). (5) A microphone wind screen which does not affect... microphone wind screen must be used. The sound level meter must be calibrated with the acoustic calibrator as...

Final report of supplementary comparison AFRIMETS.AUV.A-S1: primary pressure calibration of LS2aP microphones according to IEC 61094-2, over the frequency range 1 Hz to 31.5 kHz.

NASA Astrophysics Data System (ADS)

Nel, R.; Barrera-Figueroa, S.; Dobrowolska, D.; Defilippo Soares, Z. M.; Maina, A. K.; Hof, C.

2016-01-01

This is the final report of the AFRIMETS.AUV-S1 comparison of the pressure sensitivity, modulus and phase, of LS2aP microphones in the frequency range 1 Hz to 31.5 kHz in accordance with IEC 61094-2. Six national metrology institutes from three different regional metrology organisations participated in the comparison for which two LS2aP microphones were circulated simultaneously to all the participants in a hybrid-star configuration. The comparison reference values were calculated as the weighted mean for modulus and phase for each individual microphone. Main text To reach the main text of this paper, click on Final Report. Note that this text is that which appears in Appendix B of the BIPM key comparison database kcdb.bipm.org/. The final report has been peer-reviewed and approved for publication by the CCAUV, according to the provisions of the CIPM Mutual Recognition Arrangement (CIPM MRA).

Estimation of aircraft angular coordinates using a directional-microphone array--An experimental study.

PubMed

Genescà, Meritxell; Svensson, U Peter; Taraldsen, Gunnar

2015-04-01

Ground reflections cause problems when estimating the direction of arrival of aircraft noise. In traditional methods, based on the time differences between the microphones of a compact array, they may cause a significant loss of accuracy in the vertical direction. This study evaluates the use of first-order directional microphones, instead of omnidirectional, with the aim of reducing the amplitude of the reflected sound. Such a modification allows the problem to be treated as in free field conditions. Although further tests are needed for a complete evaluation of the method, the experimental results presented here show that under the particular conditions tested the vertical angle error is reduced ∼10° for both jet and propeller aircraft by selecting an appropriate directivity pattern. It is also shown that the final level of error depends on the vertical angle of arrival of the sound, and that the estimates of the horizontal angle of arrival are not influenced by the directivity pattern of the microphones nor by the reflective properties of the ground.

Final report of key comparison AFRIMETS.AUV.A-K5: primary pressure calibration of LS1P microphones according to IEC 61094-2, over the frequency range 2 Hz to 10 kHz.

NASA Astrophysics Data System (ADS)

Nel, R.; Avison, J.; Harris, P.; Blabla, M.; Hämäläinen, J.

2017-01-01

The degrees of equivalence of the AFRIMETS.AUV.A-K5 regional key comparison are reported here as the final report. The scope of the comparison covered the complex pressure sensitivities of two LS1P microphones over the frequency range 2 Hz to 10 kHz in accordance with IEC 61094-2: 2009. Four national metrology institutes from two different regional metrology organisations participated in the comparison. Two LS1P microphones were circulated simultaneously to all the participants in a circular configuration. One of the microphones sensitivity shifted and all results associated with this microphone were subsequently excluded from further analysis and linking. The AFRIMETS.AUV.A-K5 comparison results were linked to the CCAUV.A-K5 comparison results via dual participation in the CCAUV.A-K5 and AFRIMETS.AUV.A-K5 comparisons. The degrees of equivalence, linked to the CCAUV.A-K5 comparison, were calculated for all participants of this comparison. Main text To reach the main text of this paper, click on Final Report. Note that this text is that which appears in Appendix B of the BIPM key comparison database kcdb.bipm.org/. The final report has been peer-reviewed and approved for publication by the CCAUV, according to the provisions of the CIPM Mutual Recognition Arrangement (CIPM MRA).

Characteristics of a new automatic hail recorder

NASA Astrophysics Data System (ADS)

Löffler-Mang, Martin; Schön, Dominik; Landry, Markus

2011-06-01

An automatic hail sensor was developed, based on signal production with microphones, a quick signal analysis and recording possibility. For this hail recorder (HARE) small piezo-electric microphones inside a Makrolon body are used to detect hailstones. The prototype has an octagonal shape, two microphones on the top and bottom plates situated in the middle of the device, and an electronic board. A hailstone striking the surface produces waves on the sensor body and a voltage in the piezo-electric microphones. Each hail event is stored in the internal memory including the time and date. The memory can be read out via a USB port at any time after one or more hail events. HARE was tested and calibrated with the help of a newly constructed pneumatic hail gun. The voltage signal at the top plate microphone of HARE increases linearly proportional to hailstone momentum, whereas at the bottom plate it increases linearly proportional to hailstone kinetic energy. For large hailstones the accuracy of HARE is in the order of 10%. Calibration of HARE is still in progress and it has not been tested in real hailfalls as yet. An online device as well as an autonomous one is available for a large number of possible applications. Lately there has been interest to use HARE at solar power plants in Southern Europe to prevent the expensive modules from becoming damaged. Perhaps HARE could also participate in new and existing hail observing networks.

The effects of digital signal processing features on children's speech recognition and loudness perception.

PubMed

Crukley, Jeffery; Scollie, Susan D

2014-03-01

The purpose of this study was to determine the effects of hearing instruments set to Desired Sensation Level version 5 (DSL v5) hearing instrument prescription algorithm targets and equipped with directional microphones and digital noise reduction (DNR) on children's sentence recognition in noise performance and loudness perception in a classroom environment. Ten children (ages 8-17 years) with stable, congenital sensorineural hearing losses participated in the study. Participants were fitted bilaterally with behind-the-ear hearing instruments set to DSL v5 prescriptive targets. Sentence recognition in noise was evaluated using the Bamford-Kowal-Bench Speech in Noise Test (Niquette et al., 2003). Loudness perception was evaluated using a modified version of the Contour Test of Loudness Perception (Cox, Alexander, Taylor, & Gray, 1997). Children's sentence recognition in noise performance was significantly better when using directional microphones alone or in combination with DNR than when using omnidirectional microphones alone or in combination with DNR. Children's loudness ratings for sounds above 72 dB SPL were lowest when fitted with the DSL v5 Noise prescription combined with directional microphones. DNR use showed no effect on loudness ratings. Use of the DSL v5 Noise prescription with a directional microphone improved sentence recognition in noise performance and reduced loudness perception ratings for loud sounds relative to a typical clinical reference fitting with the DSL v5 Quiet prescription with no digital signal processing features enabled. Potential clinical strategies are discussed.

An investigation of the usability of sound recognition for source separation of packaging wastes in reverse vending machines.

PubMed

Korucu, M Kemal; Kaplan, Özgür; Büyük, Osman; Güllü, M Kemal

2016-10-01

In this study, we investigate the usability of sound recognition for source separation of packaging wastes in reverse vending machines (RVMs). For this purpose, an experimental setup equipped with a sound recording mechanism was prepared. Packaging waste sounds generated by three physical impacts such as free falling, pneumatic hitting and hydraulic crushing were separately recorded using two different microphones. To classify the waste types and sizes based on sound features of the wastes, a support vector machine (SVM) and a hidden Markov model (HMM) based sound classification systems were developed. In the basic experimental setup in which only free falling impact type was considered, SVM and HMM systems provided 100% classification accuracy for both microphones. In the expanded experimental setup which includes all three impact types, material type classification accuracies were 96.5% for dynamic microphone and 97.7% for condenser microphone. When both the material type and the size of the wastes were classified, the accuracy was 88.6% for the microphones. The modeling studies indicated that hydraulic crushing impact type recordings were very noisy for an effective sound recognition application. In the detailed analysis of the recognition errors, it was observed that most of the errors occurred in the hitting impact type. According to the experimental results, it can be said that the proposed novel approach for the separation of packaging wastes could provide a high classification performance for RVMs. Copyright © 2016 Elsevier Ltd. All rights reserved.

Effect of type of noise and loudspeaker array on the performance of omnidirectional and directional microphones.

PubMed

Valente, Michael; Mispagel, Karen M; Tchorz, Juergen; Fabry, David

2006-06-01

Differences in performance between omnidirectional and directional microphones were evaluated between two loudspeaker conditions (single loudspeaker at 180 degrees; diffuse using eight loudspeakers set 45 degrees apart) and two types of noise (steady-state HINT noise; R-Space restaurant noise). Twenty-five participants were fit bilaterally with Phonak Perseo hearing aids using the manufacturer's recommended procedure. After wearing the hearing aids for one week, the parameters were fine-tuned based on subjective comments. Four weeks later, differences in performance between omnidirectional and directional microphones were assessed using HINT sentences presented at 0 degrees with the two types of background noise held constant at 65 dBA and under the two loudspeaker conditions. Results revealed significant differences in Reception Thresholds for Sentences (RTS in dB) where directional performance was significantly better than omnidirectional. Performance in the 180 degrees condition was significantly better than the diffuse condition, and performance was significantly better using the HINT noise in comparison to the R-Space restaurant noise. In addition, results revealed that within each loudspeaker array, performance was significantly better for the directional microphone. Looking across loudspeaker arrays, however, significant differences were not present in omnidirectional performance, but directional performance was significantly better in the 180 degrees condition when compared to the diffuse condition. These findings are discussed in terms of results reported in the past and counseling patients on the potential advantages of directional microphones as the listening situation and type of noise changes.

Influence of the middle ear anatomy on the performance of a membrane sensor in the incudostapedial joint gap.

PubMed

Koch, Martin; Seidler, Hannes; Hellmuth, Alexander; Bornitz, Matthias; Lasurashvili, Nikoloz; Zahnert, Thomas

2013-07-01

There is a great demand for implantable microphones for future generations of implantable hearing aids, especially Cochlea Implants. An implantable middle ear microphone based on a piezoelectric membrane sensor for insertion into the incudostapedial gap is investigated. The sensor is designed to measure the sound-induced forces acting on the center of the membrane. The sensor mechanically couples to the adjacent ossicles via two contact areas, the sensor membrane and the sensor housing. The sensing element is a piezoelectric single crystal bonded on a titanium membrane. The sensor allows a minimally invasive and reversible implantation without removal of ossicles and without additional sensor fixation in the tympanic cavity. This study investigates the implantable microphone sensor and its implantation concept. It intends to quantify the influence of the sensor's insertion position on the achievable microphone sensitivity. The investigation considers anatomical and pathological variations of the middle ear geometry and its space limitations. Temporal bone experiments on a laboratory model show that anatomical and pathological variations of the middle ear geometry can prevent the sensor from being placed optimally within the incudostapedial joint. Beyond scattering of transfer functions due to anatomic variations of individual middle ears there is the impact of variations in the sensor position within the ossicular chain that has a considerable effect on the transfer characteristics of the middle ear microphone. The centering of the sensor between incus and stapes, the direction of insertion (membrane to stapes or to incus) and the effect of additional contact points with surrounding anatomic structures affect the signal yield of the implanted sensor. The presence of additional contact points has a considerably impact on the sensitivity, yet the microphone sensitivity is quite robust against small changes in the positioning of the incus on the sensor. Signal losses can be avoided by adjusting the position of the sensor within the joint. The findings allow the development of an improved surgical insertion technique to ensure maximally achievable signal yield of the membrane sensor in the ISJ and provides valuable knowledge for a future design considerations including sensor miniaturization and geometry. Measurements of the implanted sensor in temporal bone specimens showed a microphone sensitivity in the order of 1 mV/Pa. This article is part of a special issue entitled "MEMRO 2012". Copyright © 2012 Elsevier B.V. All rights reserved.

On the use of mobile phones and wearable microphones for noise exposure measurements: Calibration and measurement accuracy

NASA Astrophysics Data System (ADS)

Dumoulin, Romain

Despite the fact that noise-induced hearing loss remains the number one occupational disease in developed countries, individual noise exposure levels are still rarely known and infrequently tracked. Indeed, efforts to standardize noise exposure levels present disadvantages such as costly instrumentation and difficulties associated with on site implementation. Given their advanced technical capabilities and widespread daily usage, mobile phones could be used to measure noise levels and make noise monitoring more accessible. However, the use of mobile phones for measuring noise exposure is currently limited due to the lack of formal procedures for their calibration and challenges regarding the measurement procedure. Our research investigated the calibration of mobile phone-based solutions for measuring noise exposure using a mobile phone's built-in microphones and wearable external microphones. The proposed calibration approach integrated corrections that took into account microphone placement error. The corrections were of two types: frequency-dependent, using a digital filter and noise level-dependent, based on the difference between the C-weighted noise level minus A-weighted noise level of the noise measured by the phone. The electro-acoustical limitations and measurement calibration procedure of the mobile phone were investigated. The study also sought to quantify the effect of noise exposure characteristics on the accuracy of calibrated mobile phone measurements. Measurements were carried out in reverberant and semi-anechoic chambers with several mobiles phone units of the same model, two types of external devices (an earpiece and a headset with an in-line microphone) and an acoustical test fixture (ATF). The proposed calibration approach significantly improved the accuracy of the noise level measurements in diffuse and free fields, with better results in the diffuse field and with ATF positions causing little or no acoustic shadowing. Several sources of errors and uncertainties were identified including the errors associated with the inter-unit-variability, the presence of signal saturation and the microphone placement relative to the source and the wearer. The results of the investigations and validation measurements led to recommendations regarding the measurement procedure including the use of external microphones having lower sensitivity and provided the basis for a standardized and unique factory default calibration method intended for implementation in any mobile phone. A user-defined adjustment was proposed to minimize the errors associated with calibration and the acoustical field. Mobile phones implementing the proposed laboratory calibration and used with external microphones showed great potential as noise exposure instruments. Combined with their potential as training and prevention tools, the expansion of their use could significantly help reduce the risks of noise-induced hearing loss.

Waveguide Calibrator for Multi-Element Probe Calibration

NASA Technical Reports Server (NTRS)

Sommerfeldt, Scott D.; Blotter, Jonathan D.

2007-01-01

A calibrator, referred to as the spider design, can be used to calibrate probes incorporating multiple acoustic sensing elements. The application is an acoustic energy density probe, although the calibrator can be used for other types of acoustic probes. The calibrator relies on the use of acoustic waveguide technology to produce the same acoustic field at each of the sensing elements. As a result, the sensing elements can be separated from each other, but still calibrated through use of the acoustic waveguides. Standard calibration techniques involve placement of an individual microphone into a small cavity with a known, uniform pressure to perform the calibration. If a cavity is manufactured with sufficient size to insert the energy density probe, it has been found that a uniform pressure field can only be created at very low frequencies, due to the size of the probe. The size of the energy density probe prevents one from having the same pressure at each microphone in a cavity, due to the wave effects. The "spider" design probe is effective in calibrating multiple microphones separated from each other. The spider design ensures that the same wave effects exist for each microphone, each with an indivdual sound path. The calibrator s speaker is mounted at one end of a 14-cm-long and 4.1-cm diameter small plane-wave tube. This length was chosen so that the first evanescent cross mode of the plane-wave tube would be attenuated by about 90 dB, thus leaving just the plane wave at the termination plane of the tube. The tube terminates with a small, acrylic plate with five holes placed symmetrically about the axis of the speaker. Four ports are included for the four microphones on the probe. The fifth port is included for the pre-calibrated reference microphone. The ports in the acrylic plate are in turn connected to the probe sensing elements via flexible PVC tubes. These five tubes are the same length, so the acoustic wave effects are the same in each tube. The flexible nature of the tubes allows them to be positioned so that each tube terminates at one of the microphones of the energy density probe, which is mounted in the acrylic structure, or the calibrated reference microphone. Tests performed verify that the pressure did not vary due to bends in the tubes. The results of these tests indicate that the average sound pressure level in the tubes varied by only 0.03 dB as the tubes were bent to various angles. The current calibrator design is effective up to a frequency of approximately 4.5 kHz. This upper design frequency is largely due to the diameter of the plane-wave tubes.

The effect of different cochlear implant microphones on acoustic hearing individuals’ binaural benefits for speech perception in noise

PubMed Central

Aronoff, Justin M.; Freed, Daniel J.; Fisher, Laurel M.; Pal, Ivan; Soli, Sigfrid D.

2011-01-01

Objectives Cochlear implant microphones differ in placement, frequency response, and other characteristics such as whether they are directional. Although normal hearing individuals are often used as controls in studies examining cochlear implant users’ binaural benefits, the considerable differences across cochlear implant microphones make such comparisons potentially misleading. The goal of this study was to examine binaural benefits for speech perception in noise for normal hearing individuals using stimuli processed by head-related transfer functions (HRTFs) based on the different cochlear implant microphones. Design HRTFs were created for different cochlear implant microphones and used to test participants on the Hearing in Noise Test. Experiment 1 tested cochlear implant users and normal hearing individuals with HRTF-processed stimuli and with sound field testing to determine whether the HRTFs adequately simulated sound field testing. Experiment 2 determined the measurement error and performance-intensity function for the Hearing in Noise Test with normal hearing individuals listening to stimuli processed with the various HRTFs. Experiment 3 compared normal hearing listeners’ performance across HRTFs to determine how the HRTFs affected performance. Experiment 4 evaluated binaural benefits for normal hearing listeners using the various HRTFs, including ones that were modified to investigate the contributions of interaural time and level cues. Results The results indicated that the HRTFs adequately simulated sound field testing for the Hearing in Noise Test. They also demonstrated that the test-retest reliability and performance-intensity function were consistent across HRTFs, and that the measurement error for the test was 1.3 dB, with a change in signal-to-noise ratio of 1 dB reflecting a 10% change in intelligibility. There were significant differences in performance when using the various HRTFs, with particularly good thresholds for the HRTF based on the directional microphone when the speech and masker were spatially separated, emphasizing the importance of measuring binaural benefits separately for each HRTF. Evaluation of binaural benefits indicated that binaural squelch and spatial release from masking were found for all HRTFs and binaural summation was found for all but one HRTF, although binaural summation was less robust than the other types of binaural benefits. Additionally, the results indicated that neither interaural time nor level cues dominated binaural benefits for the normal hearing participants. Conclusions This study provides a means to measure the degree to which cochlear implant microphones affect acoustic hearing with respect to speech perception in noise. It also provides measures that can be used to evaluate the independent contributions of interaural time and level cues. These measures provide tools that can aid researchers in understanding and improving binaural benefits in acoustic hearing individuals listening via cochlear implant microphones. PMID:21412155

Development of a Novel Bone Conduction Verification Tool Using a Surface Microphone: Validation With Percutaneous Bone Conduction Users.

PubMed

Hodgetts, William; Scott, Dylan; Maas, Patrick; Westover, Lindsey

2018-03-23

To determine if a newly-designed, forehead-mounted surface microphone would yield equivalent estimates of audibility when compared to audibility measured with a skull simulator for adult bone conduction users. Data was analyzed using a within subjects, repeated measures design. There were two different sensors (skull simulator and surface microphone) measuring the same hearing aid programmed to the same settings for all subjects. We were looking for equivalent results. Twenty-one adult percutaneous bone conduction users (12 females and 9 males) were recruited for this study. Mean age was 54.32 years with a standard deviation of 14.51 years. Nineteen of the subjects had conductive/mixed hearing loss and two had single-sided deafness. To define audibility, we needed to establish two things: (1) in situ-level thresholds at each audiometric frequency in force (skull simulator) and in sound pressure level (SPL; surface microphone). Next, we measured the responses of the preprogrammed test device in force on the skull simulator and in SPL on the surface mic in response to pink noise at three input levels: 55, 65, and 75 dB SPL. The skull simulator responses were converted to real head force responses by means of an individual real head to coupler difference transform. Subtracting the real head force level thresholds from the real head force output of the test aid yielded the audibility for each audiometric frequency for the skull simulator. Subtracting the SPL thresholds from the surface microphone from the SPL output of the test aid yielded the audibility for each audiometric frequency for the surface microphone. The surface microphone was removed and retested to establish the test-retest reliability of the tool. We ran a 2 (sensor) × 3 (input level) × 10 (frequency) mixed analysis of variance to determine if there were any significant main effects and interactions. There was a significant three-way interaction, so we proceeded to explore our planned comparisons. There were 90 planned comparisons of interest, three at each frequency (3 × 10) for the three input levels (30 × 3). Therefore, to minimize a type 1 error associated with multiple comparisons, we adjusted alpha using the Holm-Bonferroni method. There were five comparisons that yielded significant differences between the skull simulator and surface microphone (test and retest) in the estimation of audibility. However, the mean difference in these effects was small at 3.3 dB. Both sensors yielded equivalent results for the majority of comparisons. Models of bone conduction devices that have intact skin cannot be measured with the skull simulator. This study is the first to present and evaluate a new tool for bone conduction verification. The surface microphone is capable of yielding equivalent audibility measurements as the skull simulator for percutaneous bone conduction users at multiple input levels. This device holds potential for measuring other bone conduction devices (Sentio, BoneBridge, Attract, Soft headband devices) that do not have a percutaneous implant.

Effects of Orientation and Weatherproofing on the Detection of Bat Echolocation Calls

Treesearch

E. Britzke; B. Slack; M Armstrong; S. Loeb

2010-01-01

Ultrasonic detectors are powerful tools for the study of bat ecology. Many options are available for deploying acoustic detectors including various weatherproofing designs and microphone orientations, but the impacts of these options on the quantity and quality of the bat calls that are recorded are unknown. We compared the impacts of three microphone orientations (...

Measurement of Gravitational Acceleration Using a Computer Microphone Port

ERIC Educational Resources Information Center

Khairurrijal; Eko Widiatmoko; Srigutomo, Wahyu; Kurniasih, Neny

2012-01-01

A method has been developed to measure the swing period of a simple pendulum automatically. The pendulum position is converted into a signal frequency by employing a simple electronic circuit that detects the intensity of infrared light reflected by the pendulum. The signal produced by the electronic circuit is sent to the microphone port and…

Acoustic Measurement of Potato Cannon Velocity

ERIC Educational Resources Information Center

Courtney, Michael; Courtney, Amy

2007-01-01

Potato cannon velocity can be measured with a digitized microphone signal. A microphone is attached to the potato cannon muzzle, and a potato is fired at an aluminum target about 10 m away. Flight time can be determined from the acoustic waveform by subtracting the time in the barrel and time for sound to return from the target. The potato…

49 CFR 325.79 - Application of correction factors.

Code of Federal Regulations, 2014 CFR

2014-10-01

... microphone location point and the microphone target point is 60 feet (18.3 m) and that the measurement area... vehicle would be 87 dB(A), calculated as follows: 88 dB(A)Uncorrected average of readings −3 dB(A)Distance correction factor +2 dB(A)Ground surface correction factor _____ 87 dB(A)Corrected reading ...

49 CFR 325.79 - Application of correction factors.

Code of Federal Regulations, 2013 CFR

2013-10-01

... microphone location point and the microphone target point is 60 feet (18.3 m) and that the measurement area... vehicle would be 87 dB(A), calculated as follows: 88 dB(A)Uncorrected average of readings −3 dB(A)Distance correction factor +2 dB(A)Ground surface correction factor _____ 87 dB(A)Corrected reading ...

49 CFR 325.79 - Application of correction factors.

Code of Federal Regulations, 2012 CFR

2012-10-01

... microphone location point and the microphone target point is 60 feet (18.3 m) and that the measurement area... vehicle would be 87 dB(A), calculated as follows: 88 dB(A)Uncorrected average of readings −3 dB(A)Distance correction factor +2 dB(A)Ground surface correction factor _____ 87 dB(A)Corrected reading ...

Motherboards, Microphones and Metaphors: Re-Examining New Literacies and Black Feminist Thought through Technologies of Self

ERIC Educational Resources Information Center

Ellison, Tisha Lewis; Kirkland, David E.

2014-01-01

This article examines how two African American females composed counter-selves using a computer motherboard and a stand-alone microphone as critical identity texts. Situated within sociocultural and critical traditions in new literacy studies and black feminist thought, the authors extend conceptions of language, literacy and black femininity via…

Design of Small MEMS Microphone Array Systems for Direction Finding of Outdoors Moving Vehicles

PubMed Central

Zhang, Xin; Huang, Jingchang; Song, Enliang; Liu, Huawei; Li, Baoqing; Yuan, Xiaobing

2014-01-01

In this paper, a MEMS microphone array system scheme is proposed which implements real-time direction of arrival (DOA) estimation for moving vehicles. Wind noise is the primary source of unwanted noise on microphones outdoors. A multiple signal classification (MUSIC) algorithm is used in this paper for direction finding associated with spatial coherence to discriminate between the wind noise and the acoustic signals of a vehicle. The method is implemented in a SHARC DSP processor and the real-time estimated DOA is uploaded through Bluetooth or a UART module. Experimental results in different places show the validity of the system and the deviation is no bigger than 6° in the presence of wind noise. PMID:24603636

Design of small MEMS microphone array systems for direction finding of outdoors moving vehicles.

PubMed

Zhang, Xin; Huang, Jingchang; Song, Enliang; Liu, Huawei; Li, Baoqing; Yuan, Xiaobing

2014-03-05

In this paper, a MEMS microphone array system scheme is proposed which implements real-time direction of arrival (DOA) estimation for moving vehicles. Wind noise is the primary source of unwanted noise on microphones outdoors. A multiple signal classification (MUSIC) algorithm is used in this paper for direction finding associated with spatial coherence to discriminate between the wind noise and the acoustic signals of a vehicle. The method is implemented in a SHARC DSP processor and the real-time estimated DOA is uploaded through Bluetooth or a UART module. Experimental results in different places show the validity of the system and the deviation is no bigger than 6° in the presence of wind noise.

Human Action Recognition Using Wireless Wearable In-Ear Microphone

NASA Astrophysics Data System (ADS)

Nishimura, Jun; Kuroda, Tadahiro

To realize the ubiquitous eating habits monitoring, we proposed the use of sounds sensed by an in-ear placed wireless wearable microphone. A prototype of wireless wearable in-ear microphone was developed by utilizing a common Bluetooth headset. We proposed a robust chewing action recognition algorithm which consists of two recognition stages: “chew-like” signal detection and chewing sound verification stages. We also provide empirical results on other action recognition using in-ear sound including swallowing, cough, belch, and etc. The average chewing number counting error rate of 1.93% is achieved. Lastly, chewing sound mapping is proposed as a new prototypical approach to provide an additional intuitive feedback on food groups to be able to infer the eating habits in their daily life context.

Microphone Handling Noise: Measurements of Perceptual Threshold and Effects on Audio Quality

PubMed Central

Kendrick, Paul; Jackson, Iain R.; Fazenda, Bruno M.; Cox, Trevor J.; Li, Francis F.

2015-01-01

A psychoacoustic experiment was carried out to test the effects of microphone handling noise on perceived audio quality. Handling noise is a problem affecting both amateurs using their smartphones and cameras, as well as professionals using separate microphones and digital recorders. The noises used for the tests were measured from a variety of devices, including smartphones, laptops and handheld microphones. The signal features that characterise these noises are analysed and presented. The sounds include various types of transient, impact noises created by tapping or knocking devices, as well as more sustained sounds caused by rubbing. During the perceptual tests, listeners auditioned speech podcasts and were asked to rate the degradation of any unwanted sounds they heard. A representative design test methodology was developed that tried to encourage everyday rather than analytical listening. Signal-to-noise ratio (SNR) of the handling noise events was shown to be the best predictor of quality degradation. Other factors such as noise type or background noise in the listening environment did not significantly affect quality ratings. Podcast, microphone type and reproduction equipment were found to be significant but only to a small extent. A model allowing the prediction of degradation from the SNR is presented. The SNR threshold at which 50% of subjects noticed handling noise was found to be 4.2 ± 0.6 dBA. The results from this work are important for the understanding of our perception of impact sound and resonant noises in recordings, and will inform the future development of an automated predictor of quality for handling noise. PMID:26473498

Robust speaker's location detection in a vehicle environment using GMM models.

PubMed

Hu, Jwu-Sheng; Cheng, Chieh-Cheng; Liu, Wei-Han

2006-04-01

Abstract-Human-computer interaction (HCI) using speech communication is becoming increasingly important, especially in driving where safety is the primary concern. Knowing the speaker's location (i.e., speaker localization) not only improves the enhancement results of a corrupted signal, but also provides assistance to speaker identification. Since conventional speech localization algorithms suffer from the uncertainties of environmental complexity and noise, as well as from the microphone mismatch problem, they are frequently not robust in practice. Without a high reliability, the acceptance of speech-based HCI would never be realized. This work presents a novel speaker's location detection method and demonstrates high accuracy within a vehicle cabinet using a single linear microphone array. The proposed approach utilize Gaussian mixture models (GMM) to model the distributions of the phase differences among the microphones caused by the complex characteristic of room acoustic and microphone mismatch. The model can be applied both in near-field and far-field situations in a noisy environment. The individual Gaussian component of a GMM represents some general location-dependent but content and speaker-independent phase difference distributions. Moreover, the scheme performs well not only in nonline-of-sight cases, but also when the speakers are aligned toward the microphone array but at difference distances from it. This strong performance can be achieved by exploiting the fact that the phase difference distributions at different locations are distinguishable in the environment of a car. The experimental results also show that the proposed method outperforms the conventional multiple signal classification method (MUSIC) technique at various SNRs.

Chemical and explosive detections using photo-acoustic effect and quantum cascade lasers

NASA Astrophysics Data System (ADS)

Choa, Fow-Sen

2013-12-01

Photoacoustic (PA) effect is a sensitive spectroscopic technique for chemical sensing. In recent years, with the development of quantum cascade lasers (QCLs), significant progress has been achieved for PA sensing applications. Using high-power, tunable mid-IR QCLs as laser sources, PA chemical sensor systems have demonstrated parts-pertrillion- level detection sensitivity. Many of these high sensitivity measurements were demonstrated locally in PA cells. Recently, we have demonstrated standoff PA detection of isopropanol vapor for more than 41 feet distance using a quantum cascade laser and a microphone with acoustic reflectors. We also further demonstrated solid phase TNT detections at a standoff distance of 8 feet. To further calibrate the detection sensitivity, we use nerve gas simulants that were generated and calibrated by a commercial vapor generator. Standoff detection of gas samples with calibrated concentration of 2.3 ppm was achieved at a detection distance of more than 2 feet. An extended detection distance up to 14 feet was observed for a higher gas concentration of 13.9 ppm. For field operations, array of microphones and microphone-reflector pairs can be utilized to achieve noise rejection and signal enhancement. We have experimentally demonstrated that the signal and noise spectra of the 4 microphone/4 reflector system with a combined SNR of 12.48 dB. For the 16-microphone and one reflector case, an SNR of 17.82 was achieved. These successful chemical sensing demonstrations will likely create new demands for widely tunable QCLs with ultralow threshold (for local fire-alarm size detection systems) and high-power (for standoff detection systems) performances.

Acoustic Levitator Maintains Resonance

NASA Technical Reports Server (NTRS)

Barmatz, M. B.; Gaspar, M. S.

1986-01-01

Transducer loading characteristics allow resonance tracked at high temperature. Acoustic-levitation chamber length automatically adjusted to maintain resonance at constant acoustic frequency as temperature changes. Developed for containerless processing of materials at high temperatures, system does not rely on microphones as resonance sensors, since microphones are difficult to fabricate for use at temperatures above 500 degrees C. Instead, system uses acoustic transducer itself as sensor.

75 FR 39939 - Notice of Public Information Collection(s) Being Submitted for Review and Approval to the Office...

Federal Register 2010, 2011, 2012, 2013, 2014

2010-07-13

... forms of information technology; and (e) ways to further reduce the information collection burden for... Stations (Including Wireless Microphones). Form Number: N/A. Type of Review: Revision of a currently... Commission to clear the 700 MHz band of wireless microphones and provide them a home in the core TV spectrum...

Adaptive Noise Reduction Techniques for Airborne Acoustic Sensors

DTIC Science & Technology

2012-01-01

and Preamplifiers . . . . . . . . . . . . . . . . . . . . 16 3.3.2 Audio Recorders . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20 iv 4...consuming less energy than active systems such as radar, lidar, or sonar [5]. Ground and marine-based acoustic arrays are currently employed in a variety of...factors for the performance of an airborne acoustic array. 3.3.1 Audio Microphones and Preamplifiers An audio microphone is a transducer that converts

A New Kind of Laser Microphone for Photoacoustic Applications

DTIC Science & Technology

2008-12-01

1 A NEW KIND OF LASER MICROPHONE FOR PHOTOACOUSTIC APPLICATIONS Chen-Chia Wang, Sudhir Trivedi, and Feng Jin Brimrose ...NUMBER 5e. TASK NUMBER 5f. WORK UNIT NUMBER 7. PERFORMING ORGANIZATION NAME(S) AND ADDRESS(ES) Brimrose Corp. of America, 7720 Belair Road...laser microphone’s performance are also developed with preliminary experimental validation. ACKONWLEDGMENTS The authors from Brimrose

Dynamic Pressure Microphones

NASA Astrophysics Data System (ADS)

Werner, E.

In 1876, Alexander Graham Bell described his first telephone with a microphone using magnetic induction to convert the voice input into an electric output signal. The basic principle led to a variety of designs optimized for different needs, from hearing impaired users to singers or broadcast announcers. From the various sound pressure versions, only the moving coil design is still in mass production for speech and music application.

Direct Measurement of the Speed of Sound Using a Microphone and a Speaker

ERIC Educational Resources Information Center

Gómez-Tejedor, José A.; Castro-Palacio, Juan C.; Monsoriu, Juan A.

2014-01-01

We present a simple and accurate experiment to obtain the speed of sound in air using a conventional speaker and a microphone connected to a computer. A free open source digital audio editor and recording computer software application allows determination of the time-of-flight of the wave for different distances, from which the speed of sound is…

The frequency range of TMJ sounds.

PubMed

Widmalm, S E; Williams, W J; Djurdjanovic, D; McKay, D C

2003-04-01

There are conflicting opinions about the frequency range of temporomandibular joint (TMJ) sounds. Some authors claim that the upper limit is about 650 Hz. The aim was to test the hypothesis that TMJ sounds may contain frequencies well above 650 Hz but that significant amounts of their energy are lost if the vibrations are recorded using contact sensors and/or travel far through the head tissues. Time-frequency distributions of 172 TMJ clickings (three subjects) were compared between recordings with one microphone in the ear canal and a skin contact transducer above the clicking joint and between recordings from two microphones, one in each ear canal. The energy peaks of the clickings recorded with a microphone in the ear canal on the clicking side were often well above 650 Hz and always in a significantly higher area (range 117-1922 Hz, P < 0.05 or lower) than in recordings obtained with contact sensors (range 47-375 Hz) or in microphone recordings from the opposite ear canal (range 141-703 Hz). Future studies are required to establish normative frequency range values of TMJ sounds but need methods also capable of recording the high frequency vibrations.

Application of the remote microphone method to active noise control in a mobile phone.

PubMed

Cheer, Jordan; Elliott, Stephen J; Oh, Eunmi; Jeong, Jonghoon

2018-04-01

Mobile phones are used in a variety of situations where environmental noise may interfere with the ability of the near-end user to communicate with the far-end user. To overcome this problem, it might be possible to use active noise control technology to reduce the noise experienced by the near-end user. This paper initially demonstrates that when an active noise control system is used in a practical mobile phone configuration to minimise the noise measured by an error microphone mounted on the mobile phone, the attenuation achieved at the user's ear depends strongly on the position of the source generating the acoustic interference. To help overcome this problem, a remote microphone processing strategy is investigated that estimates the pressure at the user's ear from the pressure measured by the microphone on the mobile phone. Through an experimental implementation, it is demonstrated that this arrangement achieves a significant improvement in the attenuation measured at the ear of the user, compared to the standard active control strategy. The robustness of the active control system to changes in both the interfering sound field and the position of the mobile device relative to the ear of the user is also investigated experimentally.

Wind noise measured at the ground surface.

PubMed

Yu, Jiao; Raspet, Richard; Webster, Jeremy; Abbott, Johnpaul

2011-02-01

Measurements of the wind noise measured at the ground surface outdoors are analyzed using the mirror flow model of anisotropic turbulence by Kraichnan [J. Acoust. Soc. Am. 28(3), 378-390 (1956)]. Predictions of the resulting behavior of the turbulence spectrum with height are developed, as well as predictions of the turbulence-shear interaction pressure at the surface for different wind velocity profiles and microphone mounting geometries are developed. The theoretical results of the behavior of the velocity spectra with height are compared to measurements to demonstrate the applicability of the mirror flow model to outdoor turbulence. The use of a logarithmic wind velocity profile for analysis is tested using meteorological models for wind velocity profiles under different stability conditions. Next, calculations of the turbulence-shear interaction pressure are compared to flush microphone measurements at the surface and microphone measurements with a foam covering flush with the surface. The measurements underneath the thin layers of foam agree closely with the predictions, indicating that the turbulence-shear interaction pressure is the dominant source of wind noise at the surface. The flush microphones measurements are intermittently larger than the predictions which may indicate other contributions not accounted for by the turbulence-shear interaction pressure.

The effects of motion artifact on mechanomyography: A comparative study of microphones and accelerometers.

PubMed

Posatskiy, A O; Chau, T

2012-04-01

Mechanomyography (MMG) is an important kinesiological tool and potential communication pathway for individuals with disabilities. However, MMG is highly susceptible to contamination by motion artifact due to limb movement. A better understanding of the nature of this contamination and its effects on different sensing methods is required to inform robust MMG sensor design. Therefore, in this study, we recorded MMG from the extensor carpi ulnaris of six able-bodied participants using three different co-located condenser microphone and accelerometer pairings. Contractions at 30% MVC were recorded with and without a shaker-induced single-frequency forearm motion artifact delivered via a custom test rig. Using a signal-to-signal-plus-noise-ratio and the adaptive Neyman curve-based statistic, we found that microphone-derived MMG spectra were significantly less influenced by motion artifact than corresponding accelerometer-derived spectra (p⩽0.05). However, non-vanishing motion artifact harmonics were present in both spectra, suggesting that simple bandpass filtering may not remove artifact influences permeating into typical MMG bands of interest. Our results suggest that condenser microphones are preferred for MMG recordings when the mitigation of motion artifact effects is important. Copyright © 2011. Published by Elsevier Ltd.

High channel count microphone array accurately and precisely localizes ultrasonic signals from freely-moving mice.

PubMed

Warren, Megan R; Sangiamo, Daniel T; Neunuebel, Joshua P

2018-03-01

An integral component in the assessment of vocal behavior in groups of freely interacting animals is the ability to determine which animal is producing each vocal signal. This process is facilitated by using microphone arrays with multiple channels. Here, we made important refinements to a state-of-the-art microphone array based system used to localize vocal signals produced by freely interacting laboratory mice. Key changes to the system included increasing the number of microphones as well as refining the methodology for localizing and assigning vocal signals to individual mice. We systematically demonstrate that the improvements in the methodology for localizing mouse vocal signals led to an increase in the number of signals detected as well as the number of signals accurately assigned to an animal. These changes facilitated the acquisition of larger and more comprehensive data sets that better represent the vocal activity within an experiment. Furthermore, this system will allow more thorough analyses of the role that vocal signals play in social communication. We expect that such advances will broaden our understanding of social communication deficits in mouse models of neurological disorders. Copyright © 2018 Elsevier B.V. All rights reserved.

Implementation of a Virtual Microphone Array to Obtain High Resolution Acoustic Images

PubMed Central

Izquierdo, Alberto; Suárez, Luis; Suárez, David

2017-01-01

Using arrays with digital MEMS (Micro-Electro-Mechanical System) microphones and FPGA-based (Field Programmable Gate Array) acquisition/processing systems allows building systems with hundreds of sensors at a reduced cost. The problem arises when systems with thousands of sensors are needed. This work analyzes the implementation and performance of a virtual array with 6400 (80 × 80) MEMS microphones. This virtual array is implemented by changing the position of a physical array of 64 (8 × 8) microphones in a grid with 10 × 10 positions, using a 2D positioning system. This virtual array obtains an array spatial aperture of 1 × 1 m2. Based on the SODAR (SOund Detection And Ranging) principle, the measured beampattern and the focusing capacity of the virtual array have been analyzed, since beamforming algorithms assume to be working with spherical waves, due to the large dimensions of the array in comparison with the distance between the target (a mannequin) and the array. Finally, the acoustic images of the mannequin, obtained for different frequency and range values, have been obtained, showing high angular resolutions and the possibility to identify different parts of the body of the mannequin. PMID:29295485

A unified acquisition system for acoustic data

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J.; Holmes, H. K.

1977-01-01

A multichannel, acoustic AM carrier system was developed for a wide variety of applications, particularly for aircraft noise and sonic boom measurements. Each data acquisition channel consists of a condenser microphone, an acoustic signal converter, and a Zero Drive amplifier, along with peripheral supporting equipment. A control network insures continuous optimal tuning of the converter and permits remote calibration of the condenser microphone. With a 12.70-mm (1/2-in.) condenser microphone, the converter/Zero Drive amplifier combination has a frequency response from 0 Hz to 20 kHz (-3 db), a dynamic range exceeding 70 db, and a minimum noise floor of 50 db ref. 20 micro Pa) in the band 22.4 Hz to 22.4 kHz. The system requires no external impedance matching networks and is insensitive to cable length, at least up to 900 m (3,000 ft). System gain varies only + or - 1 db over the temperature range 4 to 54 C (40 to 130 F). Adapters are available to accommodate 23.77-mm (1-in.) and 6.35-mm (1/4-in.) microphones and to provide 30-db attenuation. A field test to obtain the acoustical time history of a helicopter flyover proved successful.

Compact Infrasonic Windscreen

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J.; Shams, Qamar A.; Sealey, Bradley S.; Comeaux, Toby

2005-01-01

A compact windscreen has been conceived for a microphone of a type used outdoors to detect atmospheric infrasound from a variety of natural and manmade sources. Wind at the microphone site contaminates received infrasonic signals (defined here as sounds having frequencies <20 Hz), because a microphone cannot distinguish between infrasonic pressures (which propagate at the speed of sound) and convective pressure fluctuations generated by wind turbulence. Hence, success in measurement of outdoor infrasound depends on effective screening of the microphone from the wind. The present compact windscreen is based on a principle: that infrasound at sufficiently large wavelength can penetrate any barrier of practical thickness. Thus, a windscreen having solid, non-porous walls can block convected pressure fluctuations from the wind while transmitting infrasonic acoustic waves. The transmission coefficient depends strongly upon the ratio between the acoustic impedance of the windscreen and that of air. Several materials have been found to have impedance ratios that render them suitable for use in constructing walls that have practical thicknesses and are capable of high transmission of infrasound. These materials (with their impedance ratios in parentheses) are polyurethane foam (222), space shuttle tile material (332), balsa (323), cedar (3,151), and pine (4,713).

Method for determining artillery position

NASA Technical Reports Server (NTRS)

Fischer, Johannes; Loges, Werner; Meuser, Wilfried

1988-01-01

A method is disclosed for determining the position of cannon from measurement sites whose distance from each other lies in the same order of magnitude as the distance between the cannons -- that distance being in the kilometer range -- with the help of the travel time evaluation of muzzle blasts received at the measurement sites. There are at least two measurement sites, consisting of a cruciform of four microphones each positioned so that one axis is oriented to an arbitrarily chosen reference direction with the microphones spaced closely together. In this arrangement of diametrically opposed microphones, the respective travel times are determined and placed in a relationship whose arctangent is a radio bearing to the reference direction in which radio bearings are determined with consideration of their position and their opposing distance from the cannon position.

Laboratory exercises on oscillation modes of pipes

NASA Astrophysics Data System (ADS)

Haeberli, Willy

2009-03-01

This paper describes an improved lab setup to study the vibrations of air columns in pipes. Features of the setup include transparent pipes which reveal the position of a movable microphone inside the pipe; excitation of pipe modes with a miniature microphone placed to allow access to the microphone stem for open, closed, or conical pipes; and sound insulation to avoid interference between different setups in a student lab. The suggested experiments on the modes of open, closed, and conical pipes, the transient response of a pipe, and the effect of pipe diameter are suitable for introductory physics laboratories, including laboratories for nonscience majors and music students, and for more advanced undergraduate laboratories. For honors students or for advanced laboratory exercises, the quantitative relation between the resonance width and damping time constant is of interest.

Measurement of Model Noise in a Hard-Wall Wind Tunnel

NASA Technical Reports Server (NTRS)

Soderman, Paul T.

2006-01-01

Identification, analysis, and control of fluid-mechanically-generated sound from models of aircraft and automobiles in special low-noise, semi-anechoic wind tunnels are an important research endeavor. Such studies can also be done in aerodynamic wind tunnels that have hard walls if phased microphone arrays are used to focus on the noise-source regions and reject unwanted reflections or background noise. Although it may be difficult to simulate the total flyover or drive-by noise in a closed wind tunnel, individual noise sources can be isolated and analyzed. An acoustic and aerodynamic study was made of a 7-percent-scale aircraft model in a NASA Ames 7-by-10-ft (about 2-by-3-m) wind tunnel for the purpose of identifying and attenuating airframe noise sources. Simulated landing, takeoff, and approach configurations were evaluated at Mach 0.26. Using a phased microphone array mounted in the ceiling over the inverted model, various noise sources in the high-lift system, landing gear, fins, and miscellaneous other components were located and compared for sound level and frequency at one flyover location. Numerous noise-alleviation devices and modifications of the model were evaluated. Simultaneously with acoustic measurements, aerodynamic forces were recorded to document aircraft conditions and any performance changes caused by geometric modifications. Most modern microphone-array systems function in the frequency domain in the sense that spectra of the microphone outputs are computed, then operations are performed on the matrices of microphone-signal cross-spectra. The entire acoustic field at one station in such a system is acquired quickly and interrogated during postprocessing. Beam-forming algorithms are employed to scan a plane near the model surface and locate noise sources while rejecting most background noise and spurious reflections. In the case of the system used in this study, previous studies in the wind tunnel have identified noise sources up to 19 dB below the normal background noise of the wind tunnel. Theoretical predictions of array performance are used to minimize the width and the side lobes of the beam pattern of the microphone array for a given test arrangement. To capture flyover noise of the inverted model, a 104-element microphone array in a 622-mm-diameter cluster was installed in a 19-mm-thick poly(methyl methacrylate) plate in the ceiling of the test section of the wind tunnel above the aircraft model (see Figure 1). The microphones were of the condenser type, and their diaphragms were mounted flush in the array plate, which was recessed 12.7 mm into the ceiling and covered by a porous aromatic polyamide cloth (not shown in the figure) to minimize boundary-layer noise. This design caused the level of flow noise to be much less than that of flush-mount designs. The drawback of this design was that the cloth attenuated sound somewhat and created acoustic resonances that could grow to several dB at a frequency of 10 kHz.

Parallel Processing of Large Scale Microphone Arrays for Sound Capture

NASA Astrophysics Data System (ADS)

Jan, Ea-Ee.

1995-01-01

Performance of microphone sound pick up is degraded by deleterious properties of the acoustic environment, such as multipath distortion (reverberation) and ambient noise. The degradation becomes more prominent in a teleconferencing environment in which the microphone is positioned far away from the speaker. Besides, the ideal teleconference should feel as easy and natural as face-to-face communication with another person. This suggests hands-free sound capture with no tether or encumbrance by hand-held or body-worn sound equipment. Microphone arrays for this application represent an appropriate approach. This research develops new microphone array and signal processing techniques for high quality hands-free sound capture in noisy, reverberant enclosures. The new techniques combine matched-filtering of individual sensors and parallel processing to provide acute spatial volume selectivity which is capable of mitigating the deleterious effects of noise interference and multipath distortion. The new method outperforms traditional delay-and-sum beamformers which provide only directional spatial selectivity. The research additionally explores truncated matched-filtering and random distribution of transducers to reduce complexity and improve sound capture quality. All designs are first established by computer simulation of array performance in reverberant enclosures. The simulation is achieved by a room model which can efficiently calculate the acoustic multipath in a rectangular enclosure up to a prescribed order of images. It also calculates the incident angle of the arriving signal. Experimental arrays were constructed and their performance was measured in real rooms. Real room data were collected in a hard-walled laboratory and a controllable variable acoustics enclosure of similar size, approximately 6 x 6 x 3 m. An extensive speech database was also collected in these two enclosures for future research on microphone arrays. The simulation results are shown to be consistent with the real room data. Localization of sound sources has been explored using cross-power spectrum time delay estimation and has been evaluated using real room data under slightly, moderately and highly reverberant conditions. To improve the accuracy and reliability of the source localization, an outlier detector that removes incorrect time delay estimation has been invented. To provide speaker selectivity for microphone array systems, a hands-free speaker identification system has been studied. A recently invented feature using selected spectrum information outperforms traditional recognition methods. Measured results demonstrate the capabilities of speaker selectivity from a matched-filtered array. In addition, simulation utilities, including matched -filtering processing of the array and hands-free speaker identification, have been implemented on the massively -parallel nCube super-computer. This parallel computation highlights the requirements for real-time processing of array signals.

Development of a Microphone Phased Array Capability for the Langley 14- by 22-Foot Subsonic Tunnel

NASA Technical Reports Server (NTRS)

Humphreys, William M.; Brooks, Thomas F.; Bahr, Christopher J.; Spalt, Taylor B.; Bartram, Scott M.; Culliton, William G.; Becker, Lawrence E.

2014-01-01

A new aeroacoustic measurement capability has been developed for use in open-jet testing in the NASA Langley 14- by 22-Foot Subsonic Tunnel (14x22 tunnel). A suite of instruments has been developed to characterize noise source strengths, locations, and directivity for both semi-span and full-span test articles in the facility. The primary instrument of the suite is a fully traversable microphone phased array for identification of noise source locations and strengths on models. The array can be mounted in the ceiling or on either side of the facility test section to accommodate various test article configurations. Complementing the phased array is an ensemble of streamwise traversing microphones that can be placed around the test section at defined locations to conduct noise source directivity studies along both flyover and sideline axes. A customized data acquisition system has been developed for the instrumentation suite that allows for command and control of all aspects of the array and microphone hardware, and is coupled with a comprehensive data reduction system to generate information in near real time. This information includes such items as time histories and spectral data for individual microphones and groups of microphones, contour presentations of noise source locations and strengths, and hemispherical directivity data. The data acquisition system integrates with the 14x22 tunnel data system to allow real time capture of facility parameters during acquisition of microphone data. The design of the phased array system has been vetted via a theoretical performance analysis based on conventional monopole beamforming and DAMAS deconvolution. The performance analysis provides the ability to compute figures of merit for the array as well as characterize factors such as beamwidths, sidelobe levels, and source discrimination for the types of noise sources anticipated in the 14x22 tunnel. The full paper will summarize in detail the design of the instrumentation suite, the construction of the hardware system, and the results of the performance analysis. Although the instrumentation suite is designed to characterize noise for a variety of test articles in the 14x22 tunnel, this paper will concentrate on description of the instruments for two specific test campaigns in the facility, namely a full-span NASA Hybrid Wing Body (HWB) model entry and a semi-span Gulfstream aircraft model entry, tested in the facility in the winter of 2012 and spring of 2013, respectively.

Study of the Microphonics for Prospective Space-Based Neutron and Gamma-Ray Detectors and Methods for its Suppression

NASA Astrophysics Data System (ADS)

Vostrukhin, A. A.; Golovin, D. V.; Kozyrev, A. S.; Litvak, M. L.; Malakhov, A. V.; Mitrofanov, I. G.; Mokrousov, M. I.; Tomilina, T. M.; Bobrovnitskiy, Yu. I.; Grebennikov, A. S.; Laktionova, M. M.; Bakhtin, B. N.; Sotov, A. V.

2018-05-01

The results of testing a number of space-based detectors that contain PMTs or high-voltage electrodes for the noise from the microphonics that occurs in the signal path due to external mechanical action have been presented. A method for the vibration isolation of instruments aboard a spacecraft has been proposed to reduce their responsivity to vibrations.

Detection of Humans and Light Vehicles Using Acoustic-to-Seismic Coupling

DTIC Science & Technology

2009-08-31

microphones, video cameras (regular and infrared), magnetic sensors, and active Doppler radar and sonar systems. These sensors could be located at... sonar systems due to dramatic absorption/reflection of electromagnetic/ultrasonic waves [8,9]. 6...engine was turned off, and the car continued moving. This eliminated the engine sound. A PCB microphone, 377B41, with preamplifier , 426A30, and with

Toward Active Control of Noise from Hot Supersonic Jets

DTIC Science & Technology

2013-02-15

measurements were obtained to allow analysis of some of these issues. This data, acquired in July 2012, includes 12 B& K microphones on a far-field arc...t i« /^ \\ k 100 J\\l ^^~ 95 ^^ JV^Ä - 5.5, Tjd/T...mounted Kulite, the PCB transducer, and a calibration traceable B& K 1/4-inch microphone will be place as equivalent (x, r) positions separated by

49 CFR 325.73 - Microphone distance correction factors. 1

Code of Federal Regulations, 2013 CFR

2013-10-01

... the observed sound level reading is— 31 feet (9.5 m) or more but less than 35 feet (10.7 m) −4 35 feet... more but less than 83 feet (25.3 m) +2 [40 FR 42437, Sept. 12, 1975, as amended at 54 FR 50385, Dec. 6... 49 Transportation 5 2013-10-01 2013-10-01 false Microphone distance correction factors. 1 325.73...

49 CFR 325.73 - Microphone distance correction factors. 1

Code of Federal Regulations, 2012 CFR

2012-10-01

... the observed sound level reading is— 31 feet (9.5 m) or more but less than 35 feet (10.7 m) −4 35 feet... more but less than 83 feet (25.3 m) +2 [40 FR 42437, Sept. 12, 1975, as amended at 54 FR 50385, Dec. 6... 49 Transportation 5 2012-10-01 2012-10-01 false Microphone distance correction factors. 1 325.73...

49 CFR 325.73 - Microphone distance correction factors. 1

Code of Federal Regulations, 2014 CFR

2014-10-01

... the observed sound level reading is— 31 feet (9.5 m) or more but less than 35 feet (10.7 m) −4 35 feet... more but less than 83 feet (25.3 m) +2 [40 FR 42437, Sept. 12, 1975, as amended at 54 FR 50385, Dec. 6... 49 Transportation 5 2014-10-01 2014-10-01 false Microphone distance correction factors. 1 325.73...

Application of a New Infrasound Sensor Technology in a Long Range Infrasound Propagation Experiment

NASA Astrophysics Data System (ADS)

Talmadge, C. L.; Waxler, R.; Hetzer, C. H.; Kleniert, D. E., Jr.; Dillion, K.; Assink, J.; Aydin, A.

2009-12-01

A low-cost ruggedized infrasound sensor has been developed at the NCPA laboratory of the University of Mississippi for outdoor infrasound measurements. This sensor has similar performance characteristics to other "standard" infrasound sensors, such as the Chaparral 50. A total of 50 sensors were constructed for this experiment, of which 42 were deployed on the Nevada and Utah desert for a period of four months. A long-range infrasound propagation experiment using these sensors was performed during the summer and fall of 2009. Source sizes varied in size from 4, 20 and 80 equivalent tons of TNT. The blasts were carried out typically on the Monday of each week in the afternoon, and were part of a scheduled demolition of first, second and third stages of trident missiles. In addition to a source capture location 23-km south of the site of the blasts, a series of 8 5-element arrays are located to the west of the blast location, at approximate ranges of 180 through 250 km in 10-km steps. Each array consisted of elements at -150-m, -50-m, 0-m, 50-m and 150-m relative to the center of the array along an east-west direction, and all microphones were equipped with 4 50-ft porous hoses connected to the microphone manifold for wind noise suppression. The signals from the microphones were digitized using GPS-synchronized, 24-bit DAQ systems. A Westerly direction for the deployment of the microphones was motivated by the presence of a strong stratospheric duct that persists through the summer months in the northern hemisphere at these latitudes. In this paper, we will discuss feasibility issues related the design of the NCPA microphone that makes possible deployments on these on large scales. Signal to noise issues related to temperature and wind fluctuations will also be discussed. Future plans include a larger scale deployment of several hundred microphones during 2010. We will discuss how the lessons learned from this series of measurements impacts that future deployment.

Evaluation of speech reception threshold in noise in young Cochlear™ Nucleus® system 6 implant recipients using two different digital remote microphone technologies and a speech enhancement sound processing algorithm.

PubMed

Razza, Sergio; Zaccone, Monica; Meli, Aannalisa; Cristofari, Eliana

2017-12-01

Children affected by hearing loss can experience difficulties in challenging and noisy environments even when deafness is corrected by Cochlear implant (CI) devices. These patients have a selective attention deficit in multiple listening conditions. At present, the most effective ways to improve the performance of speech recognition in noise consists of providing CI processors with noise reduction algorithms and of providing patients with bilateral CIs. The aim of this study was to compare speech performances in noise, across increasing noise levels, in CI recipients using two kinds of wireless remote-microphone radio systems that use digital radio frequency transmission: the Roger Inspiro accessory and the Cochlear Wireless Mini Microphone accessory. Eleven Nucleus Cochlear CP910 CI young user subjects were studied. The signal/noise ratio, at a speech reception threshold (SRT) value of 50%, was measured in different conditions for each patient: with CI only, with the Roger or with the MiniMic accessory. The effect of the application of the SNR-noise reduction algorithm in each of these conditions was also assessed. The tests were performed with the subject positioned in front of the main speaker, at a distance of 2.5 m. Another two speakers were positioned at 3.50 m. The main speaker at 65 dB issued disyllabic words. Babble noise signal was delivered through the other speakers, with variable intensity. The use of both wireless remote microphones improved the SRT results. Both systems improved gain of speech performances. The gain was higher with the Mini Mic system (SRT = -4.76) than the Roger system (SRT = -3.01). The addition of the NR algorithm did not statistically further improve the results. There is significant improvement in speech recognition results with both wireless digital remote microphone accessories, in particular with the Mini Mic system when used with the CP910 processor. The use of a remote microphone accessory surpasses the benefit of application of NR algorithm. Copyright © 2017. Published by Elsevier B.V.

Conceptual Sound System Design for Clifford Odets' "GOLDEN BOY"

NASA Astrophysics Data System (ADS)

Yang, Yen Chun

There are two different aspects in the process of sound design, "Arts" and "Science". In my opinion, the sound design should engage both aspects strongly and in interaction with each other. I started the process of designing the sound for GOLDEN BOY by building the city soundscape of New York City in 1937. The scenic design for this piece is designed in the round, putting the audience all around the stage; this gave me a great opportunity to use surround and specialization techniques to transform the space into a different sonic world. My specialization design is composed of two subsystems -- one is the four (4) speakers center cluster diffusing towards the four (4) sections of audience, and the other is the four (4) speakers on the four (4) corners of the theatre. The outside ring provides rich sound source localization and the inside ring provides more support for control of the specialization details. In my design four (4) lavalier microphones are hung under the center iron cage from the four (4) corners of the stage. Each microphone is ten (10) feet above the stage. The signal for each microphone is sent to the two (2) center speakers in the cluster diagonally opposite the microphone. With the appropriate level adjustment of the microphones, the audience will not notice the amplification of the voices; however, through my specialization system, the presence and location of the voices of all actors are preserved for all audiences clearly. With such vocal reinforcements provided by the microphones, I no longer need to worry about overwhelming the dialogue on stage by the underscoring. A successful sound system design should not only provide a functional system, but also take the responsibility of bringing actors' voices to the audience and engaging the audience with the world that we create on stage. By designing a system which reinforces the actors' voices while at the same time providing control over localization of movement of sound effects, I was able not only to make the text present and clear for the audiences, but also to support the storyline strongly through my composed music, environmental soundscapes, and underscoring.

A dynamic multi-channel speech enhancement system for distributed microphones in a car environment

NASA Astrophysics Data System (ADS)

Matheja, Timo; Buck, Markus; Fingscheidt, Tim

2013-12-01

Supporting multiple active speakers in automotive hands-free or speech dialog applications is an interesting issue not least due to comfort reasons. Therefore, a multi-channel system for enhancement of speech signals captured by distributed distant microphones in a car environment is presented. Each of the potential speakers in the car has a dedicated directional microphone close to his position that captures the corresponding speech signal. The aim of the resulting overall system is twofold: On the one hand, a combination of an arbitrary pre-defined subset of speakers' signals can be performed, e.g., to create an output signal in a hands-free telephone conference call for a far-end communication partner. On the other hand, annoying cross-talk components from interfering sound sources occurring in multiple different mixed output signals are to be eliminated, motivated by the possibility of other hands-free applications being active in parallel. The system includes several signal processing stages. A dedicated signal processing block for interfering speaker cancellation attenuates the cross-talk components of undesired speech. Further signal enhancement comprises the reduction of residual cross-talk and background noise. Subsequently, a dynamic signal combination stage merges the processed single-microphone signals to obtain appropriate mixed signals at the system output that may be passed to applications such as telephony or a speech dialog system. Based on signal power ratios between the particular microphone signals, an appropriate speaker activity detection and therewith a robust control mechanism of the whole system is presented. The proposed system may be dynamically configured and has been evaluated for a car setup with four speakers sitting in the car cabin disturbed in various noise conditions.

Factors affecting the performance of large-aperture microphone arrays.

PubMed

Silverman, Harvey F; Patterson, William R; Sachar, Joshua

2002-05-01

Large arrays of microphones have been proposed and studied as a possible means of acquiring data in offices, conference rooms, and auditoria without requiring close-talking microphones. When such an array essentially surrounds all possible sources, it is said to have a large aperture. Large-aperture arrays have attractive properties of spatial resolution and signal-to-noise enhancement. This paper presents a careful comparison of theoretical and measured performance for an array of 256 microphones using simple delay-and-sum beamforming. This is the largest currently functional, all digital-signal-processing array that we know of. The array is wall-mounted in the moderately adverse environment of a general-purpose laboratory (8 m x 8 m x 3 m). The room has a T60 reverberation time of 550 ms. Reverberation effects in this room severely impact the array's performance. However, the width of the main lobe remains comparable to that of a simplified prediction. Broadband spatial resolution shows a single central peak with 10 dB gain about 0.4 m in diameter at the -3 dB level. Away from that peak, the response is approximately flat over most of the room. Optimal weighting for signal-to-noise enhancement degrades the spatial resolution minimally. Experimentally, we verify that signal-to-noise gain is less than proportional to the square root of the number of microphones probably due to the partial correlation of the noise between channels, to variation of signal intensity with polar angle about the source, and to imperfect correlation of the signal over the array caused by reverberations. We show measurements of the relative importance of each effect in our environment.

Factors affecting the performance of large-aperture microphone arrays

NASA Astrophysics Data System (ADS)

Silverman, Harvey F.; Patterson, William R.; Sachar, Joshua

2002-05-01

Large arrays of microphones have been proposed and studied as a possible means of acquiring data in offices, conference rooms, and auditoria without requiring close-talking microphones. When such an array essentially surrounds all possible sources, it is said to have a large aperture. Large-aperture arrays have attractive properties of spatial resolution and signal-to-noise enhancement. This paper presents a careful comparison of theoretical and measured performance for an array of 256 microphones using simple delay-and-sum beamforming. This is the largest currently functional, all digital-signal-processing array that we know of. The array is wall-mounted in the moderately adverse environment of a general-purpose laboratory (8 m×8 m×3 m). The room has a T60 reverberation time of 550 ms. Reverberation effects in this room severely impact the array's performance. However, the width of the main lobe remains comparable to that of a simplified prediction. Broadband spatial resolution shows a single central peak with 10 dB gain about 0.4 m in diameter at the -3 dB level. Away from that peak, the response is approximately flat over most of the room. Optimal weighting for signal-to-noise enhancement degrades the spatial resolution minimally. Experimentally, we verify that signal-to-noise gain is less than proportional to the square root of the number of microphones probably due to the partial correlation of the noise between channels, to variation of signal intensity with polar angle about the source, and to imperfect correlation of the signal over the array caused by reverberations. We show measurements of the relative importance of each effect in our environment.

Noise alters hair-bundle mechanics at the cochlear apex

NASA Astrophysics Data System (ADS)

Strimbu, C. Elliott; Fridberger, Anders

2015-12-01

Exposure to loud sounds can lead to both permanent and short term changes in auditory sensitivity. Permanent hearing loss is often associated with gross changes in cochlear morphology including the loss of hair cells and auditory nerve fibers while the mechanisms of short term threshold shifts are much less well understood and may vary at different locations across the cochlea. Previous reports suggest that exposure to loud sounds leads to a decrease in the cochlear microphonic potential and in the stiffness of the organ of Corti. Because the cochlear microphonic reflects changes in the membrane potential of the hair cells, this suggests that hair-bundle motion should be reversibly altered following exposure to loud sounds. Using an in vitro preparation of the guinea pig temporal bone we investigate changes in the micro-mechanical response near the cochlear apex following a brief (up to 10 - 20 minutes) exposure to loud (˜ 120 dB) tones near the best frequency at this location. We use time-resolved confocal imaging to record the motion of outer hair cell bundles before and after acoustic overstimulation. We have also recorded larger-scale structural views of the organ of Corti before and after exposure to the loud sound. Conventional electrophysiological techniques are used measure the cochlear microphonic potential. As has been previously reported, following acoustic overexposure the cochlear microphonic declines in value and typically recovers on the order of 30 - 60 minutes. Hair-bundle trajectories are affected following the loud sound and typically recover on a somewhat faster time scale than the microphonic potential, although the results vary considerably across preparations. Preliminary results also suggest reversible changes in the hair cell's resting potential following the loud sound.

Comment on "Acoustical observation of bubble oscillations induced by bubble popping"

NASA Astrophysics Data System (ADS)

Blanc, É.; Ollivier, F.; Antkowiak, A.; Wunenburger, R.

2015-03-01

We have reproduced the experiment of acoustic monitoring of spontaneous popping of single soap bubbles standing in air reported by Ding et al. [2aa Phys. Rev. E 75, 041601 (2007), 10.1103/PhysRevE.75.041601]. By using a single microphone and two different signal acquisition systems recording in parallel the signal at the microphone output, among them the system used by Ding et al., we have experimentally evidenced that the acoustic precursors of bubble popping events detected by Ding et al. actually result from an acausal artifact of the signal processing performed by their acquisition system which lies outside of its prescribed working frequency range. No acoustic precursor of popping could be evidenced with the microphone used in these experiments, whose sensitivity is 1 V Pa-1 and frequency range is 500 Hz-100 kHz.

Adaptive Suppression of Noise in Voice Communications

NASA Technical Reports Server (NTRS)

Kozel, David; DeVault, James A.; Birr, Richard B.

2003-01-01

A subsystem for the adaptive suppression of noise in a voice communication system effects a high level of reduction of noise that enters the system through microphones. The subsystem includes a digital signal processor (DSP) plus circuitry that implements voice-recognition and spectral- manipulation techniques. The development of the adaptive noise-suppression subsystem was prompted by the following considerations: During processing of the space shuttle at Kennedy Space Center, voice communications among test team members have been significantly impaired in several instances because some test participants have had to communicate from locations with high ambient noise levels. Ear protection for the personnel involved is commercially available and is used in such situations. However, commercially available noise-canceling microphones do not provide sufficient reduction of noise that enters through microphones and thus becomes transmitted on outbound communication links.

Graphene electrostatic microphone and ultrasonic radio

PubMed Central

Zhou, Qin; Zheng, Jinglin; Onishi, Seita; Crommie, M. F.; Zettl, Alex K.

2015-01-01

We present a graphene-based wideband microphone and a related ultrasonic radio that can be used for wireless communication. It is shown that graphene-based acoustic transmitters and receivers have a wide bandwidth, from the audible region (20∼20 kHz) to the ultrasonic region (20 kHz to at least 0.5 MHz). Using the graphene-based components, we demonstrate efficient high-fidelity information transmission using an ultrasonic band centered at 0.3 MHz. The graphene-based microphone is also shown to be capable of directly receiving ultrasound signals generated by bats in the field, and the ultrasonic radio, coupled to electromagnetic (EM) radio, is shown to function as a high-accuracy rangefinder. The ultrasonic radio could serve as a useful addition to wireless communication technology where the propagation of EM waves is difficult. PMID:26150483

Joint Eglin Acoustics Week 2013 Data Report

DTIC Science & Technology

2017-10-01

during this test. The M-model HH-60 (Tail Number 04-27001), with the new wide-chord blade that is principally characterized by its unique tapered...cards located within each remote unit. Upon termination of each run , sufficient data metrics and system health information are transmitted back to the...command computer to assure that good data were acquired at each microphone station during the run . A typical WAMS microphone station deployment is

Analysis of Factors Affecting System Performance in the ASpIRE Challenge

DTIC Science & Technology

2015-12-13

performance in the ASpIRE (Automatic Speech recognition In Reverberant Environments) challenge. In particular, overall word error rate (WER) of the solver...systems is analyzed as a function of room, distance between talker and microphone, and microphone type. We also analyze speech activity detection...analysis will inform the design of future challenges and provide insight into the efficacy of current solutions addressing noisy reverberant speech

Signal Processing and Interpretation Using Multilevel Signal Abstractions.

DTIC Science & Technology

1986-06-01

mappings expressed in the Fourier domain. Pre- viously proposed causal analysis techniques for diagnosis are based on the analysis of intermediate data ...can be processed either as individual one-dimensional waveforms or as multichannel data 26 I P- - . . . ." " ." h9. for source detection and direction...microphone data . The signal processing for both spectral analysis of microphone signals and direc- * tion determination of acoustic sources involves

Locating arbitrarily time-dependent sound sources in three dimensional space in real time.

PubMed

Wu, Sean F; Zhu, Na

2010-08-01

This paper presents a method for locating arbitrarily time-dependent acoustic sources in a free field in real time by using only four microphones. This method is capable of handling a wide variety of acoustic signals, including broadband, narrowband, impulsive, and continuous sound over the entire audible frequency range, produced by multiple sources in three dimensional (3D) space. Locations of acoustic sources are indicated by the Cartesian coordinates. The underlying principle of this method is a hybrid approach that consists of modeling of acoustic radiation from a point source in a free field, triangulation, and de-noising to enhance the signal to noise ratio (SNR). Numerical simulations are conducted to study the impacts of SNR, microphone spacing, source distance and frequency on spatial resolution and accuracy of source localizations. Based on these results, a simple device that consists of four microphones mounted on three mutually orthogonal axes at an optimal distance, a four-channel signal conditioner, and a camera is fabricated. Experiments are conducted in different environments to assess its effectiveness in locating sources that produce arbitrarily time-dependent acoustic signals, regardless whether a sound source is stationary or moves in space, even toward behind measurement microphones. Practical limitations on this method are discussed.

Long-term usage of modern signal processing by listeners with severe or profound hearing loss: a retrospective survey.

PubMed

Keidser, Gitte; Hartley, David; Carter, Lyndal

2008-12-01

To investigate the long-term benefit of multichannel wide dynamic range compression (WDRC) alone and in combination with directional microphones and noise reduction/speech enhancement for listeners with severe or profound hearing loss. At the conclusion of a research project, 39 participants with severe or profound hearing loss were fitted with WDRC in one program and WDRC with directional microphones and speech enhancement enabled in a 2nd program. More than 2 years after the 1st participants exited the project, a retrospective survey was conducted to determine the participants' use of, and satisfaction with, the 2 programs. From the 30 returned questionnaires, it seems that WDRC is used with a high degree of satisfaction in general everyday listening situations. The reported benefit from the addition of a directional microphone and speech enhancement for listening in noisy environments was lower and varied among the users. This variable was significantly correlated with how much the program was used. The less frequent and more varied use of the program with directional microphones and speech enhancement activated in combination suggests that these features may be best offered in a 2nd listening program for listeners with severe or profound hearing loss.

A method for improving the drop test performance of a MEMS microphone

NASA Astrophysics Data System (ADS)

Winter, Matthias; Ben Aoun, Seifeddine; Feiertag, Gregor; Leidl, Anton; Scheele, Patrick; Seidel, Helmut

2009-05-01

Most micro electro mechanical system (MEMS) microphones are designed as capacitive microphones where a thin conductive membrane is located in front of a rigid counter electrode. The membrane is exposed to the environment to convert sound into vibrations of the membrane. The movement of the membrane causes a change in the capacitance between the membrane and the counter electrode. The resonance frequency of the membrane is designed to occur above the acoustic spectrum to achieve a linear frequency response. To obtain a good sensitivity the thickness of the membrane must be as small as possible, typically below 0.5 μm. These fragile membranes may be damaged by rapid pressure changes. For cell phones, drop tests are among the most relevant reliability tests. The extremely high acceleration during the drop impact leads to fast pressure changes in the microphone which could result in a rupture of the membrane. To overcome this problem a stable protection layer can be placed at a small distance to the membrane. The protective layer has small holes to form a low pass filter for air pressure. The low pass filter reduces pressure changes at high frequencies so that damage to the membrane by excitation in resonance will be prevented.

Active Self-Testing Noise Measurement Sensors for Large-Scale Environmental Sensor Networks

PubMed Central

Domínguez, Federico; Cuong, Nguyen The; Reinoso, Felipe; Touhafi, Abdellah; Steenhaut, Kris

2013-01-01

Large-scale noise pollution sensor networks consist of hundreds of spatially distributed microphones that measure environmental noise. These networks provide historical and real-time environmental data to citizens and decision makers and are therefore a key technology to steer environmental policy. However, the high cost of certified environmental microphone sensors render large-scale environmental networks prohibitively expensive. Several environmental network projects have started using off-the-shelf low-cost microphone sensors to reduce their costs, but these sensors have higher failure rates and produce lower quality data. To offset this disadvantage, we developed a low-cost noise sensor that actively checks its condition and indirectly the integrity of the data it produces. The main design concept is to embed a 13 mm speaker in the noise sensor casing and, by regularly scheduling a frequency sweep, estimate the evolution of the microphone's frequency response over time. This paper presents our noise sensor's hardware and software design together with the results of a test deployment in a large-scale environmental network in Belgium. Our middle-range-value sensor (around €50) effectively detected all experienced malfunctions, in laboratory tests and outdoor deployments, with a few false positives. Future improvements could further lower the cost of our sensor below €10. PMID:24351634

Comparisons of spectral characteristics of wind noise between omnidirectional and directional microphones.

PubMed

Chung, King

2012-06-01

Wind noise reduction is a topic of ongoing research and development for hearing aids and cochlear implants. The purposes of this study were to examine spectral characteristics of wind noise generated by directional (DIR) and omnidirectional (OMNI) microphones on different styles of hearing aids and to derive wind noise reduction strategies. Three digital hearing aids (BTE, ITE, and ITC) were fitted to Knowles Electronic Manikin for Acoustic Research. They were programmed to have linear amplification and matching frequency responses between the DIR and OMNI modes. Flow noise recordings were made from 0° to 360° azimuths at flow velocities of 4.5, 9.0, and 13.5 m/s in a quiet wind tunnel. Noise levels were analyzed in one-third octave bands from 100 to 8000 Hz. Comparison of wind noise revealed that DIR generally produced higher noise levels than OMNI for all hearing aids, but it could result in lower levels than OMNI at some frequencies and head angles. Wind noise reduction algorithms can be designed to detect noise levels of DIR and OMNI outputs in each frequency channel, remove the constraint to switch to OMNI in low-frequency channel(s) only, and adopt the microphone mode with lower noise levels to take advantage of the microphone differences.

Airframe Noise from a Hybrid Wing Body Aircraft Configuration

NASA Technical Reports Server (NTRS)

Hutcheson, Florence V.; Spalt, Taylor B.; Brooks, Thomas F.; Plassman, Gerald E.

2016-01-01

A high fidelity aeroacoustic test was conducted in the NASA Langley 14- by 22-Foot Subsonic Tunnel to establish a detailed database of component noise for a 5.8% scale HWB aircraft configuration. The model has a modular design, which includes a drooped and a stowed wing leading edge, deflectable elevons, twin verticals, and a landing gear system with geometrically scaled wheel-wells. The model is mounted inverted in the test section and noise measurements are acquired at different streamwise stations from an overhead microphone phased array and from overhead and sideline microphones. Noise source distribution maps and component noise spectra are presented for airframe configurations representing two different approach flight conditions. Array measurements performed along the aircraft flyover line show the main landing gear to be the dominant contributor to the total airframe noise, followed by the nose gear, the inboard side-edges of the LE droop, the wing tip/LE droop outboard side-edges, and the side-edges of deployed elevons. Velocity dependence and flyover directivity are presented for the main noise components. Decorrelation effects from turbulence scattering on spectral levels measured with the microphone phased array are discussed. Finally, noise directivity maps obtained from the overhead and sideline microphone measurements for the landing gear system are provided for a broad range of observer locations.

Evaluation of hearing protection used by police officers in the shooting range.

PubMed

Guida, Heraldo Lorena; Taxini, Carla Linhares; Gonçalves, Claudia Giglio de Oliveira; Valenti, Vitor Engrácia

2014-01-01

Impact noise is characterized by acoustic energy peaks that last less than a second, at intervals of more than 1s. To quantify the levels of impact noise to which police officers are exposed during activities at the shooting range and to evaluate the attenuation of the hearing protector. Measurements were performed in the shooting range of a military police department. An SV 102 audiodosimeter (Svantek) was used to measure sound pressure levels. Two microphones were used simultaneously: one external and one insertion type; the firearm used was a 0.40 Taurus® rimless pistol. The values obtained with the external microphone were 146 dBC (peak), and a maximum sound level of 129.4 dBC (fast). The results obtained with the insertion microphone were 138.7 dBC (peak), and a maximum sound level of 121.6 dBC (fast). The findings showed high levels of sound pressure in the shooting range, which exceeded the maximum recommended noise (120 dBC), even when measured through the insertion microphone. Therefore, alternatives to improve the performance of hearing protection should be considered. Copyright © 2014 Associação Brasileira de Otorrinolaringologia e Cirurgia Cérvico-Facial. Published by Elsevier Editora Ltda. All rights reserved.

Micromachined optical microphone structures with low thermal-mechanical noise levels.

PubMed

Hall, Neal A; Okandan, Murat; Littrell, Robert; Bicen, Baris; Degertekin, F Levent

2007-10-01

Micromachined microphones with diffraction-based optical displacement detection have been introduced previously [Hall et al., J. Acoust. Soc. Am. 118, 3000-3009 (2005)]. The approach has the advantage of providing high displacement detection resolution of the microphone diaphragm independent of device size and capacitance-creating an unconstrained design space for the mechanical structure itself. Micromachined microphone structures with 1.5-mm-diam polysilicon diaphragms and monolithically integrated diffraction grating electrodes are presented in this work with backplate architectures that deviate substantially from traditional perforated plate designs. These structures have been designed for broadband frequency response and low thermal mechanical noise levels. Rigorous experimental characterization indicates a diaphragm displacement detection resolution of 20 fm radicalHz and a thermal mechanical induced diaphragm displacement noise density of 60 fm radicalHz, corresponding to an A-weighted sound pressure level detection limit of 24 dB(A) for these structures. Measured thermal mechanical displacement noise spectra are in excellent agreement with simulations based on system parameters derived from dynamic frequency response characterization measurements, which show a diaphragm resonance limited bandwidth of approximately 20 kHz. These designs are substantial improvements over initial prototypes presented previously. The high performance-to-size ratio achievable with this technology is expected to have an impact on a variety of instrumentation and hearing applications.

Solar hydrogen by thermochemical water splitting cycles: design, modeling, and demonstration of a novel receiver/reactor for the high temperature decomposition of zno using concentrated sunlight

NASA Astrophysics Data System (ADS)

Kaiser, Zachary David Epping

Documenting the presence of rare bat species can be difficult. The current summer survey protocol for the federally endangered Indiana bat ( Myotis sodalis) requires passive acoustic sampling with directional microphones (e.g., Anabats), but there are still questions about best practices for choosing survey sites and appropriate detector models. Indiana bats are capable of foraging in an array of cover types, including structurally-complex, interior forests. Further, data acquisition among different commercially available bat detectors is likely highly variable, due to the use of proprietary microphones with different frequency responses, sensitivities, and directionality. We paired omnidirectional Wildlife Acoustic SM2BAT+ (SM2) and directional Titley Scientific Anabat SD2 (Anabat) detectors at 71 random points near Indianapolis, Indiana from May-August 2012-2013 to compare data acquisition by phonic group (low, mid, Myotis) and to determine what factors affect probability of detection and site occupancy for Indiana bats when sampling with acoustics near an active maternity colony (0.20--8.39 km away). Weatherproofing for Anabat microphones was 45° angle PVC tubes and for SM2 microphones was their foam shielding; microphones were paired at 2 m and 5 m heights. Habitat and landscape covariates were measured in the field or via ArcGIS. We adjusted file parameters to make SM2 and Anabat data comparable. Files were identified using Bat Call ID software, with visual inspection of Indiana bat calls. The effects of detector type, phonic group, height, and their interactions on mean files recorded per site were assessed using generalized estimating equations and LSD pairwise comparisons. We reduced probability of detection (p) and site occupancy (ψ) model covariates with Pearson's correlation and PCA. We used Presence 6.1 software and Akaike's Information Criteria to assess models for p and ψ. Anabats and SM2s did not perform equally. Anabats recorded more low and midrange files, but fewer Myotis files per site than SM2s. When comparing the same model of detectors, deployment height did not impact data acquisition. Weatherproofing may limit the ability of Anabats to record Myotis, but Anabat microphones may have greater detection ranges for low and midrange bats. Indiana bat detections were low for both detector types, representing only 4.4% of identifiable bat files recorded by SM2s. We detected Indiana bats at 43.7% of sampled sites and on 31.4% of detector-nights; detectability increased as "forest closure" and mean nightly temperature increased, likely due to reduced clutter and increased bat activity, respectively. Proximity to colony trees and specific cover types generally did not affect occupancy, suggesting that Indiana bats use a variety of cover types in this landscape. Omnidirectional SMX-US microphones may be more appropriate for Indiana bat surveys than directional Anabat microphones. However, we conclude that 2 nights of passive acoustic sampling per site may be insufficient for reliably detecting this species when it is present. In turn, the use of acoustic monitoring as a means to document presence or probable absence should be reassessed.

Development of an Audio Microphone for the Mars Surveyor 98 Lander

NASA Astrophysics Data System (ADS)

Delory, G. T.; Luhmann, J. G.; Curtis, D. W.; Friedman, L. D.; Primbsch, J. H.; Mozer, F. S.

1998-01-01

In December 1999, the next Mars Surveyor Lander will bring the first microphone to the surface of Mars. The Mars Microphone represents a joint effort between the Planetary Society and the University of California at Berkeley Space Sciences Laboratory and is riding on the lander as part of the LIDAR instrument package provided by the Russian Academy of Sciences in Moscow. The inclusion of a microphone on the Mars Surveyor Lander represents a unique opportunity to sample for the first time the acoustic environment on the surface, including both natural and lander-generated sounds. Sounds produced by martian meteorology are among the signals to be recorded, including wind and impacts of sand particles on the instrument. Photographs from the Viking orbiters as well as Pathfinder images show evidence of small tornado-like vortices that may be acoustically detected, along with noise generated by static discharges possible during sandstorms. Lander-generated sounds that will be measured include the motion and digging of the lander arm as it gathers soil samples for analysis. Along with these scientific objectives, the Mars Microphone represents a powerful tool for public outreach by providing sound samples on the Internet recorded during the mission. The addition of audio capability to the lander brings us one step closer to a true virtual presence on the Mars surface by complementing the visual capabilities of the Mars Surveyor cameras. The Mars Microphone is contained in a 5 x 5 x 1 cm box, weighs less than 50 g, and uses 0.1 W of power during its most active times. The microphone used is a standard hearing-aid electret. The sound sampling and processing system relies on an RSC-164 speech processor chip, which performs 8-bit A/ D sampling and sound compression. An onboard flight program enables several modes for the instrument, including varying sample ranges of 5 kHz and 20 kHz, and a selectable gain setting with 64x dynamic range. The device automatically triggers on the loudest sound during a collection period for storage in an internal flash memory. Data returned by the lander consist of a compressed time-series acoustic waveform, between 2 and 10 s long, depending on the sample rate. In addition to the discrete waveform. capture, the instrument continuously records the mean power in each of six frequency bands in order to provide an average characterization of the martian acoustic environment. Once the data are retrieved from the telemetry, the compressed time series is expanded into a standard PC-compatible WAV file for analysis, which will include representation in spectral format using FFTs for quantitative characterization of the sound data. The WAV files will be used to share the data with the public via the Internet. The Mars Microphone will thus fulfill a dual role on the Mars Surveyor mission, one as a possible precursor to a more sophisticated acoustic instrument on future landers. and one as a mechanism to increase public awareness of efforts to explore and understand the martian climate and planetary history.

SNMP Over Wi-Fi Wireless Networks

DTIC Science & Technology

2003-03-01

headphone, 1 x microphone, 1 x AC adapter 73 Wireless connectivity IrDA, Wi-Fi (IEEE 802.11b) Power Battery installed (max) 1 x Lithium Ion battery ...headphone, 1 x microphone, 1 x AC adapter Wireless connectivity Bluetooth, IrDA, Wi-Fi Power Battery installed (max) 1 x Lithium Ion battery ...is required. However Microsoft released the new version of Embedded Visual Tool that integrated Pocket PC 2002 SDK and Smartphone 2002 SDK on

A study of space shuttle structural integrity test and assessment. Part 1

NASA Technical Reports Server (NTRS)

Anderson, R. E.; Poe, R. G.

1972-01-01

The ultrasonics technique for assessing the structural integrity of the primary surface of the space shuttle vehicles is discussed and evaluated. Analysis was made of transducers, transducer coupling test structure fabrication, flaws, and ultrasonic testing. Graphs of microphone response curves from the initial noise tests, accelerometer response curves from the final noise tests, and microphone curves from the final noise tests are included along with a glossary, bibliography, and results.

Removing Background Noise with Phased Array Signal Processing

NASA Technical Reports Server (NTRS)

Podboy, Gary; Stephens, David

2015-01-01

Preliminary results are presented from a test conducted to determine how well microphone phased array processing software could pull an acoustic signal out of background noise. The array consisted of 24 microphones in an aerodynamic fairing designed to be mounted in-flow. The processing was conducted using Functional Beam forming software developed by Optinav combined with cross spectral matrix subtraction. The test was conducted in the free-jet of the Nozzle Acoustic Test Rig at NASA GRC. The background noise was produced by the interaction of the free-jet flow with the solid surfaces in the flow. The acoustic signals were produced by acoustic drivers. The results show that the phased array processing was able to pull the acoustic signal out of the background noise provided the signal was no more than 20 dB below the background noise level measured using a conventional single microphone equipped with an aerodynamic forebody.

Characterization of air-coupled ultrasound transducers in the frequency range 40 kHz-2 mHz using light diffraction tomography.

PubMed

Almqvist, M; Holm, A; Persson, H W; Lindström, K

2000-01-01

The aim of this work was to show the applicability of light diffraction tomography on airborne ultrasound in the frequency range 40 kHz-2 MHz. Seven different air-coupled transducers were measured to show the method's performance regarding linearity, absolute pressure measurements, phase measurements, frequency response, S/N ratio and spatial resolution. A calibrated microphone and the pulse-echo method were used to evaluate the results. The absolute measurements agreed within the calibrated microphone's uncertainty range. Pulse waveforms and corresponding FFT diagrams show the method's higher bandwidth compared with the microphone. Further, the method offers non-perturbing measurements with high spatial resolution, which was especially advantageous for measurements close to the transducer surfaces. The S/N ratio was higher than or in the same range as that of the two comparison methods.

Doppler distortion correction based on microphone array and matching pursuit algorithm for a wayside train bearing monitoring system

NASA Astrophysics Data System (ADS)

Liu, Xingchen; Hu, Zhiyong; He, Qingbo; Zhang, Shangbin; Zhu, Jun

2017-10-01

Doppler distortion and background noise can reduce the effectiveness of wayside acoustic train bearing monitoring and fault diagnosis. This paper proposes a method of combining a microphone array and matching pursuit algorithm to overcome these difficulties. First, a dictionary is constructed based on the characteristics and mechanism of a far-field assumption. Then, the angle of arrival of the train bearing is acquired when applying matching pursuit to analyze the acoustic array signals. Finally, after obtaining the resampling time series, the Doppler distortion can be corrected, which is convenient for further diagnostic work. Compared with traditional single-microphone Doppler correction methods, the advantages of the presented array method are its robustness to background noise and its barely requiring pre-measuring parameters. Simulation and experimental study show that the proposed method is effective in performing wayside acoustic bearing fault diagnosis.

XV-15 Tiltrotor Low Noise Terminal Area Operations

NASA Technical Reports Server (NTRS)

Conner, David A.; Marcolini, Michael A.; Edwards, Bryan D.; Brieger, John T.

1998-01-01

Acoustic data have been acquired for the XV-15 tiltrotor aircraft performing a variety of terminal area operating procedures. This joint NASA/Bell/Army test program was conducted in two phases. During Phase 1 the XV-15 was flown over a linear array of microphones, deployed perpendicular to the flight path, at a number of fixed operating conditions. This documented the relative noise differences between the various conditions. During Phase 2 the microphone array was deployed over a large area to directly measure the noise footprint produced during realistic approach and departure procedures. The XV-15 flew approach profiles that culminated in IGE hover over a landing pad, then takeoffs from the hover condition back out over the microphone array. Results from Phase 1 identify noise differences between selected operating conditions, while those from Phase 2 identify differences in noise footprints between takeoff and approach conditions and changes in noise footprint due to variation in approach procedures.

Location of aerodynamic noise sources from a 200 kW vertical-axis wind turbine

NASA Astrophysics Data System (ADS)

Ottermo, Fredric; Möllerström, Erik; Nordborg, Anders; Hylander, Jonny; Bernhoff, Hans

2017-07-01

Noise levels emitted from a 200 kW H-rotor vertical-axis wind turbine have been measured using a microphone array at four different positions, each at a hub-height distance from the tower. The microphone array, comprising 48 microphones in a spiral pattern, allows for directional mapping of the noise sources in the range of 500 Hz to 4 kHz. The produced images indicate that most of the noise is generated in a narrow azimuth-angle range, compatible with the location where increased turbulence is known to be present in the flow, as a result of the previous passage of a blade and its support arms. It is also shown that a semi-empirical model for inflow-turbulence noise seems to produce noise levels of the correct order of magnitude, based on the amount of turbulence that could be expected from power extraction considerations.

Virtual microphone sensing through vibro-acoustic modelling and Kalman filtering

NASA Astrophysics Data System (ADS)

van de Walle, A.; Naets, F.; Desmet, W.

2018-05-01

This work proposes a virtual microphone methodology which enables full field acoustic measurements for vibro-acoustic systems. The methodology employs a Kalman filtering framework in order to combine a reduced high-fidelity vibro-acoustic model with a structural excitation measurement and small set of real microphone measurements on the system under investigation. By employing model order reduction techniques, a high order finite element model can be converted in a much smaller model which preserves the desired accuracy and maintains the main physical properties of the original model. Due to the low order of the reduced-order model, it can be effectively employed in a Kalman filter. The proposed methodology is validated experimentally on a strongly coupled vibro-acoustic system. The virtual sensor vastly improves the accuracy with respect to regular forward simulation. The virtual sensor also allows to recreate the full sound field of the system, which is very difficult/impossible to do through classical measurements.

Characteristics and measurement of supersonic projectile shock waves by a 32-microphone ring array

NASA Astrophysics Data System (ADS)

Chang, Ho; Wu, Yan-Chyuan; Tsung, Tsing-Tshih

2011-08-01

This paper discusses about the characteristics of supersonic projectile shock wave in muzzle region during firing of high explosive anti-tank (HEAT) and high explosive (HE) projectiles. HEAT projectiles are fired horizontally at a muzzle velocity of Mach 3.5 from a medium caliber tank gun equipped with a newly designed multi-perforated muzzle brake, whereas HE projectiles are fired at elevation angles at a muzzle velocity of Mach 2 from a large caliber howitzer equipped with a newly designed double-baffle muzzle brake. In the near field, pressure signatures of the N-wave generated from projectiles are measured by 32-microphone ring array wrapped by cotton sheath. Records measured by the microphone array are used to demonstrate several key characteristics of the shock wave of supersonic projectile. All measurements made in this study can be a significant reference for developing guns, tanks, or the chassis of fighting vehicles.

Firefighters' communication transceiver test plan

NASA Astrophysics Data System (ADS)

Wallace, R. J.

1984-05-01

The requirements for the operational testing of the firefighters communication transceiver were identified. The major concerns centered around the integrity and reliability of the firefighter/microphone interface. The major concern about the radio hardware was that it be intrinsically safe in hazardous atmospheres and that the system not interfere with the fit or facial seal of self-contained breathing apparatus (SCBA). The greatest concern for operational testing purposes as the reliability and clarity of the line of communication between the firefighter and those on the fireground with whom he must maintain contact. A desire to test any units developed in both training exercises and in real responses to hazardous material incidents was expressed. It is felt that a VOX-microphone built into the SCBA facemask gives the best performance. A voice-pickup product device which combines a bone conduction microphone and a speaker into a single ear mounted unit is examined.

Multimodal physiological sensor for motion artefact rejection.

PubMed

Goverdovsky, Valentin; Looney, David; Kidmose, Preben; Mandic, Danilo P

2014-01-01

This work introduces a novel physiological sensor, which combines electrical and mechanical modalities in a co-located arrangement, to reject motion-induced artefacts. The mechanically sensitive element consists of an electret condenser microphone containing a light diaphragm, allowing it to detect local mechanical displacements and disregard large-scale whole body movements. The electrically sensitive element comprises a highly flexible membrane, conductive on one side and insulating on the other. It covers the sound hole of the microphone, thereby forming an isolated pocket of air between the membrane and the diaphragm. The co-located arrangement of the modalities allows the microphone to sense mechanical disturbances directly through the electrode, thus providing an accurate proxy to artefacts caused by relative motion between the skin and the electrode. This proxy is used to reject such artefacts in the electrical physiological signals, enabling enhanced recording quality in wearable health applications.

Design optimization of condenser microphone: a design of experiment perspective.

PubMed

Tan, Chee Wee; Miao, Jianmin

2009-06-01

A well-designed condenser microphone backplate is very important in the attainment of good frequency response characteristics--high sensitivity and wide bandwidth with flat response--and low mechanical-thermal noise. To study the design optimization of the backplate, a 2(6) factorial design with a single replicate, which consists of six backplate parameters and four responses, has been undertaken on a comprehensive condenser microphone model developed by Zuckerwar. Through the elimination of insignificant parameters via normal probability plots of the effect estimates, the projection of an unreplicated factorial design into a replicated one can be performed to carry out an analysis of variance on the factorial design. The air gap and slot have significant effects on the sensitivity, mechanical-thermal noise, and bandwidth while the slot/hole location interaction has major influence over the latter two responses. An organized and systematic approach of designing the backplate is summarized.

Modified Skvor/Starr approach in the mechanical-thermal noise analysis of condenser microphone.

PubMed

Tan, Chee Wee; Miao, Jianmin

2009-11-01

Simple analytical expressions of mechanical resistance, such as those formulated by Skvor/Starr, are widely used to describe the mechanical-thermal noise performance of a condenser microphone. However, the Skvor/Starr approach does not consider the location effect of acoustic holes in the backplate and overestimates the total equivalent mechanical resistance and mechanical-thermal noise. In this paper, a modified form of the Skvor/Starr approach is proposed to address this hole location dependent effect. A mode shape factor, which consists of the zero order Bessel and modified Bessel functions, is included in Skvor's mechanical resistance formulation to consider the effect of the hole location in the backplate. With reference to two B&K microphones, the theoretical results of the A-weighted mechanical-thermal noise obtained by the modified Skvor/Starr approach are in good agreements with those reported experimental ones.

Wind Noise Reduction in a Non-Porous Subsurface Windscreen

NASA Technical Reports Server (NTRS)

Zuckerwar, Allan J.; Shams, Qamar A.; Knight, H. Keith

2012-01-01

Measurements of wind noise reduction were conducted on a box-shaped, subsurface windscreen made of closed cell polyurethane foam. The windscreen was installed in the ground with the lid flush with the ground surface. The wind was generated by means of a fan, situated on the ground, and the wind speed was measured at the center of the windscreen lid with an ultrasonic anemometer. The wind speed was controlled by moving the fan to selected distances from the windscreen. The wind noise was measured on a PCB Piezotronics 3†electret microphone. Wind noise spectra were measured with the microphone exposed directly to the wind (atop the windscreen lid) and with the microphone installed inside the windscreen. The difference between the two spectra comprises the wind noise reduction. At wind speeds of 3, 5, and 7 m/s, the wind noise reduction is typically 15 dB over the frequency range of 0.1-20 Hz.

Instrumentation for measurement of aircraft noise and sonic boom

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J. (Inventor)

1975-01-01

A jet aircraft noise and sonic boom measuring device which converts sound pressure into electric current is described. An electric current proportional to the sound pressure level at a condenser microphone is produced and transmitted over a cable, amplified by a zero drive amplifier and recorded on magnetic tape. The converter is comprised of a local oscillator, a dual-gate field-effect transistor (FET) mixer and a voltage regulator/impedance translator. A carrier voltage that is applied to one of the gates of the FET mixer is generated by the local oscillator. The microphone signal is mixed with the carrier to produce an electrical current at the frequency of vibration of the microphone diaphragm by the FET mixer. The voltage of the local oscillator and mixer stages is regulated, the carrier at the output is eliminated, and a low output impedance at the cable terminals is provided by the voltage regulator/impedance translator.

Nanogenerator-based dual-functional and self-powered thin patch loudspeaker or microphone for flexible electronics.

PubMed

Li, Wei; Torres, David; Díaz, Ramón; Wang, Zhengjun; Wu, Changsheng; Wang, Chuan; Lin Wang, Zhong; Sepúlveda, Nelson

2017-05-16

Ferroelectret nanogenerators were recently introduced as a promising alternative technology for harvesting kinetic energy. Here we report the device's intrinsic properties that allow for the bidirectional conversion of energy between electrical and mechanical domains; thus extending its potential use in wearable electronics beyond the power generation realm. This electromechanical coupling, combined with their flexibility and thin film-like form, bestows dual-functional transducing capabilities to the device that are used in this work to demonstrate its use as a thin, wearable and self-powered loudspeaker or microphone patch. To determine the device's performance and applicability, sound pressure level is characterized in both space and frequency domains for three different configurations. The confirmed device's high performance is further validated through its integration in three different systems: a music-playing flag, a sound recording film and a flexible microphone for security applications.

Nanogenerator-based dual-functional and self-powered thin patch loudspeaker or microphone for flexible electronics

NASA Astrophysics Data System (ADS)

Li, Wei; Torres, David; Díaz, Ramón; Wang, Zhengjun; Wu, Changsheng; Wang, Chuan; Lin Wang, Zhong; Sepúlveda, Nelson

2017-05-01

Ferroelectret nanogenerators were recently introduced as a promising alternative technology for harvesting kinetic energy. Here we report the device's intrinsic properties that allow for the bidirectional conversion of energy between electrical and mechanical domains; thus extending its potential use in wearable electronics beyond the power generation realm. This electromechanical coupling, combined with their flexibility and thin film-like form, bestows dual-functional transducing capabilities to the device that are used in this work to demonstrate its use as a thin, wearable and self-powered loudspeaker or microphone patch. To determine the device's performance and applicability, sound pressure level is characterized in both space and frequency domains for three different configurations. The confirmed device's high performance is further validated through its integration in three different systems: a music-playing flag, a sound recording film and a flexible microphone for security applications.

Acoustic Localization with Infrasonic Signals

NASA Astrophysics Data System (ADS)

Threatt, Arnesha; Elbing, Brian

2015-11-01

Numerous geophysical and anthropogenic events emit infrasonic frequencies (<20 Hz), including volcanoes, hurricanes, wind turbines and tornadoes. These sounds, which cannot be heard by the human ear, can be detected from large distances (in excess of 100 miles) due to low frequency acoustic signals having a very low decay rate in the atmosphere. Thus infrasound could be used for long-range, passive monitoring and detection of these events. An array of microphones separated by known distances can be used to locate a given source, which is known as acoustic localization. However, acoustic localization with infrasound is particularly challenging due to contamination from other signals, sensitivity to wind noise and producing a trusted source for system development. The objective of the current work is to create an infrasonic source using a propane torch wand or a subwoofer and locate the source using multiple infrasonic microphones. This presentation will present preliminary results from various microphone configurations used to locate the source.

Characteristics and measurement of supersonic projectile shock waves by a 32-microphone ring array.

PubMed

Chang, Ho; Wu, Yan-Chyuan; Tsung, Tsing-Tshih

2011-08-01

This paper discusses about the characteristics of supersonic projectile shock wave in muzzle region during firing of high explosive anti-tank (HEAT) and high explosive (HE) projectiles. HEAT projectiles are fired horizontally at a muzzle velocity of Mach 3.5 from a medium caliber tank gun equipped with a newly designed multi-perforated muzzle brake, whereas HE projectiles are fired at elevation angles at a muzzle velocity of Mach 2 from a large caliber howitzer equipped with a newly designed double-baffle muzzle brake. In the near field, pressure signatures of the N-wave generated from projectiles are measured by 32-microphone ring array wrapped by cotton sheath. Records measured by the microphone array are used to demonstrate several key characteristics of the shock wave of supersonic projectile. All measurements made in this study can be a significant reference for developing guns, tanks, or the chassis of fighting vehicles.

Firefighters' communication transceiver test plan

NASA Technical Reports Server (NTRS)

Wallace, R. J.

1984-01-01

The requirements for the operational testing of the firefighters communication transceiver were identified. The major concerns centered around the integrity and reliability of the firefighter/microphone interface. The major concern about the radio hardware was that it be intrinsically safe in hazardous atmospheres and that the system not interfere with the fit or facial seal of self-contained breathing apparatus (SCBA). The greatest concern for operational testing purposes as the reliability and clarity of the line of communication between the firefighter and those on the fireground with whom he must maintain contact. A desire to test any units developed in both training exercises and in real responses to hazardous material incidents was expressed. It is felt that a VOX-microphone built into the SCBA facemask gives the best performance. A voice-pickup product device which combines a bone conduction microphone and a speaker into a single ear mounted unit is examined.

Nanogenerator-based dual-functional and self-powered thin patch loudspeaker or microphone for flexible electronics

PubMed Central

Li, Wei; Torres, David; Díaz, Ramón; Wang, Zhengjun; Wu, Changsheng; Wang, Chuan; Lin Wang, Zhong; Sepúlveda, Nelson

2017-01-01

Ferroelectret nanogenerators were recently introduced as a promising alternative technology for harvesting kinetic energy. Here we report the device's intrinsic properties that allow for the bidirectional conversion of energy between electrical and mechanical domains; thus extending its potential use in wearable electronics beyond the power generation realm. This electromechanical coupling, combined with their flexibility and thin film-like form, bestows dual-functional transducing capabilities to the device that are used in this work to demonstrate its use as a thin, wearable and self-powered loudspeaker or microphone patch. To determine the device's performance and applicability, sound pressure level is characterized in both space and frequency domains for three different configurations. The confirmed device's high performance is further validated through its integration in three different systems: a music-playing flag, a sound recording film and a flexible microphone for security applications. PMID:28508862

X-wing noise data acquisition program

NASA Technical Reports Server (NTRS)

Healy, G. J.

1983-01-01

The X-wing circulation controlled rotor system model was tested for hover performance. During these performance tests, noise data from 12 microphones was recorded on magnetic tape for subsequent data reduction. The rotor system was operated at 4 tip speeds ranging from 529 to 650 ft./sec. (404 to 497 rpm), collective angles of attack fro 0 deg to 8.5 deg (maximum), and blade pressure ratios from 1.0 (no blowing) to a maximum of 2.1. The 12 microphones included 11 in the far field, and one in the transmission area. Following completion of the rotor and subsystem noise measurements, sound field calibration measurements were made of both the rotor 'bowl' and the loudspeaker system used in the 'bowl' calibration measurements. The location of 10 far field microphones was measured by a surveyor. Additionally, detailed tape logs were prepared for the six reels of tape used for the program.

Automated speech recognition for time recording in out-of-hospital emergency medicine-an experimental approach.

PubMed

Gröschel, J; Philipp, F; Skonetzki, St; Genzwürker, H; Wetter, Th; Ellinger, K

2004-02-01

Precise documentation of medical treatment in emergency medical missions and for resuscitation is essential from a medical, legal and quality assurance point of view [Anästhesiologie und Intensivmedizin, 41 (2000) 737]. All conventional methods of time recording are either too inaccurate or elaborate for routine application. Automated speech recognition may offer a solution. A special erase programme for the documentation of all time events was developed. Standard speech recognition software (IBM ViaVoice 7.0) was adapted and installed on two different computer systems. One was a stationary PC (500MHz Pentium III, 128MB RAM, Soundblaster PCI 128 Soundcard, Win NT 4.0), the other was a mobile pen-PC that had already proven its value during emergency missions [Der Notarzt 16, p. 177] (Fujitsu Stylistic 2300, 230Mhz MMX Processor, 160MB RAM, embedded soundcard ESS 1879 chipset, Win98 2nd ed.). On both computers two different microphones were tested. One was a standard headset that came with the recognition software, the other was a small microphone (Lavalier-Kondensatormikrofon EM 116 from Vivanco), that could be attached to the operators collar. Seven women and 15 men spoke a text with 29 phrases to be recognised. Two emergency physicians tested the system in a simulated emergency setting using the collar microphone and the pen-PC with an analogue wireless connection. Overall recognition was best for the PC with a headset (89%) followed by the pen-PC with a headset (85%), the PC with a microphone (84%) and the pen-PC with a microphone (80%). Nevertheless, the difference was not statistically significant. Recognition became significantly worse (89.5% versus 82.3%, P<0.0001 ) when numbers had to be recognised. The gender of speaker and the number of words in a sentence had no influence. Average recognition in the simulated emergency setting was 75%. At no time did false recognition appear. Time recording with automated speech recognition seems to be possible in emergency medical missions. Although results show an average recognition of only 75%, it is possible that missing elements may be reconstructed more precisely. Future technology should integrate a secure wireless connection between microphone and mobile computer. The system could then prove its value for real out-of-hospital emergencies.

Microphone Array

NASA Astrophysics Data System (ADS)

Bader, Rolf

This chapter deals with microphone arrays. It is arranged according to the different methods available to proceed through the different problems and through the different mathematical methods. After discussing general properties of different array types, such as plane arrays, spherical arrays, or scanning arrays, it proceeds to the signal processing tools that are most used in speech processing. In the third section, backpropagating methods based on the Helmholtz-Kirchhoff integral are discussed, which result in spatial radiation patterns of vibrating bodies or air.

Multisensor speech input

NASA Astrophysics Data System (ADS)

Viswanathan, V. R.; Karnofsky, K. F.; Stevens, K. N.; Alakel, M. N.

1983-12-01

The use of multiple sensors to transduce speech was investigated. A data base of speech and noise was collected from a number of transducers located on and around the head of the speaker. The transducers included pressure, first order gradient, second order gradient microphones and an accelerometer. The effort analyzed this data and evaluated the performance of a multiple sensor configuration. The conclusion was: multiple transducer configurations can provide a signal containing more useable speech information than that provided by a microphone.

Direct measurement of the speed of sound using a microphone and a speaker

NASA Astrophysics Data System (ADS)

Gómez-Tejedor, José A.; Castro-Palacio, Juan C.; Monsoriu, Juan A.

2014-05-01

We present a simple and accurate experiment to obtain the speed of sound in air using a conventional speaker and a microphone connected to a computer. A free open source digital audio editor and recording computer software application allows determination of the time-of-flight of the wave for different distances, from which the speed of sound is calculated. The result is in very good agreement with the reported value in the literature.

Microphone Array Phased Processing System (MAPPS): Version 4.0 Manual

NASA Technical Reports Server (NTRS)

Watts, Michael E.; Mosher, Marianne; Barnes, Michael; Bardina, Jorge

1999-01-01

A processing system has been developed to meet increasing demands for detailed noise measurement of individual model components. The Microphone Array Phased Processing System (MAPPS) uses graphical user interfaces to control all aspects of data processing and visualization. The system uses networked parallel computers to provide noise maps at selected frequencies in a near real-time testing environment. The system has been successfully used in the NASA Ames 7- by 10-Foot Wind Tunnel.

Rocket Motor Microphone Investigation

NASA Technical Reports Server (NTRS)

Pilkey, Debbie; Herrera, Eric; Gee, Kent L.; Giraud, Jerom H.; Young, Devin J.

2010-01-01

At ATK's facility in Utah, large full-scale solid rocket motors are tested. The largest is a five-segment version of the reusable solid rocket motor, which is for use on the Ares I launch vehicle. As a continuous improvement project, ATK and BYU investigated the use of microphones on these static tests, the vibration and temperature to which the instruments are subjected, and in particular the use of vent tubes and the effects these vents have at low frequencies.

Various Sensors Aboard UAVs

DTIC Science & Technology

1999-07-01

suffer from a microphonic noise . For that reason the previous integral Stirling coolers ( cfr fig.AlO ) have been replaced by the split Stirling coolers in...order to avoid that micropho-nic noise of the detector which rings like a bell when it is pinched. B. 2. Physics and Investigation of Thermal Images...by resorting to QWIPs (Quantum Well Infrared Photodetector) whose working is explained by fig.A17 and its caption; in the case of a QWIP , the cut-of

Final report on COOMET.AUV.A-S1: Technical report on supplementary comparison 'Comparison of national standards of the sound pressure unit in air through calibration of working reference microphones'

NASA Astrophysics Data System (ADS)

Pozdeeva, Valentina; Chalyy, Vladimir

2014-01-01

The supplementary comparison COOMET.AUV.A-S1 for secondary calibration methods using WS1 and WS2 measurement microphones was carried out from 2009 to 2010. The results were submitted to and approved by CCAUV in April 2014. Four National Metrology Institutes took part in this comparison and are as follows: BelGIM (Belarus), VNIIFTRI (Russia), SMU (Slovakia) and DP NDI 'Sistema' (Ukraine). Three of the above NMIs (VNIIFTRI, SMU and DP NDI 'Sistema') had earlier participated in COOMET key comparisons and one NMI (VNIIFTRI) had also participated in CCAUV key comparisons. The Comparison Reference Values were calculated as the weighted mean values from results obtained by three institutes. The comparison results show agreement for all participants in the frequency range from 20 Hz to 12.5 kHz for WS1 microphones, and in the frequency range from 20 Hz to 16 kHz for WS2 microphones. Main text. To reach the main text of this paper, click on Final Report. Note that this text is that which appears in Appendix B of the BIPM key comparison database kcdb.bipm.org/. The final report has been peer-reviewed and approved for publication by the CCAUV, according to the provisions of the CIPM Mutual Recognition Arrangement (CIPM MRA).

Detecting and Classifying Human Touches in a Social Robot Through Acoustic Sensing and Machine Learning

PubMed Central

Alonso-Martín, Fernando; Gamboa-Montero, Juan José; Castillo, José Carlos; Castro-González, Álvaro; Salichs, Miguel Ángel

2017-01-01

An important aspect in Human–Robot Interaction is responding to different kinds of touch stimuli. To date, several technologies have been explored to determine how a touch is perceived by a social robot, usually placing a large number of sensors throughout the robot’s shell. In this work, we introduce a novel approach, where the audio acquired from contact microphones located in the robot’s shell is processed using machine learning techniques to distinguish between different types of touches. The system is able to determine when the robot is touched (touch detection), and to ascertain the kind of touch performed among a set of possibilities: stroke, tap, slap, and tickle (touch classification). This proposal is cost-effective since just a few microphones are able to cover the whole robot’s shell since a single microphone is enough to cover each solid part of the robot. Besides, it is easy to install and configure as it just requires a contact surface to attach the microphone to the robot’s shell and plug it into the robot’s computer. Results show the high accuracy scores in touch gesture recognition. The testing phase revealed that Logistic Model Trees achieved the best performance, with an F-score of 0.81. The dataset was built with information from 25 participants performing a total of 1981 touch gestures. PMID:28509865

Virtual design and optimization studies for industrial silicon microphones applying tailored system-level modeling

NASA Astrophysics Data System (ADS)

Kuenzig, Thomas; Dehé, Alfons; Krumbein, Ulrich; Schrag, Gabriele

2018-05-01

Maxing out the technological limits in order to satisfy the customers’ demands and obtain the best performance of micro-devices and-systems is a challenge of today’s manufacturers. Dedicated system simulation is key to investigate the potential of device and system concepts in order to identify the best design w.r.t. the given requirements. We present a tailored, physics-based system-level modeling approach combining lumped with distributed models that provides detailed insight into the device and system operation at low computational expense. The resulting transparent, scalable (i.e. reusable) and modularly composed models explicitly contain the physical dependency on all relevant parameters, thus being well suited for dedicated investigation and optimization of MEMS devices and systems. This is demonstrated for an industrial capacitive silicon microphone. The performance of such microphones is determined by distributed effects like viscous damping and inhomogeneous capacitance variation across the membrane as well as by system-level phenomena like package-induced acoustic effects and the impact of the electronic circuitry for biasing and read-out. The here presented model covers all relevant figures of merit and, thus, enables to evaluate the optimization potential of silicon microphones towards high fidelity applications. This work was carried out at the Technical University of Munich, Chair for Physics of Electrotechnology. Thomas Kuenzig is now with Infineon Technologies AG, Neubiberg.

Lightweight fiber optic microphones and accelerometers

NASA Astrophysics Data System (ADS)

Bucaro, J. A.; Lagakos, N.

2001-06-01

We have designed, fabricated, and tested two lightweight fiber optic sensors for the dynamic measurement of acoustic pressure and acceleration. These sensors, one a microphone and the other an accelerometer, are required for active blanket sound control technology under development in our laboratory. The sensors were designed to perform to certain specifications dictated by our active sound control application and to do so without exhibiting sensitivity to the high electrical voltages expected to be present. Furthermore, the devices had to be small (volumes less than 1.5 cm3) and light (less than 2 g). To achieve these design criteria, we modified and extended fiber optic reflection microphone and fiber microbend displacement device designs reported in the literature. After fabrication, the performances of each sensor type were determined from measurements made in a dynamic pressure calibrator and on a shaker table. The fiber optic microbend accelerometer, which weighs less than 1.8 g, was found to meet all performance goals including 1% linearity, 90 dB dynamic range, and a minimum detectable acceleration of 0.2 mg/√Hz . The fiber optic microphone, which weighs less than 1.3 g, also met all goals including 1% linearity, 85 dB dynamic range, and a minimum detectable acoustic pressure level of 0.016 Pa/√Hz . In addition to our specific use in active sound control, these sensors appear to have application in a variety of other areas.

Development of sound measurement systems for auditory functional magnetic resonance imaging.

PubMed

Nam, Eui-Cheol; Kim, Sam Soo; Lee, Kang Uk; Kim, Sang Sik

2008-06-01

Auditory functional magnetic resonance imaging (fMRI) requires quantification of sound stimuli in the magnetic environment and adequate isolation of background noise. We report the development of two novel sound measurement systems that accurately measure the sound intensity inside the ear, which can simultaneously provide the similar or greater amount of scanner- noise protection than ear-muffs. First, we placed a 2.6 x 2.6-mm microphone in an insert phone that was connected to a headphone [microphone-integrated, foam-tipped insert-phone with a headphone (MIHP)]. This attenuated scanner noise by 37.8+/-4.6 dB, a level better than the reference amount obtained using earmuffs. The nonmetallic optical microphone was integrated with a headphone [optical microphone in a headphone (OMHP)] and it effectively detected the change of sound intensity caused by variable compression on the cushions of the headphone. Wearing the OMHP reduced the noise by 28.5+/-5.9 dB and did not affect echoplanar magnetic resonance images. We also performed an auditory fMRI study using the MIHP system and presented increase in the auditory cortical activation following 10-dB increment in the intensity of sound stimulation. These two newly developed sound measurement systems successfully achieved the accurate quantification of sound stimuli with maintaining the similar level of noise protection of wearing earmuffs in the auditory fMRI experiment.

MEMS capacitive accelerometer-based middle ear microphone.

PubMed

Young, Darrin J; Zurcher, Mark A; Semaan, Maroun; Megerian, Cliff A; Ko, Wen H

2012-12-01

The design, implementation, and characterization of a microelectromechanical systems (MEMS) capacitive accelerometer-based middle ear microphone are presented in this paper. The microphone is intended for middle ear hearing aids as well as future fully implantable cochlear prosthesis. Human temporal bones acoustic response characterization results are used to derive the accelerometer design requirements. The prototype accelerometer is fabricated in a commercial silicon-on-insulator (SOI) MEMS process. The sensor occupies a sensing area of 1 mm × 1 mm with a chip area of 2 mm × 2.4 mm and is interfaced with a custom-designed low-noise electronic IC chip over a flexible substrate. The packaged sensor unit occupies an area of 2.5 mm × 6.2 mm with a weight of 25 mg. The sensor unit attached to umbo can detect a sound pressure level (SPL) of 60 dB at 500 Hz, 35 dB at 2 kHz, and 57 dB at 8 kHz. An improved sound detection limit of 34-dB SPL at 150 Hz and 24-dB SPL at 500 Hz can be expected by employing start-of-the-art MEMS fabrication technology, which results in an articulation index of approximately 0.76. Further micro/nanofabrication technology advancement is needed to enhance the microphone sensitivity for improved understanding of normal conversational speech.

Impedance Eduction in Ducts with Higher-Order Modes and Flow

NASA Technical Reports Server (NTRS)

Watson, Willie R.; Jones, Michael G.

2009-01-01

An impedance eduction technique, previously validated for ducts with plane waves at the source and duct termination planes, has been extended to support higher-order modes at these locations. Inputs for this method are the acoustic pressures along the source and duct termination planes, and along a microphone array located in a wall either adjacent or opposite to the test liner. A second impedance eduction technique is then presented that eliminates the need for the microphone array. The integrity of both methods is tested using three sound sources, six Mach numbers, and six selected frequencies. Results are presented for both a hardwall and a test liner (with known impedance) consisting of a perforated plate bonded to a honeycomb core. The primary conclusion of the study is that the second method performs well in the presence of higher-order modes and flow. However, the first method performs poorly when most of the microphones are located near acoustic pressure nulls. The negative effects of the acoustic pressure nulls can be mitigated by a judicious choice of the mode structure in the sound source. The paper closes by using the first impedance eduction method to design a rectangular array of 32 microphones for accurate impedance eduction in the NASA LaRC Curved Duct Test Rig in the presence of expected measurement uncertainties, higher order modes, and mean flow.

Comparison of Computational and Experimental Microphone Array Results for an 18%-Scale Aircraft Model

NASA Technical Reports Server (NTRS)

Lockard, David P.; Humphreys, William M.; Khorrami, Mehdi R.; Fares, Ehab; Casalino, Damiano; Ravetta, Patricio A.

2015-01-01

An 18%-scale, semi-span model is used as a platform for examining the efficacy of microphone array processing using synthetic data from numerical simulations. Two hybrid RANS/LES codes coupled with Ffowcs Williams-Hawkings solvers are used to calculate 97 microphone signals at the locations of an array employed in the NASA LaRC 14x22 tunnel. Conventional, DAMAS, and CLEAN-SC array processing is applied in an identical fashion to the experimental and computational results for three different configurations involving deploying and retracting the main landing gear and a part span flap. Despite the short time records of the numerical signals, the beamform maps are able to isolate the noise sources, and the appearance of the DAMAS synthetic array maps is generally better than those from the experimental data. The experimental CLEAN-SC maps are similar in quality to those from the simulations indicating that CLEAN-SC may have less sensitivity to background noise. The spectrum obtained from DAMAS processing of synthetic array data is nearly identical to the spectrum of the center microphone of the array, indicating that for this problem array processing of synthetic data does not improve spectral comparisons with experiment. However, the beamform maps do provide an additional means of comparison that can reveal differences that cannot be ascertained from spectra alone.

Detecting and Classifying Human Touches in a Social Robot Through Acoustic Sensing and Machine Learning.

PubMed

Alonso-Martín, Fernando; Gamboa-Montero, Juan José; Castillo, José Carlos; Castro-González, Álvaro; Salichs, Miguel Ángel

2017-05-16

An important aspect in Human-Robot Interaction is responding to different kinds of touch stimuli. To date, several technologies have been explored to determine how a touch is perceived by a social robot, usually placing a large number of sensors throughout the robot's shell. In this work, we introduce a novel approach, where the audio acquired from contact microphones located in the robot's shell is processed using machine learning techniques to distinguish between different types of touches. The system is able to determine when the robot is touched (touch detection), and to ascertain the kind of touch performed among a set of possibilities: stroke , tap , slap , and tickle (touch classification). This proposal is cost-effective since just a few microphones are able to cover the whole robot's shell since a single microphone is enough to cover each solid part of the robot. Besides, it is easy to install and configure as it just requires a contact surface to attach the microphone to the robot's shell and plug it into the robot's computer. Results show the high accuracy scores in touch gesture recognition. The testing phase revealed that Logistic Model Trees achieved the best performance, with an F -score of 0.81. The dataset was built with information from 25 participants performing a total of 1981 touch gestures.

Noise Reduction with Microphone Arrays for Speaker Identification

DOE Office of Scientific and Technical Information (OSTI.GOV)

Cohen, Z

Reducing acoustic noise in audio recordings is an ongoing problem that plagues many applications. This noise is hard to reduce because of interfering sources and non-stationary behavior of the overall background noise. Many single channel noise reduction algorithms exist but are limited in that the more the noise is reduced; the more the signal of interest is distorted due to the fact that the signal and noise overlap in frequency. Specifically acoustic background noise causes problems in the area of speaker identification. Recording a speaker in the presence of acoustic noise ultimately limits the performance and confidence of speaker identificationmore » algorithms. In situations where it is impossible to control the environment where the speech sample is taken, noise reduction filtering algorithms need to be developed to clean the recorded speech of background noise. Because single channel noise reduction algorithms would distort the speech signal, the overall challenge of this project was to see if spatial information provided by microphone arrays could be exploited to aid in speaker identification. The goals are: (1) Test the feasibility of using microphone arrays to reduce background noise in speech recordings; (2) Characterize and compare different multichannel noise reduction algorithms; (3) Provide recommendations for using these multichannel algorithms; and (4) Ultimately answer the question - Can the use of microphone arrays aid in speaker identification?« less

Noise reduction technologies implemented in head-worn preprocessors for improving cochlear implant performance in reverberant noise fields.

PubMed

Chung, King; Nelson, Lance; Teske, Melissa

2012-09-01

The purpose of this study was to investigate whether a multichannel adaptive directional microphone and a modulation-based noise reduction algorithm could enhance cochlear implant performance in reverberant noise fields. A hearing aid was modified to output electrical signals (ePreprocessor) and a cochlear implant speech processor was modified to receive electrical signals (eProcessor). The ePreprocessor was programmed to flat frequency response and linear amplification. Cochlear implant listeners wore the ePreprocessor-eProcessor system in three reverberant noise fields: 1) one noise source with variable locations; 2) three noise sources with variable locations; and 3) eight evenly spaced noise sources from 0° to 360°. Listeners' speech recognition scores were tested when the ePreprocessor was programmed to omnidirectional microphone (OMNI), omnidirectional microphone plus noise reduction algorithm (OMNI + NR), and adaptive directional microphone plus noise reduction algorithm (ADM + NR). They were also tested with their own cochlear implant speech processor (CI_OMNI) in the three noise fields. Additionally, listeners rated overall sound quality preferences on recordings made in the noise fields. Results indicated that ADM+NR produced the highest speech recognition scores and the most preferable rating in all noise fields. Factors requiring attention in the hearing aid-cochlear implant integration process are discussed. Copyright © 2012 Elsevier B.V. All rights reserved.

Design of a hydrophone for an Ocean World lander

NASA Astrophysics Data System (ADS)

Smith, Heather D.; Duncan, Andrew G.

2017-10-01

For this presentation we describe the science return, and design of a microphone on- board a Europa lander mission. In addition to the E/PO benefit of a hydrophone to listen to the Europa Ocean, a microphone also provides scientific data on the properties of the subsurface ocean.A hydrophone is a small light-weight instrument that could be used to achieve two of the three Europa Lander mission anticipated science goals of: 1) Asses the habitability (particularly through quantitative compositional measurements of Europa via in situ techniques uniquely available to a landed mission. And 2) Characterize surface properties at the scale of the lander to support future exploration, including the local geologic context.Acoustic properties of the ocean would lead to a better understanding of the water density, currents, seafloor topography and other physical properties of the ocean as well as lead to an understanding of the salinity of the ocean. Sound from water movement (tidal movement, currents, subsurface out-gassing, ocean homogeneity (clines), sub-surface morphology, and biological sounds.The engineering design of the hydrophone instrument will be designed to fit within a portion of the resource allocation of the current best estimates of the Europa lander payload (26.6 Kg, 24,900 cm3, 2,500 W-hrs and 2700 Mbits). The hydrophone package will be designed to ensure planetary protection is maintained and will function under the cur- rent Europa lander mission operations scenario of a two-year cruise phase, and 30-day surface operational phase on Europa.Although the microphone could be used on the surface, it is designed to be lowered into the subsurface ocean. As such, planetary protection (forward contamination) is a primary challenge for a subsurface microphone/ camera. The preliminary design is based on the Navy COTS optical microphone.Reference: Pappalardo, R. T., et al. "Science potential from a Europa lander." Astrobiology 13.8 (2013): 740-773.

Dual-microphone and binaural noise reduction techniques for improved speech intelligibility by hearing aid users

NASA Astrophysics Data System (ADS)

Yousefian Jazi, Nima

Spatial filtering and directional discrimination has been shown to be an effective pre-processing approach for noise reduction in microphone array systems. In dual-microphone hearing aids, fixed and adaptive beamforming techniques are the most common solutions for enhancing the desired speech and rejecting unwanted signals captured by the microphones. In fact, beamformers are widely utilized in systems where spatial properties of target source (usually in front of the listener) is assumed to be known. In this dissertation, some dual-microphone coherence-based speech enhancement techniques applicable to hearing aids are proposed. All proposed algorithms operate in the frequency domain and (like traditional beamforming techniques) are purely based on the spatial properties of the desired speech source and does not require any knowledge of noise statistics for calculating the noise reduction filter. This benefit gives our algorithms the ability to address adverse noise conditions, such as situations where interfering talker(s) speaks simultaneously with the target speaker. In such cases, the (adaptive) beamformers lose their effectiveness in suppressing interference, since the noise channel (reference) cannot be built and updated accordingly. This difference is the main advantage of the proposed techniques in the dissertation over traditional adaptive beamformers. Furthermore, since the suggested algorithms are independent of noise estimation, they offer significant improvement in scenarios that the power level of interfering sources are much more than that of target speech. The dissertation also shows the premise behind the proposed algorithms can be extended and employed to binaural hearing aids. The main purpose of the investigated techniques is to enhance the intelligibility level of speech, measured through subjective listening tests with normal hearing and cochlear implant listeners. However, the improvement in quality of the output speech achieved by the algorithms are also presented to show that the proposed methods can be potential candidates for future use in commercial hearing aids and cochlear implant devices.

Speech understanding in background noise with the two-microphone adaptive beamformer BEAM in the Nucleus Freedom Cochlear Implant System.

PubMed

Spriet, Ann; Van Deun, Lieselot; Eftaxiadis, Kyriaky; Laneau, Johan; Moonen, Marc; van Dijk, Bas; van Wieringen, Astrid; Wouters, Jan

2007-02-01

This paper evaluates the benefit of the two-microphone adaptive beamformer BEAM in the Nucleus Freedom cochlear implant (CI) system for speech understanding in background noise by CI users. A double-blind evaluation of the two-microphone adaptive beamformer BEAM and a hardware directional microphone was carried out with five adult Nucleus CI users. The test procedure consisted of a pre- and post-test in the lab and a 2-wk trial period at home. In the pre- and post-test, the speech reception threshold (SRT) with sentences and the percentage correct phoneme scores for CVC words were measured in quiet and background noise at different signal-to-noise ratios. Performance was assessed for two different noise configurations (with a single noise source and with three noise sources) and two different noise materials (stationary speech-weighted noise and multitalker babble). During the 2-wk trial period at home, the CI users evaluated the noise reduction performance in different listening conditions by means of the SSQ questionnaire. In addition to the perceptual evaluation, the noise reduction performance of the beamformer was measured physically as a function of the direction of the noise source. Significant improvements of both the SRT in noise (average improvement of 5-16 dB) and the percentage correct phoneme scores (average improvement of 10-41%) were observed with BEAM compared to the standard hardware directional microphone. In addition, the SSQ questionnaire and subjective evaluation in controlled and real-life scenarios suggested a possible preference for the beamformer in noisy environments. The evaluation demonstrates that the adaptive noise reduction algorithm BEAM in the Nucleus Freedom CI-system may significantly increase the speech perception by cochlear implantees in noisy listening conditions. This is the first monolateral (adaptive) noise reduction strategy actually implemented in a mainstream commercial CI.

One of many microphones arrayed under the path of the F-5E SSBE aircraft to record sonic booms

NASA Image and Video Library

2004-01-13

One of many microphones arrayed under the path of the F-5E SSBE (Shaped Sonic Boom Experiment) aircraft to record sonic booms. The SSBE (Shaped Sonic Boom Experiment) was formerly known as the Shaped Sonic Boom Demonstration, or SSBD, and is part of DARPA's Quiet Supersonic Platform (QSP) program. On August 27, 2003, the F-5E SSBD aircraft demonstrated a method to reduce the intensity of sonic booms.

Instrumentation for measuring aircraft noise and sonic boom

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J. (Inventor)

1976-01-01

Improved instrumentation suitable for measuring aircraft noise and sonic booms is described. An electric current proportional to the sound pressure level at a condenser microphone is produced and transmitted over a cable and amplified by a zero drive amplifier. The converter consists of a local oscillator, a dual-gate field-effect transistor mixer, and a voltage regulator/impedance translator. The improvements include automatic tuning compensation against changes in static microphone capacitance and means for providing a remote electrical calibration capability.

Real-time distributed fiber microphone based on phase-OTDR.

PubMed

Franciscangelis, Carolina; Margulis, Walter; Kjellberg, Leif; Soderquist, Ingemar; Fruett, Fabiano

2016-12-26

The use of an optical fiber as a real-time distributed microphone is demonstrated employing a phase-OTDR with direct detection. The method comprises a sample-and-hold circuit capable of both tuning the receiver to an arbitrary section of the fiber considered of interest and to recover in real-time the detected acoustic wave. The system allows listening to the sound of a sinusoidal disturbance with variable frequency, music and human voice with ~60 cm of spatial resolution through a 300 m long optical fiber.

Miniaturized Nanocomposite Piezoelectric Microphones for UAS Applications

DTIC Science & Technology

2012-10-22

volume fraction for three different materials: ZnO/SU-8 composite, ZnO thin film, and PZT thin film. This was computed for a microphone of outer...radius, 2 400R mμ= , and a thickness 1t mμ= . Note the significant increase in sensitivity compared to a solid ZnO or PZT film. This arises because, as...predicted range. An optimal volume fraction of 0.3 yielded a 17-fold increase in sensitivity over ZnO and a 49-fold increase over PZT . Figure 6

Optical microphone

DOEpatents

Veligdan, James T.

2000-01-11

An optical microphone includes a laser and beam splitter cooperating therewith for splitting a laser beam into a reference beam and a signal beam. A reflecting sensor receives the signal beam and reflects it in a plurality of reflections through sound pressure waves. A photodetector receives both the reference beam and reflected signal beam for heterodyning thereof to produce an acoustic signal for the sound waves. The sound waves vary the local refractive index in the path of the signal beam which experiences a Doppler frequency shift directly analogous with the sound waves.

Method of fan sound mode structure determination computer program user's manual: Modal calculation program

NASA Technical Reports Server (NTRS)

Pickett, G. F.; Wells, R. A.; Love, R. A.

1977-01-01

A computer user's manual describing the operation and the essential features of the Modal Calculation Program is presented. The modal Calculation Program calculates the amplitude and phase of modal structures by means of acoustic pressure measurements obtained from microphones placed at selected locations within the fan inlet duct. In addition, the Modal Calculation Program also calculates the first-order errors in the modal coefficients that are due to tolerances in microphone location coordinates and inaccuracies in the acoustic pressure measurements.

Directional acoustic measurements by laser Doppler velocimeters. [for jet aircraft noise

NASA Technical Reports Server (NTRS)

Mazumder, M. K.; Overbey, R. L.; Testerman, M. K.

1976-01-01

Laser Doppler velocimeters (LDVs) were used as velocity microphones to measure sound pressure level in the range of 90-130 db, spectral components, and two-point cross correlation functions for acoustic noise source identification. Close agreement between LDV and microphone data is observed. It was concluded that directional sensitivity and the ability to measure remotely make LDVs useful tools for acoustic measurement where placement of any physical probe is difficult or undesirable, as in the diagnosis of jet aircraft noise.

Acoustic results of the Boeing model 360 whirl tower test

NASA Astrophysics Data System (ADS)

Watts, Michael E.; Jordan, David

1990-09-01

An evaluation is presented for whirl tower test results of the Model 360 helicopter's advanced, high-performance four-bladed composite rotor system intended to facilitate over-200-knot flight. During these performance measurements, acoustic data were acquired by seven microphones. A comparison of whirl-tower tests with theory indicate that theoretical prediction accuracies vary with both microphone position and the inclusion of ground reflection. Prediction errors varied from 0 to 40 percent of the measured signal-to-peak amplitude.

The effect of hearing aid technologies on listening in an automobile.

PubMed

Wu, Yu-Hsiang; Stangl, Elizabeth; Bentler, Ruth A; Stanziola, Rachel W

2013-06-01

Communication while traveling in an automobile often is very difficult for hearing aid users. This is because the automobile/road noise level is usually high, and listeners/drivers often do not have access to visual cues. Since the talker of interest usually is not located in front of the listener/driver, conventional directional processing that places the directivity beam toward the listener's front may not be helpful and, in fact, could have a negative impact on speech recognition (when compared to omnidirectional processing). Recently, technologies have become available in commercial hearing aids that are designed to improve speech recognition and/or listening effort in noisy conditions where talkers are located behind or beside the listener. These technologies include (1) a directional microphone system that uses a backward-facing directivity pattern (Back-DIR processing), (2) a technology that transmits audio signals from the ear with the better signal-to-noise ratio (SNR) to the ear with the poorer SNR (Side-Transmission processing), and (3) a signal processing scheme that suppresses the noise at the ear with the poorer SNR (Side-Suppression processing). The purpose of the current study was to determine the effect of (1) conventional directional microphones and (2) newer signal processing schemes (Back-DIR, Side-Transmission, and Side-Suppression) on listener's speech recognition performance and preference for communication in a traveling automobile. A single-blinded, repeated-measures design was used. Twenty-five adults with bilateral symmetrical sensorineural hearing loss aged 44 through 84 yr participated in the study. The automobile/road noise and sentences of the Connected Speech Test (CST) were recorded through hearing aids in a standard van moving at a speed of 70 mph on a paved highway. The hearing aids were programmed to omnidirectional microphone, conventional adaptive directional microphone, and the three newer schemes. CST sentences were presented from the side and back of the hearing aids, which were placed on the ears of a manikin. The recorded stimuli were presented to listeners via earphones in a sound-treated booth to assess speech recognition performance and preference with each programmed condition. Compared to omnidirectional microphones, conventional adaptive directional processing had a detrimental effect on speech recognition when speech was presented from the back or side of the listener. Back-DIR and Side-Transmission processing improved speech recognition performance (relative to both omnidirectional and adaptive directional processing) when speech was from the back and side, respectively. The performance with Side-Suppression processing was better than with adaptive directional processing when speech was from the side. The participants' preferences for a given processing scheme were generally consistent with speech recognition results. The finding that performance with adaptive directional processing was poorer than with omnidirectional microphones demonstrates the importance of selecting the correct microphone technology for different listening situations. The results also suggest the feasibility of using hearing aid technologies to provide a better listening experience for hearing aid users in automobiles. American Academy of Audiology.

Assistive hearing technologies among students with hearing impairment: factors that promote satisfaction.

PubMed

Rekkedal, Ann Mette

2012-01-01

Hearing technology can play an essential part in the education of deaf and hard-of-hearing children in inclusive schools. Few studies have examined these children's experiences with this technology. This article explores factors pertaining to children's use of and attitudes toward hearing technologies, such as hearing aids, cochlear implants, teacher-worn microphones, and student-worn microphones. The study included 153 deaf and hard-of-hearing students. All students communicated orally and were in inclusive schools from grades 5-10. The results suggest that males view hearing technology more positively than do females. Having severe hearing loss also promoted positive attitudes toward hearing aids and cochlear implants, but not toward microphones. The students with positive self-descriptions tended to be more satisfied with hearing aids or cochlear implants than the students with negative self-descriptions. The main factors promoting the use of hearing aids were severe hearing loss, positive attitudes toward hearing aids, and the sound quality of hearing aids.

Speech Enhancement of Mobile Devices Based on the Integration of a Dual Microphone Array and a Background Noise Elimination Algorithm.

PubMed

Chen, Yung-Yue

2018-05-08

Mobile devices are often used in our daily lives for the purposes of speech and communication. The speech quality of mobile devices is always degraded due to the environmental noises surrounding mobile device users. Regretfully, an effective background noise reduction solution cannot easily be developed for this speech enhancement problem. Due to these depicted reasons, a methodology is systematically proposed to eliminate the effects of background noises for the speech communication of mobile devices. This methodology integrates a dual microphone array with a background noise elimination algorithm. The proposed background noise elimination algorithm includes a whitening process, a speech modelling method and an H ₂ estimator. Due to the adoption of the dual microphone array, a low-cost design can be obtained for the speech enhancement of mobile devices. Practical tests have proven that this proposed method is immune to random background noises, and noiseless speech can be obtained after executing this denoise process.

The average direct current offset values for small digital audio recorders in an acoustically consistent environment.

PubMed

Koenig, Bruce E; Lacey, Douglas S

2014-07-01

In this research project, nine small digital audio recorders were tested using five sets of 30-min recordings at all available recording modes, with consistent audio material, identical source and microphone locations, and identical acoustic environments. The averaged direct current (DC) offset values and standard deviations were measured for 30-sec and 1-, 2-, 3-, 6-, 10-, 15-, and 30-min segments. The research found an inverse association between segment lengths and the standard deviation values and that lengths beyond 30 min may not meaningfully reduce the standard deviation values. This research supports previous studies indicating that measured averaged DC offsets should only be used for exclusionary purposes in authenticity analyses and exhibit consistent values when the general acoustic environment and microphone/recorder configurations were held constant. Measured average DC offset values from exemplar recorders may not be directly comparable to those of submitted digital audio recordings without exactly duplicating the acoustic environment and microphone/recorder configurations. © 2014 American Academy of Forensic Sciences.

Fiber Optic Microphone

NASA Technical Reports Server (NTRS)

Cho, Y. C.; George, Thomas; Norvig, Peter (Technical Monitor)

1999-01-01

Research into advanced pressure sensors using fiber-optic technology is aimed at developing compact size microphones. Fiber optic sensors are inherently immune to electromagnetic noise, and are very sensitive, light weight, and highly flexible. In FY 98, NASA researchers successfully designed and assembled a prototype fiber-optic microphone. The sensing technique employed was fiber optic Fabry-Perot interferometry. The sensing head is composed of an optical fiber terminated in a miniature ferrule with a thin, silicon-microfabricated diaphragm mounted on it. The optical fiber is a single mode fiber with a core diameter of 8 micron, with the cleaved end positioned 50 micron from the diaphragm surface. The diaphragm is made up of a 0.2 micron thick silicon nitride membrane whose inner surface is metallized with layers of 30 nm titanium, 30 nm platinum, and 0.2 micron gold for efficient reflection. The active sensing area is approximately 1.5 mm in diameter. The measured differential pressure tolerance of this diaphragm is more than 1 bar, yielding a dynamic range of more than 100 dB.

Improved methods for fan sound field determination

NASA Technical Reports Server (NTRS)

Cicon, D. E.; Sofrin, T. G.; Mathews, D. C.

1981-01-01

Several methods for determining acoustic mode structure in aircraft turbofan engines using wall microphone data were studied. A method for reducing data was devised and implemented which makes the definition of discrete coherent sound fields measured in the presence of engine speed fluctuation more accurate. For the analytical methods, algorithms were developed to define the dominant circumferential modes from full and partial circumferential arrays of microphones. Axial arrays were explored to define mode structure as a function of cutoff ratio, and the use of data taken at several constant speeds was also evaluated in an attempt to reduce instrumentation requirements. Sensitivities of the various methods to microphone density, array size and measurement error were evaluated and results of these studies showed these new methods to be impractical. The data reduction method used to reduce the effects of engine speed variation consisted of an electronic circuit which windowed the data so that signal enhancement could occur only when the speed was within a narrow range.

Genetic analysis of vertebrate sensory hair cell mechanosensation: the zebrafish circler mutants.

PubMed

Nicolson, T; Rüsch, A; Friedrich, R W; Granato, M; Ruppersberg, J P; Nüsslein-Volhard, C

1998-02-01

The molecular basis of sensory hair cell mechanotransduction is largely unknown. In order to identify genes that are essential for mechanosensory hair cell function, we characterized a group of recently isolated zebrafish motility mutants. These mutants are defective in balance and swim in circles but have no obvious morphological defects. We examined the mutants using calcium imaging of acoustic-vibrational and tactile escape responses, high resolution microscopy of sensory neuroepithelia in live larvae, and recordings of extracellular hair cell potentials (microphonics). Based on the analyses, we have identified several classes of genes. Mutations in sputnik and mariner affect hair bundle integrity. Mutant astronaut and cosmonaut hair cells have relatively normal microphonics and thus appear to affect events downstream of mechanotransduction. Mutant orbiter, mercury, and gemini larvae have normal hair cell morphology and yet do not respond to acoustic-vibrational stimuli. The microphonics of lateral line hair cells of orbiter, mercury, and gemini larvae are absent or strongly reduced. Therefore, these genes may encode components of the transduction apparatus.

Acoustic mode measurements in the inlet of a model turbofan using a continuously rotating rake: Data collection/analysis techniques

NASA Technical Reports Server (NTRS)

Hall, David G.; Heidelberg, Laurence; Konno, Kevin

1993-01-01

The rotating microphone measurement technique and data analysis procedures are documented which are used to determine circumferential and radial acoustic mode content in the inlet of the Advanced Ducted Propeller (ADP) model. Circumferential acoustic mode levels were measured at a series of radial locations using the Doppler frequency shift produced by a rotating inlet microphone probe. Radial mode content was then computed using a least squares curve fit with the measured radial distribution for each circumferential mode. The rotating microphone technique is superior to fixed-probe techniques because it results in minimal interference with the acoustic modes generated by rotor-stator interaction. This effort represents the first experimental implementation of a measuring technique developed by T. G. Sofrin. Testing was performed in the NASA Lewis Low Speed Anechoic Wind Tunnel at a simulated takeoff condition of Mach 0.2. The design is included of the data analysis software and the performance of the rotating rake apparatus. The effect of experiment errors is also discussed.

Acoustic mode measurements in the inlet of a model turbofan using a continuously rotating rake - Data collection/analysis techniques

NASA Technical Reports Server (NTRS)

Hall, David G.; Heidelberg, Laurence; Konno, Kevin

1993-01-01

The rotating microphone measurement technique and data analysis procedures are documented which are used to determine circumferential and radial acoustic mode content in the inlet of the Advanced Ducted Propeller (ADP) model. Circumferential acoustic mode levels were measured at a series of radial locations using the Doppler frequency shift produced by a rotating inlet microphone probe. Radial mode content was then computed using a least squares curve fit with the measured radial distribution for each circumferential mode. The rotating microphone technique is superior to fixed-probe techniques because it results in minimal interference with the acoustic modes generated by rotor-stator interaction. This effort represents the first experimental implementation of a measuring technique developed by T. G. Sofrin. Testing was performed in the NASA Lewis Low Speed Anechoic Wind Tunnel at a simulated takeoff condition of Mach 0.2. The design is included of the data analysis software and the performance of the rotating rake apparatus. The effect of experiment errors is also discussed.

Electrode surface profile and the performance of condenser microphones.

PubMed

Fletcher, N H; Thwaites, S

2002-12-01

Condenser microphones of all types are traditionally made with a planar electrode parallel to an electrically conducting diaphragm, additional diaphragm stiffness at acoustic frequencies being provided by the air enclosed in a cavity behind the diaphragm. In all designs, the motion of the diaphragm in response to an acoustic signal is greatest near its center and reduces to zero at its edges. Analysis shows that this construction leads to less than optimal sensitivity and to harmonic distortion at high sound levels when the diaphragm motion is appreciable compared with its spacing from the electrode. Microphones of this design are also subject to acoustic collapse of the diaphragm under the influence of pressure pulses such as might be produced by wind. A new design is proposed in which the electrode is shaped as a shallow dish, and it is shown that this construction increases the sensitivity by about 4.5 dB, and also completely eliminates harmonic distortion originating in the cartridge.

Microphone and electroglottographic data from dysphonic patients: type 1, 2 and 3 signals.

PubMed

Behrman, A; Agresti, C J; Blumstein, E; Lee, N

1998-06-01

Recently, it has been suggested that statistics which are dependent upon the reliable extraction of a single fundamental period, such as jitter and shimmer, are valid only for nearly periodic signals. This study explored the incidence of nearly periodic and nonperiodic microphone and electroglottographic signals obtained from 202 dysphonic patients. It was found that approximately 42% were type 1 (nearly periodic); approximately 35% were type 2 (containing bifurcations, modulations or subharmonic structure); and approximately 22% were type 3 (chaotic). Discriminating between type 2 and 3 signals was very difficult for 40% of the signals which were ultimately rated type 3. This was due to the brevity of the apparently chaotic segment, and/or the persistence of some harmonic structure within the chaos. Irrespective of that difficulty, the results suggest that there may be a substantial incidence of nontype 1 signals in a given clinical population. It was concluded, therefore, that signal typing is a necessary step in the analyses of microphone and electoglottographic data.

Estimation of multiple sound sources with data and model uncertainties using the EM and evidential EM algorithms

NASA Astrophysics Data System (ADS)

Wang, Xun; Quost, Benjamin; Chazot, Jean-Daniel; Antoni, Jérôme

2016-01-01

This paper considers the problem of identifying multiple sound sources from acoustical measurements obtained by an array of microphones. The problem is solved via maximum likelihood. In particular, an expectation-maximization (EM) approach is used to estimate the sound source locations and strengths, the pressure measured by a microphone being interpreted as a mixture of latent signals emitted by the sources. This work also considers two kinds of uncertainties pervading the sound propagation and measurement process: uncertain microphone locations and uncertain wavenumber. These uncertainties are transposed to the data in the belief functions framework. Then, the source locations and strengths can be estimated using a variant of the EM algorithm, known as the Evidential EM (E2M) algorithm. Eventually, both simulation and real experiments are shown to illustrate the advantage of using the EM in the case without uncertainty and the E2M in the case of uncertain measurement.

Sound source localization on an axial fan at different operating points

NASA Astrophysics Data System (ADS)

Zenger, Florian J.; Herold, Gert; Becker, Stefan; Sarradj, Ennes

2016-08-01

A generic fan with unskewed fan blades is investigated using a microphone array method. The relative motion of the fan with respect to the stationary microphone array is compensated by interpolating the microphone data to a virtual rotating array with the same rotational speed as the fan. Hence, beamforming algorithms with deconvolution, in this case CLEAN-SC, could be applied. Sound maps and integrated spectra of sub-components are evaluated for five operating points. At selected frequency bands, the presented method yields sound maps featuring a clear circular source pattern corresponding to the nine fan blades. Depending on the adjusted operating point, sound sources are located on the leading or trailing edges of the fan blades. Integrated spectra show that in most cases leading edge noise is dominant for the low-frequency part and trailing edge noise for the high-frequency part. The shift from leading to trailing edge noise is strongly dependent on the operating point and frequency range considered.

Development and use of a spherical microphone array for measurement of spatial properties of reverberant sound fields

NASA Astrophysics Data System (ADS)

Gover, Bradford Noel

The problem of hands-free speech pick-up is introduced, and it is identified how details of the spatial properties of the reverberant field may be useful for enhanced design of microphone arrays. From this motivation, a broadly-applicable measurement system has been developed for the analysis of the directional and spatial variations in reverberant sound fields. Two spherical, 32-element arrays of microphones are used to generate narrow beams over two different frequency ranges, together covering 300--3300 Hz. Using an omnidirectional loudspeaker as excitation in a room, the pressure impulse response in each of 60 steering directions is measured. Through analysis of these responses, the variation of arriving energy with direction is studied. The system was first validated in simple sound fields in an anechoic chamber and in a reverberation chamber. The system characterizes these sound fields as expected, both quantitatively through numerical descriptors and qualitatively from plots of the arriving energy versus direction. The system was then used to measure the sound fields in several actual rooms. Through both qualitative and quantitative output, these sound fields were seen to be highly anisotropic, influenced greatly by the direct sound and early-arriving reflections. Furthermore, the rate of sound decay was not independent of direction, sound being absorbed more rapidly in some directions than in others. These results are discussed in the context of the original motivation, and methods for their application to enhanced speech pick-up using microphone arrays are proposed.

Analysis of jet-airfoil interaction noise sources by using a microphone array technique

NASA Astrophysics Data System (ADS)

Fleury, Vincent; Davy, Renaud

2016-03-01

The paper is concerned with the characterization of jet noise sources and jet-airfoil interaction sources by using microphone array data. The measurements were carried-out in the anechoic open test section wind tunnel of Onera, Cepra19. The microphone array technique relies on the convected, Lighthill's and Ffowcs-Williams and Hawkings' acoustic analogy equation. The cross-spectrum of the source term of the analogy equation is sought. It is defined as the optimal solution to a minimal error equation using the measured microphone cross-spectra as reference. This inverse problem is ill-posed yet. A penalty term based on a localization operator is therefore added to improve the recovery of jet noise sources. The analysis of isolated jet noise data in subsonic regime shows the contribution of the conventional mixing noise source in the low frequency range, as expected, and of uniformly distributed, uncorrelated noise sources in the jet flow at higher frequencies. In underexpanded supersonic regime, a shock-associated noise source is clearly identified, too. An additional source is detected in the vicinity of the nozzle exit both in supersonic and subsonic regimes. In the presence of the airfoil, the distribution of the noise sources is deeply modified. In particular, a strong noise source is localized on the flap. For high Strouhal numbers, higher than about 2 (based on the jet mixing velocity and diameter), a significant contribution from the shear-layer near the flap is observed, too. Indications of acoustic reflections on the airfoil are also discerned.

Measurement and Characterization of Helicopter Noise in Steady-State and Maneuvering Flight

NASA Technical Reports Server (NTRS)

Schmitz, Fredric H.; Greenwood, Eric; Sickenberger, Richard D.; Gopalan, Gaurav; Sim, Ben Well-C; Conner, David; Moralez, Ernesto; Decker, William A.

2007-01-01

A special acoustic flight test program was performed on the Bell 206B helicopter outfitted with an in-flight microphone boom/array attached to the helicopter while simultaneous acoustic measurements were made using a linear ground array of microphones arranged to be perpendicular to the flight path. Air and ground noise measurements were made in steady-state longitudinal and steady turning flight, and during selected dynamic maneuvers. Special instrumentation, including direct measurement of the helicopter s longitudinal tip-path-plane (TPP) angle, Differential Global Positioning System (DGPS) and Inertial Navigation Unit (INU) measurements, and a pursuit guidance display were used to measure important noise controlling parameters and to make the task of flying precise operating conditions and flight track easier for the pilot. Special care was also made to test only in very low winds. The resulting acoustic data is of relatively high quality and shows the value of carefully monitoring and controlling the helicopter s performance state. This paper has shown experimentally, that microphones close to the helicopter can be used to estimate the specific noise sources that radiate to the far field, if the microphones are positioned correctly relative to the noise source. Directivity patterns for steady, turning flight were also developed, for the first time, and connected to the turning performance of the helicopter. Some of the acoustic benefits of combining normally separated flight segments (i.e. an accelerated segment and a descending segment) were also demonstrated.

Sonic Booms on Big Structures (SonicBOBS) Phase I Database; NASA Dryden Sensors

NASA Technical Reports Server (NTRS)

Haering, Edward A., Jr.; Arnac, Sarah Renee

2010-01-01

This DVD contains 13 channels of microphone and up to 22 channels of pressure transducer data collected in September, 2009 around several buildings located at Edwards Air Force Base. These data were recorded by NASA Dryden. Not included are data taken by NASA Langley and Gulfstream. Each day's data is in a separate folder and each pass is in a file beginning with "SonicBOBS_" (for microphone data) or "SonicBOBSBB_" (for BADS and BASS data) followed by the month, day, year as two digits each, followed by the hour, minute, sec after midnight GMT. The filename time given is for the END time of the raw recording file. In the case of the microphone data, this time may be several minutes after the sonic boom, and is according to the PC's uncalibrated clock. The Matlab data files have the actual time as provided by a GPS-based IRIG-B signal recorded concurrently with the data. Microphone data is given for 5 seconds prior to 20 seconds after the sonic boom. BADS and BASS data is given for the full recording, 6 seconds for the BADS and 10 seconds for the BASS. As an example of the naming convention, file "SonicBOBS_091209154618.mat" is from September 12, 2009 at 15:46:18 GMT. Note that data taken on September 12, 2009 prior to 01:00:00 GMT was of the Space Shuttle Discovery (a sonic boom of opportunity), which was on September 11, 2009 in local Pacific Daylight Time.

Identification of impact force acting on composite laminated plates using the radiated sound measured with microphones

NASA Astrophysics Data System (ADS)

Atobe, Satoshi; Nonami, Shunsuke; Hu, Ning; Fukunaga, Hisao

2017-09-01

Foreign object impact events are serious threats to composite laminates because impact damage leads to significant degradation of the mechanical properties of the structure. Identification of the location and force history of the impact that was applied to the structure can provide useful information for assessing the structural integrity. This study proposes a method for identifying impact forces acting on CFRP (carbon fiber reinforced plastic) laminated plates on the basis of the sound radiated from the impacted structure. Identification of the impact location and force history is performed using the sound pressure measured with microphones. To devise a method for identifying the impact location from the difference in the arrival times of the sound wave detected with the microphones, the propagation path of the sound wave from the impacted point to the sensor is examined. For the identification of the force history, an experimentally constructed transfer matrix is employed to relate the force history to the corresponding sound pressure. To verify the validity of the proposed method, impact tests are conducted by using a CFRP cross-ply laminate as the specimen, and an impulse hammer as the impactor. The experimental results confirm the validity of the present method for identifying the impact location from the arrival time of the sound wave detected with the microphones. Moreover, the results of force history identification show the feasibility of identifying the force history accurately from the measured sound pressure using the experimental transfer matrix.

Acoustic Detection Of Loose Particles In Pressure Sensors

NASA Technical Reports Server (NTRS)

Kwok, Lloyd C.

1995-01-01

Particle-impact-noise-detector (PIND) apparatus used in conjunction with computer program analyzing output of apparatus to detect extraneous particles trapped in pressure sensors. PIND tester essentially shaker equipped with microphone measuring noise in pressure sensor or other object being shaken. Shaker applies controlled vibration. Output of microphone recorded and expressed in terms of voltage, yielding history of noise subsequently processed by computer program. Data taken at sampling rate sufficiently high to enable identification of all impacts of particles on sensor diaphragm and on inner surfaces of sensor cavities.

A low-cost 3-D printed stethoscope connected to a smartphone.

PubMed

Aguilera-Astudillo, Carlos; Chavez-Campos, Marx; Gonzalez-Suarez, Alan; Garcia-Cordero, Jose L

2016-08-01

We demonstrate the fabrication of a digital stethoscope using a 3D printer and commercial off-the-shelf electronics. A chestpiece consists of an electret microphone embedded into the drum of a 3D printed chestpiece. An electronic dongle amplifies the signal from the microphone and reduces any external noise. It also adjusts the signal to the voltages accepted by the smartphones headset jack. A graphical user interface programmed in Android displays the signals processed by the dongle. The application also saves the processed signal and sends it to a physician.

Advanced Noise Control Fan: A 20-Year Retrospective

NASA Technical Reports Server (NTRS)

Sutliff, Dan

2016-01-01

The ANCF test bed is used for evaluating fan noise reduction concepts, developing noise measurement technologies, and providing a database for Aero-acoustic code development. Rig Capabilities: 4 foot 16 bladed rotor @ 2500 rpm, Auxiliary air delivery system (3 lbm/sec @ 6/12 psi), Variable configuration (rotor pitch angle, stator count/position, duct length), synthetic acoustic noise generation (tone/broadband). Measurement Capabilities: 112 channels dynamic data system, Unique rotating rake mode measuremen, Farfield (variable radius), Duct wall microphones, Stator vane microphones, Two component CTA w/ traversing, ESP for static pressures.

The effect of hearing aid technologies on listening in an automobile

PubMed Central

Wu, Yu-Hsiang; Stangl, Elizabeth; Bentler, Ruth A.; Stanziola, Rachel W.

2014-01-01

Background Communication while traveling in an automobile often is very difficult for hearing aid users. This is because the automobile /road noise level is usually high, and listeners/drivers often do not have access to visual cues. Since the talker of interest usually is not located in front of the driver/listener, conventional directional processing that places the directivity beam toward the listener’s front may not be helpful, and in fact, could have a negative impact on speech recognition (when compared to omnidirectional processing). Recently, technologies have become available in commercial hearing aids that are designed to improve speech recognition and/or listening effort in noisy conditions where talkers are located behind or beside the listener. These technologies include (1) a directional microphone system that uses a backward-facing directivity pattern (Back-DIR processing), (2) a technology that transmits audio signals from the ear with the better signal-to-noise ratio (SNR) to the ear with the poorer SNR (Side-Transmission processing), and (3) a signal processing scheme that suppresses the noise at the ear with the poorer SNR (Side-Suppression processing). Purpose The purpose of the current study was to determine the effect of (1) conventional directional microphones and (2) newer signal processing schemes (Back-DIR, Side-Transmission, and Side-Suppression) on listener’s speech recognition performance and preference for communication in a traveling automobile. Research design A single-blinded, repeated-measures design was used. Study Sample Twenty-five adults with bilateral symmetrical sensorineural hearing loss aged 44 through 84 years participated in the study. Data Collection and Analysis The automobile/road noise and sentences of the Connected Speech Test (CST) were recorded through hearing aids in a standard van moving at a speed of 70 miles/hour on a paved highway. The hearing aids were programmed to omnidirectional microphone, conventional adaptive directional microphone, and the three newer schemes. CST sentences were presented from the side and back of the hearing aids, which were placed on the ears of a manikin. The recorded stimuli were presented to listeners via earphones in a sound treated booth to assess speech recognition performance and preference with each programmed condition. Results Compared to omnidirectional microphones, conventional adaptive directional processing had a detrimental effect on speech recognition when speech was presented from the back or side of the listener. Back-DIR and Side-Transmission processing improved speech recognition performance (relative to both omnidirectional and adaptive directional processing) when speech was from the back and side, respectively. The performance with Side-Suppression processing was better than with adaptive directional processing when speech was from the side. The participants’ preferences for a given processing scheme were generally consistent with speech recognition results. Conclusions The finding that performance with adaptive directional processing was poorer than with omnidirectional microphones demonstrates the importance of selecting the correct microphone technology for different listening situations. The results also suggest the feasibility of using hearing aid technologies to provide a better listening experience for hearing aid users in automobiles. PMID:23886425

First Test of Fan Active Noise Control (ANC) Completed

NASA Technical Reports Server (NTRS)

2005-01-01

With the advent of ultrahigh-bypass engines, the space available for passive acoustic treatment is becoming more limited, whereas noise regulations are becoming more stringent. Active noise control (ANC) holds promise as a solution to this problem. It uses secondary (added) noise sources to reduce or eliminate the offending noise radiation. The first active noise control test on the low-speed fan test bed was a General Electric Company system designed to control either the exhaust or inlet fan tone. This system consists of a "ring source," an induct array of error microphones, and a control computer. Fan tone noise propagates in a duct in the form of spinning waves. These waves are detected by the microphone array, and the computer identifies their spinning structure. The computer then controls the "ring source" to generate waves that have the same spinning structure and amplitude, but 180 out of phase with the fan noise. This computer generated tone cancels the fan tone before it radiates from the duct and is heard in the far field. The "ring source" used in these tests is a cylindrical array of 16 flat-plate acoustic radiators that are driven by thin piezoceramic sheets bonded to their back surfaces. The resulting source can produce spinning waves up to mode 7 at levels high enough to cancel the fan tone. The control software is flexible enough to work on spinning mode orders from -6 to 6. In this test, the fan was configured to produce a tone of order 6. The complete modal (spinning and radial) structure of the tones was measured with two builtin sets of rotating microphone rakes. These rakes provide a measurement of the system performance independent from the control system error microphones. In addition, the far-field noise was measured with a semicircular array of 28 microphones. This test represents the first in a series of tests that demonstrate different active noise control concepts, each on a progressively more complicated modal structure. The tests are in preparation for a demonstration on a flight-type engine.

Development of an automated speech recognition interface for personal emergency response systems

PubMed Central

Hamill, Melinda; Young, Vicky; Boger, Jennifer; Mihailidis, Alex

2009-01-01

Background Demands on long-term-care facilities are predicted to increase at an unprecedented rate as the baby boomer generation reaches retirement age. Aging-in-place (i.e. aging at home) is the desire of most seniors and is also a good option to reduce the burden on an over-stretched long-term-care system. Personal Emergency Response Systems (PERSs) help enable older adults to age-in-place by providing them with immediate access to emergency assistance. Traditionally they operate with push-button activators that connect the occupant via speaker-phone to a live emergency call-centre operator. If occupants do not wear the push button or cannot access the button, then the system is useless in the event of a fall or emergency. Additionally, a false alarm or failure to check-in at a regular interval will trigger a connection to a live operator, which can be unwanted and intrusive to the occupant. This paper describes the development and testing of an automated, hands-free, dialogue-based PERS prototype. Methods The prototype system was built using a ceiling mounted microphone array, an open-source automatic speech recognition engine, and a 'yes' and 'no' response dialog modelled after an existing call-centre protocol. Testing compared a single microphone versus a microphone array with nine adults in both noisy and quiet conditions. Dialogue testing was completed with four adults. Results and discussion The microphone array demonstrated improvement over the single microphone. In all cases, dialog testing resulted in the system reaching the correct decision about the kind of assistance the user was requesting. Further testing is required with elderly voices and under different noise conditions to ensure the appropriateness of the technology. Future developments include integration of the system with an emergency detection method as well as communication enhancement using features such as barge-in capability. Conclusion The use of an automated dialog-based PERS has the potential to provide users with more autonomy in decisions regarding their own health and more privacy in their own home. PMID:19583876

Llamas: Large-area microphone arrays and sensing systems

NASA Astrophysics Data System (ADS)

Sanz-Robinson, Josue

Large-area electronics (LAE) provides a platform to build sensing systems, based on distributing large numbers of densely spaced sensors over a physically-expansive space. Due to their flexible, "wallpaper-like" form factor, these systems can be seamlessly deployed in everyday spaces. They go beyond just supplying sensor readings, but rather they aim to transform the wealth of data from these sensors into actionable inferences about our physical environment. This requires vertically integrated systems that span the entirety of the signal processing chain, including transducers and devices, circuits, and signal processing algorithms. To this end we develop hybrid LAE / CMOS systems, which exploit the complementary strengths of LAE, enabling spatially distributed sensors, and CMOS ICs, providing computational capacity for signal processing. To explore the development of hybrid sensing systems, based on vertical integration across the signal processing chain, we focus on two main drivers: (1) thin-film diodes, and (2) microphone arrays for blind source separation: 1) Thin-film diodes are a key building block for many applications, such as RFID tags or power transfer over non-contact inductive links, which require rectifiers for AC-to-DC conversion. We developed hybrid amorphous / nanocrystalline silicon diodes, which are fabricated at low temperatures (<200 °C) to be compatible with processing on plastic, and have high current densities (5 A/cm2 at 1 V) and high frequency operation (cutoff frequency of 110 MHz). 2) We designed a system for separating the voices of multiple simultaneous speakers, which can ultimately be fed to a voice-command recognition engine for controlling electronic systems. On a device level, we developed flexible PVDF microphones, which were used to create a large-area microphone array. On a circuit level we developed localized a-Si TFT amplifiers, and a custom CMOS IC, for system control, sensor readout and digitization. On a signal processing level we developed an algorithm for blind source separation in a real, reverberant room, based on beamforming and binary masking. It requires no knowledge about the location of the speakers or microphones. Instead, it uses cluster analysis techniques to determine the time delays for beamforming; thus, adapting to the unique acoustic environment of the room.

The Effects of Hearing Aid Directional Microphone and Noise Reduction Processing on Listening Effort in Older Adults with Hearing Loss.

PubMed

Desjardins, Jamie L

2016-01-01

Older listeners with hearing loss may exert more cognitive resources to maintain a level of listening performance similar to that of younger listeners with normal hearing. Unfortunately, this increase in cognitive load, which is often conceptualized as increased listening effort, may come at the cost of cognitive processing resources that might otherwise be available for other tasks. The purpose of this study was to evaluate the independent and combined effects of a hearing aid directional microphone and a noise reduction (NR) algorithm on reducing the listening effort older listeners with hearing loss expend on a speech-in-noise task. Participants were fitted with study worn commercially available behind-the-ear hearing aids. Listening effort on a sentence recognition in noise task was measured using an objective auditory-visual dual-task paradigm. The primary task required participants to repeat sentences presented in quiet and in a four-talker babble. The secondary task was a digital visual pursuit rotor-tracking test, for which participants were instructed to use a computer mouse to track a moving target around an ellipse that was displayed on a computer screen. Each of the two tasks was presented separately and concurrently at a fixed overall speech recognition performance level of 50% correct with and without the directional microphone and/or the NR algorithm activated in the hearing aids. In addition, participants reported how effortful it was to listen to the sentences in quiet and in background noise in the different hearing aid listening conditions. Fifteen older listeners with mild sloping to severe sensorineural hearing loss participated in this study. Listening effort in background noise was significantly reduced with the directional microphones activated in the hearing aids. However, there was no significant change in listening effort with the hearing aid NR algorithm compared to no noise processing. Correlation analysis between objective and self-reported ratings of listening effort showed no significant relation. Directional microphone processing effectively reduced the cognitive load of listening to speech in background noise. This is significant because it is likely that listeners with hearing impairment will frequently encounter noisy speech in their everyday communications. American Academy of Audiology.

Lightning Location Using Acoustic Signals

NASA Astrophysics Data System (ADS)

Badillo, E.; Arechiga, R. O.; Thomas, R. J.

2013-05-01

In the summer of 2011 and 2012 a network of acoustic arrays was deployed in the Magdalena mountains of central New Mexico to locate lightning flashes. A Times-Correlation (TC) ray-tracing-based-technique was developed in order to obtain the location of lightning flashes near the network. The TC technique, locates acoustic sources from lightning. It was developed to complement the lightning location of RF sources detected by the Lightning Mapping Array (LMA) developed at Langmuir Laboratory, in New Mexico Tech. The network consisted of four arrays with four microphones each. The microphones on each array were placed in a triangular configuration with one of the microphones in the center of the array. The distance between the central microphone and the rest of them was about 30 m. The distance between centers of the arrays ranged from 500 m to 1500 m. The TC technique uses times of arrival (TOA) of acoustic waves to trace back the location of thunder sources. In order to obtain the times of arrival, the signals were filtered in a frequency band of 2 to 20 hertz and cross-correlated. Once the times of arrival were obtained, the Levenberg-Marquardt algorithm was applied to locate the spatial coordinates (x,y, and z) of thunder sources. Two techniques were used and contrasted to compute the accuracy of the TC method: Nearest-Neighbors (NN), between acoustic and LMA located sources, and standard deviation from the curvature matrix of the system as a measure of dispersion of the results. For the best case scenario, a triggered lightning event, the TC method applied with four microphones, located sources with a median error of 152 m and 142.9 m using nearest-neighbors and standard deviation respectively.; Results of the TC method in the lightning event recorded at 18:47:35 UTC, August 6, 2012. Black dots represent the results computed. Light color dots represent the LMA data for the same event. The results were obtained with the MGTM station (four channels). This figure shows a map of Altitude vs Longitude (in km).

Wireless and acoustic hearing with bone-anchored hearing devices.

PubMed

Bosman, Arjan J; Mylanus, Emmanuel A M; Hol, Myrthe K S; Snik, Ad F M

2015-07-01

The efficacy of wireless connectivity in bone-anchored hearing was studied by comparing the wireless and acoustic performance of the Ponto Plus sound processor from Oticon Medical relative to the acoustic performance of its predecessor, the Ponto Pro. Nineteen subjects with more than two years' experience with a bone-anchored hearing device were included. Thirteen subjects were fitted unilaterally and six bilaterally. Subjects served as their own control. First, subjects were tested with the Ponto Pro processor. After a four-week acclimatization period performance the Ponto Plus processor was measured. In the laboratory wireless and acoustic input levels were made equal. In daily life equal settings of wireless and acoustic input were used when watching TV, however when using the telephone the acoustic input was reduced by 9 dB relative to the wireless input. Speech scores for microphone with Ponto Pro and for both input modes of the Ponto Plus processor were essentially equal when equal input levels of wireless and microphone inputs were used. Only the TV-condition showed a statistically significant (p <5%) lower speech reception threshold for wireless relative to microphone input. In real life, evaluation of speech quality, speech intelligibility in quiet and noise, and annoyance by ambient noise, when using landline phone, mobile telephone, and watching TV showed a clear preference (p <1%) for the Ponto Plus system with streamer over the microphone input. Due to the small number of respondents with landline phone (N = 7) the result for noise annoyance was only significant at the 5% level. Equal input levels for acoustic and wireless inputs results in equal speech scores, showing a (near) equivalence for acoustic and wireless sound transmission with Ponto Pro and Ponto Plus. The default 9-dB difference between microphone and wireless input when using the telephone results in a substantial wireless benefit when using the telephone. The preference of wirelessly transmitted audio when watching TV can be attributed to the relatively poor sound quality of backward facing loudspeakers in flat screen TVs. The ratio of wireless and acoustic input can be easily set to the user's preference with the streamer's volume control.

A Unique Test Facility to Measure Liner Performance with a Summary of Initial Test Results

NASA Technical Reports Server (NTRS)

Ahuja, K. K.; Gaeta, R. J., Jr.

1997-01-01

A very ambitious study was initiated to obtain detailed acoustic and flow data with and without a liner in a duct containing a mean flow so that available theoretical models of duct liners can be validated. A unique flow-duct facility equipped with a sound source, liner box, flush-walled microphones, traversable microphones and traversable pressure and temperature probes was built. A unique set of instrumentation boxes equipped with computer controlled traverses were designed and built that allowed measurements of Mach number, temperature, SPLs and phases in two planes upstream of a liner section and two planes downstream at a large number of measurement points. Each pair of planes provided acoustic pressure gradients for use in estimating the particle velocities. Specially-built microphone probes were employed to make measurements in the presence of the flow. A microphone traverse was also designed to measure the distribution of SPLs and phases from the beginning of the liner to its end along the duct axis. All measurements were made with the help of cross-correlation techniques to reject flow noise and/or other obtrusive noise, if any. The facility was designed for future use at temperatures as high as 1500 F. In order to validate 2-D models in the presence of mean flow, the flow duct was equipped with a device to modify boundary layer flow on the smaller sides of a rectangular duct to simulate 2-D flow. A massive amount of data was acquired for use in validating duct liner models and will be provided to NASA in an electronic form. It was found that the sound in the plane-wave regime is well behaved within the duct and the results are repeatable from one run to another. At the higher frequencies corresponding to the higher-order modes, the SPLs within a duct are not repeatable from run to run. In fact, when two or more modes have the same frequency (i.e., for the degenerate modes), the SPLs in the duct varied between 2 dB to 12 dB from run to run. This made the calibration of the microphone probes extremely difficult at the higher frequencies.

Use of a Microphone Phased Array to Determine Noise Sources in a Rocket Plume

NASA Technical Reports Server (NTRS)

Panda, J.; Mosher, R.

2010-01-01

A 70-element microphone phased array was used to identify noise sources in the plume of a solid rocket motor. An environment chamber was built and other precautions were taken to protect the sensitive condenser microphones from rain, thunderstorms and other environmental elements during prolonged stay in the outdoor test stand. A camera mounted at the center of the array was used to photograph the plume. In the first phase of the study the array was placed in an anechoic chamber for calibration, and validation of the indigenous Matlab(R) based beamform software. It was found that the "advanced" beamform methods, such as CLEAN-SC was partially successful in identifying speaker sources placed closer than the Rayleigh criteria. To participate in the field test all equipments were shipped to NASA Marshal Space Flight Center, where the elements of the array hardware were rebuilt around the test stand. The sensitive amplifiers and the data acquisition hardware were placed in a safe basement, and 100m long cables were used to connect the microphones, Kulites and the camera. The array chamber and the microphones were found to withstand the environmental elements as well as the shaking from the rocket plume generated noise. The beamform map was superimposed on a photo of the rocket plume to readily identify the source distribution. It was found that the plume made an exceptionally long, >30 diameter, noise source over a large frequency range. The shock pattern created spatial modulation of the noise source. Interestingly, the concrete pad of the horizontal test stand was found to be a good acoustic reflector: the beamform map showed two distinct source distributions- the plume and its reflection on the pad. The array was found to be most effective in the frequency range of 2kHz to 10kHz. As expected, the classical beamform method excessively smeared the noise sources at lower frequencies and produced excessive side-lobes at higher frequencies. The "advanced" beamform routine CLEAN-SC created a series of lumped sources which may be unphysical. We believe that the present effort is the first-ever attempt to directly measure noise source distribution in a rocket plume.

A study of the real-world noise attenuation of the current hearing protection devices in typical workplaces using Field Microphone in Real Ear method.

PubMed

Aliabadi, Mohsen; Biabani, Azam; Golmohammadi, Rostam; Farhadian, Maryam

2018-05-28

Actual noise reduction of the earmuffs is considered as one of the main challenges for the evaluation of the effectiveness of a hearing conservation program. The current study aimed to determine the real world noise attenuation of current hearing protection devices in typical workplaces using a field microphone in real ear (FMIRE) method. In this cross-sectional study, five common earmuffs were investigated among 50 workers in two industrial factories with different noise characteristics. Noise reduction data was measured with the use of earmuffs based on the ISO 11904 standard, field microphone in real ear method, using noise dosimeter (SVANTEK, model SV 102) equipped with a microphone SV 25 model. The actual insertion losses (IL) of the tested earmuffs in octave band were lower than the labeled insertion loss data (p <  0.05). The frequency nature of noise to which workers are exposed has noticeable effects on the actual noise reduction of earmuffs (p <  0.05). The results suggest that the proportion of time using earmuffs has a considerable impact on the effective noise reduction during the workday. Data about the ambient noise characteristics is a key criterion when evaluating the acoustic performance of hearing protectors in any workplaces. Comfort aspects should be considered as one of the most important criteria for long-term use and effective wearing of hearing protection devices. FMIRE could facilitate rapid and simple measurement of the actual performance of the current earmuffs employed by workers during different work activities.

The use of cochlear's SCAN and wireless microphones to improve speech understanding in noise with the Nucleus6® CP900 processor.

PubMed

De Ceulaer, Geert; Pascoal, David; Vanpoucke, Filiep; Govaerts, Paul J

2017-11-01

The newest Nucleus CI processor, the CP900, has two new options to improve speech-in-noise perception: (1) use of an adaptive directional microphone (SCAN mode) and (2) wireless connection to MiniMic1 and MiniMic2 wireless remote microphones. An analysis was made of the absolute and relative benefits of these technologies in a real-world mimicking test situation. Speech perception was tested using an adaptive speech-in-noise test (sentences-in-babble noise). In session A, SRTs were measured in three conditions: (1) Clinical Map, (2) SCAN and (3) MiniMic1. Each was assessed for three distances between speakers and CI recipient: 1 m, 2 m and 3 m. In session B, the benefit of the use of MiniMic2 was compared to benefit of MiniMic1 at 3 m. A group of 13 adult CP900 recipients participated. SCAN and MiniMic1 improved performance compared to the standard microphone with a median improvement in SRT of 2.7-3.9 dB for SCAN at 1 m and 3 m, respectively, and 4.7-10.9 dB for the MiniMic1. MiniMic1 improvements were significant. MiniMic2 showed an improvement in SRT of 22.2 dB compared to 10.0 dB for MiniMic1 (3 m). Digital wireless transmission systems (i.e. MiniMic) offer a statistically and clinically significant improvement in speech perception in challenging, realistic listening conditions.

Speaker diarization system on the 2007 NIST rich transcription meeting recognition evaluation

NASA Astrophysics Data System (ADS)

Sun, Hanwu; Nwe, Tin Lay; Koh, Eugene Chin Wei; Bin, Ma; Li, Haizhou

2007-09-01

This paper presents a speaker diarization system developed at the Institute for Infocomm Research (I2R) for NIST Rich Transcription 2007 (RT-07) evaluation task. We describe in details our primary approaches for the speaker diarization on the Multiple Distant Microphones (MDM) conditions in conference room scenario. Our proposed system consists of six modules: 1). Least-mean squared (NLMS) adaptive filter for the speaker direction estimate via Time Difference of Arrival (TDOA), 2). An initial speaker clustering via two-stage TDOA histogram distribution quantization approach, 3). Multiple microphone speaker data alignment via GCC-PHAT Time Delay Estimate (TDE) among all the distant microphone channel signals, 4). A speaker clustering algorithm based on GMM modeling approach, 5). Non-speech removal via speech/non-speech verification mechanism and, 6). Silence removal via "Double-Layer Windowing"(DLW) method. We achieves error rate of 31.02% on the 2006 Spring (RT-06s) MDM evaluation task and a competitive overall error rate of 15.32% for the NIST Rich Transcription 2007 (RT-07) MDM evaluation task.

Multi-microphone adaptive array augmented with visual cueing.

PubMed

Gibson, Paul L; Hedin, Dan S; Davies-Venn, Evelyn E; Nelson, Peggy; Kramer, Kevin

2012-01-01

We present the development of an audiovisual array that enables hearing aid users to converse with multiple speakers in reverberant environments with significant speech babble noise where their hearing aids do not function well. The system concept consists of a smartphone, a smartphone accessory, and a smartphone software application. The smartphone accessory concept is a multi-microphone audiovisual array in a form factor that allows attachment to the back of the smartphone. The accessory will also contain a lower power radio by which it can transmit audio signals to compatible hearing aids. The smartphone software application concept will use the smartphone's built in camera to acquire images and perform real-time face detection using the built-in face detection support of the smartphone. The audiovisual beamforming algorithm uses the location of talking targets to improve the signal to noise ratio and consequently improve the user's speech intelligibility. Since the proposed array system leverages a handheld consumer electronic device, it will be portable and low cost. A PC based experimental system was developed to demonstrate the feasibility of an audiovisual multi-microphone array and these results are presented.

KEY COMPARISON: Report on the Regional Comparison COOMET.AUV.A-K3

NASA Astrophysics Data System (ADS)

Barrera-Figueroa, Salvador; Nielsen, Lars; Rasmussen, Knud

2007-01-01

COOMET.AUV.A-K3 is a Regional Comparison that supplements the Key Comparison CCAUV.A-K3 organized by the CCAUV. The participating NMIs are GUM (Poland), INM (Romania), VNIIFTRI (Russia) and DP-NDI 'Systema' (Ukraine). The role of Pilot laboratory was undertaken by DPLADFM (Denmark). The measurements took place between May 2005 and February 2006. The time schedule was organized in a single star configuration. Initially, two LS2aP microphones were circulated. However, a sudden change of sensitivity of one of them forced the inclusion of an additional microphone. Nevertheless, the analysis was performed on all microphones involved. This report includes the measurement results from the participants, information about their calibration methods, and the analysis leading to the assignation of degrees of equivalence and the link to the CCAUV.A-K3. Main text. To reach the main text of this paper, click on Final Report. The final report has been peer-reviewed and approved for publication by the CCAUV, according to the provisions of the CIPM Mutual Recognition Arrangement (MRA).

Measurement Of Trailing Edge Noise using Directional Array and Coherent Output Power Methods

NASA Technical Reports Server (NTRS)

Hutcheson, Florence V.; Brooks, Thomas F.

2002-01-01

The use of a directional array of microphones for the measurement of trailing edge (TE) noise is described. The capabilities of this method are evaluated via measurements of TE noise from a NACA 63-215 airfoil model and from a cylindrical rod. This TE noise measurement approach is compared to one that is based on the cross spectral analysis of output signals from a pair of microphones (COP method). Advantages and limitations of both methods are examined. It is shown that the microphone array can accurately measures TE noise and captures its two-dimensional characteristic over a large frequency range for any TE configuration as long as noise contamination from extraneous sources is within bounds. The COP method is shown to also accurately measure TE noise but over a more limited frequency range that narrows for increased TE thickness. Finally, the applicability and generality of an airfoil self-noise prediction method was evaluated via comparison to the experimental data obtained using the COP and array measurement methods. The predicted and experimental results are shown to agree over large frequency ranges.

Application of MEMS Microphone Array Technology to Airframe Noise Measurements

NASA Technical Reports Server (NTRS)

Humphreys, William M., Jr.; Shams, Qamar A.; Graves, Sharon S.; Sealey, Bradley S.; Bartram, Scott M.; Comeaux, Toby

2005-01-01

Current generation microphone directional array instrumentation is capable of extracting accurate noise source location and directivity data on a variety of aircraft components, resulting in significant gains in test productivity. However, with this gain in productivity has come the desire to install larger and more complex arrays in a variety of ground test facilities, creating new challenges for the designers of array systems. To overcome these challenges, a research study was initiated to identify and develop hardware and fabrication technologies which could be used to construct an array system exhibiting acceptable measurement performance but at much lower cost and with much simpler installation requirements. This paper describes an effort to fabricate a 128-sensor array using commercially available Micro-Electro-Mechanical System (MEMS) microphones. The MEMS array was used to acquire noise data for an isolated 26%-scale high-fidelity Boeing 777 landing gear in the Virginia Polytechnic Institute and State University Stability Tunnel across a range of Mach numbers. The overall performance of the array was excellent, and major noise sources were successfully identified from the measurements.

Are high perches in the blackcap Sylvia atricapilla song or listening posts? A sound transmission study.

PubMed

Mathevon, Nicolas; Dabelsteen, Torben; Blumenrath, Sandra H

2005-01-01

Birds often sing from high perches referred to as song posts. However, birds also listen and keep a lookout from these perches. We used a sound transmission experiment to investigate the changes for receiving and sending conditions that a territorial songbird may experience by moving upwards in the vegetation. Representative song elements of the blackcap Sylvia atricapilla were transmitted in a forest habitat in spring using a complete factorial design with natural transmission distances and speaker and microphone heights. Four aspects of sound degradation were quantified: signal-to-noise ratio, excess attenuation, distortion within the sounds determined as a blur ratio, and prolongation of the sounds with "tails" of echoes determined as a tail-to-signal ratio. All four measures indicated that degradation decreased with speaker and microphone height. However, the decrease was considerably higher for the microphone than for the speaker. This suggests that choosing high perches in a forest at spring results in more benefits to blackcaps in terms of improved communication conditions when they act as receivers than as senders.

Cell-phone vs microphone recordings: Judging emotion in the voice.

PubMed

Green, Joshua J; Eigsti, Inge-Marie

2017-09-01

Emotional states can be conveyed by vocal cues such as pitch and intensity. Despite the ubiquity of cellular telephones, there is limited information on how vocal emotional states are perceived during cell-phone transmissions. Emotional utterances (neutral, happy, angry) were elicited from two female talkers and simultaneously recorded via microphone and cell-phone. Ten-step continua (neutral to happy, neutral to angry) were generated using the straight algorithm. Analyses compared reaction time (RT) and emotion judgment as a function of recording type (microphone vs cell-phone). Logistic regression revealed no judgment differences between recording types, though there were interactions with emotion type. Multi-level model analyses indicated that RT data were best fit by a quadratic model, with slower RT at the middle of each continuum, suggesting greater ambiguity, and slower RT for cell-phone stimuli across blocks. While preliminary, results suggest that critical acoustic cues to emotion are largely retained in cell-phone transmissions, though with effects of recording source on RT, and support the methodological utility of collecting speech samples by phone.

Ground effects on aircraft noise. [near grazing incidence

NASA Technical Reports Server (NTRS)

Willshire, W. L., Jr.; Hilton, D. A.

1979-01-01

A flight experiment was conducted to investigate air-to-ground propagation of sound near grazing incidence. A turbojet-powered aircraft was flown at low altitudes over the ends of two microphone arrays. An eight-microphone array was positioned along a 1850 m concrete runway. The second array consisted of 12 microphones positioned parallel to the runway over grass. Twenty-eight flights were flown at altitudes ranging from 10 m to 160 m. The acoustic data recorded in the field reduced to one-third-octave band spectra and time correlated with the flight and weather information. A small portion of the data was further reduced to values of ground attenuation as a function of frequency and incidence angle by two different methods. In both methods, the acoustic signals compared originated from identical sources. Attenuation results obtained by using the two methods were in general agreement. The measured ground attenuation was largest in the frequency range of 200 to 400 Hz. A strong dependence was found between ground attenuation and incidence angle with little attenuation measured for angles of incidence greater than 10 to 15 degrees.

Polyvinylidene fluoride (PVDF) vibration sensor for stethoscope and contact microphones

NASA Astrophysics Data System (ADS)

Toda, Minoru; Thompson, Mitchell

2005-09-01

This paper describes a new type of contact vibration sensor made by bonding piezoelectric PVDF film to a curved frame structure. The concave surface of the film is bonded to a rubber piece having a front contact face. Vibration is transmitted from this face through the rubber to the surface of the PVDF film. Pressure normal to the surface of the film is converted to circumferential strain, and an electric field is induced by the piezoelectric effect. The frequency response of the device was measured using an accelerometer mounted between the rubber face and a rigid vibration exciter plate. Sensitivity (voltage per unit displacement) was deduced from the device output and measured acceleration. The sensitivity was flat from 16 Hz to 3 kHz, peaking at 6 kHz due to a structural resonance. Calculations predicting performance against human tissue (stethoscope or contact microphone) show results similar to data measured against the metal vibrator. This implies that an accelerometer can be used for calibrating a stethoscope or contact microphone. The observed arterial pulse waveform showed more low-frequency content than a conventional electronic stethoscope.

Acoustic centering of sources measured by surrounding spherical microphone arrays.

PubMed

Hagai, Ilan Ben; Pollow, Martin; Vorländer, Michael; Rafaely, Boaz

2011-10-01

The radiation patterns of acoustic sources have great significance in a wide range of applications, such as measuring the directivity of loudspeakers and investigating the radiation of musical instruments for auralization. Recently, surrounding spherical microphone arrays have been studied for sound field analysis, facilitating measurement of the pressure around a sphere and the computation of the spherical harmonics spectrum of the sound source. However, the sound radiation pattern may be affected by the location of the source inside the microphone array, which is an undesirable property when aiming to characterize source radiation in a unique manner. This paper presents a theoretical analysis of the spherical harmonics spectrum of spatially translated sources and defines four measures for the misalignment of the acoustic center of a radiating source. Optimization is used to promote optimal alignment based on the proposed measures and the errors caused by numerical and array-order limitations are investigated. This methodology is examined using both simulated and experimental data in order to investigate the performance and limitations of the different alignment methods. © 2011 Acoustical Society of America

Signal Restoration of Non-stationary Acoustic Signals in the Time Domain

NASA Technical Reports Server (NTRS)

Babkin, Alexander S.

1988-01-01

Signal restoration is a method of transforming a nonstationary signal acquired by a ground based microphone to an equivalent stationary signal. The benefit of the signal restoration is a simplification of the flight test requirements because it could dispense with the need to acquire acoustic data with another aircraft flying in concert with the rotorcraft. The data quality is also generally improved because the contamination of the signal by the propeller and wind noise is not present. The restoration methodology can also be combined with other data acquisition methods, such as a multiple linear microphone array for further improvement of the test results. The methodology and software are presented for performing the signal restoration in the time domain. The method has no restrictions on flight path geometry or flight regimes. Only requirement is that the aircraft spatial position be known relative to the microphone location and synchronized with the acoustic data. The restoration process assumes that the moving source radiates a stationary signal, which is then transformed into a nonstationary signal by various modulation processes. The restoration contains only the modulation due to the source motion.

An evaluation of the performance of two binaural beamformers in complex and dynamic multitalker environments.

PubMed

Best, Virginia; Mejia, Jorge; Freeston, Katrina; van Hoesel, Richard J; Dillon, Harvey

2015-01-01

Binaural beamformers are super-directional hearing aids created by combining microphone outputs from each side of the head. While they offer substantial improvements in SNR over conventional directional hearing aids, the benefits (and possible limitations) of these devices in realistic, complex listening situations have not yet been fully explored. In this study we evaluated the performance of two experimental binaural beamformers. Testing was carried out using a horizontal loudspeaker array. Background noise was created using recorded conversations. Performance measures included speech intelligibility, localization in noise, acceptable noise level, subjective ratings, and a novel dynamic speech intelligibility measure. Participants were 27 listeners with bilateral hearing loss, fitted with BTE prototypes that could be switched between conventional directional or binaural beamformer microphone modes. Relative to the conventional directional microphones, both binaural beamformer modes were generally superior for tasks involving fixed frontal targets, but not always for situations involving dynamic target locations. Binaural beamformers show promise for enhancing listening in complex situations when the location of the source of interest is predictable.

An evaluation of the performance of two binaural beamformers in complex and dynamic multitalker environments

PubMed Central

Best, Virginia; Mejia, Jorge; Freeston, Katrina; van Hoesel, Richard J.; Dillon, Harvey

2016-01-01

Objective Binaural beamformers are super-directional hearing aids created by combining microphone outputs from each side of the head. While they offer substantial improvements in SNR over conventional directional hearing aids, the benefits (and possible limitations) of these devices in realistic, complex listening situations have not yet been fully explored. In this study we evaluated the performance of two experimental binaural beamformers. Design Testing was carried out using a horizontal loudspeaker array. Background noise was created using recorded conversations. Performance measures included speech intelligibility, localisation in noise, acceptable noise level, subjective ratings, and a novel dynamic speech intelligibility measure. Study sample Participants were 27 listeners with bilateral hearing loss, fitted with BTE prototypes that could be switched between conventional directional or binaural beamformer microphone modes. Results Relative to the conventional directional microphones, both binaural beamformer modes were generally superior for tasks involving fixed frontal targets, but not always for situations involving dynamic target locations. Conclusions Binaural beamformers show promise for enhancing listening in complex situations when the location of the source of interest is predictable. PMID:26140298

Spatial sound field synthesis and upmixing based on the equivalent source method.

PubMed

Bai, Mingsian R; Hsu, Hoshen; Wen, Jheng-Ciang

2014-01-01

Given scarce number of recorded signals, spatial sound field synthesis with an extended sweet spot is a challenging problem in acoustic array signal processing. To address the problem, a synthesis and upmixing approach inspired by the equivalent source method (ESM) is proposed. The synthesis procedure is based on the pressure signals recorded by a microphone array and requires no source model. The array geometry can also be arbitrary. Four upmixing strategies are adopted to enhance the resolution of the reproduced sound field when there are more channels of loudspeakers than the microphones. Multi-channel inverse filtering with regularization is exploited to deal with the ill-posedness in the reconstruction process. The distance between the microphone and loudspeaker arrays is optimized to achieve the best synthesis quality. To validate the proposed system, numerical simulations and subjective listening experiments are performed. The results demonstrated that all upmixing methods improved the quality of reproduced target sound field over the original reproduction. In particular, the underdetermined ESM interpolation method yielded the best spatial sound field synthesis in terms of the reproduction error, timbral quality, and spatial quality.

Background Acoustics Levels in the 9x15 Wind Tunnel and Linear Array Testing

NASA Technical Reports Server (NTRS)

Stephens, David

2011-01-01

The background noise level in the 9x15 foot wind tunnel at NASA Glenn has been documented, and the results compare favorably with historical measurements. A study of recessed microphone mounting techniques was also conducted, and a recessed cavity with a micronic wire mesh screen reduces hydrodynamic noise by around 10 dB. A three-microphone signal processing technique can provide additional benefit, rejecting up to 15 dB of noise contamination at some frequencies. The screen and cavity system offers considerable benefit to test efficiency, although there are additional calibration requirements.

AGARD Flight Test Techniques Series. Volume 9. Aircraft Exterior Noise Measurement and Analysis Techniques. (Le Bruit a l’Exterieur des Aeronefs: Techniques de Mesure et d’Analyse)

DTIC Science & Technology

1991-04-01

aircraft Fig. 4.6 Airborne test set-up to compare several microphone/nose-cone arrangements for self -noise generation on a glider plane Fig. 4.7 Comparison...of normalized self -noise spectra of ogive-nose-cone equipped condenser-microphones of different diameters F!g. 4.8 Frequency splitting in the noise...output is obtained at the last com-poet ot the sub-system. The electrical respose of the entire system is then the arithmetic Sof the ildividual respnsem

iStethoscope: a demonstration of the use of mobile devices for auscultation.

PubMed

Bentley, Peter J

2015-01-01

iStethoscope Pro is the first piece of software (an "App") produced for iOS devices, which enabled users to exploit their smartphones, music players, or tablets as stethoscopes. The software exploits the built-in microphone (and supports externally added microphones) and performs real-time amplification and filtering to enable heart sounds to be heard with high fidelity. The software also enables the heart sounds to be recorded, analyzed using a spectrogram, and to be transmitted to others via e-mail. This chapter describes the motivation, functionality, and results from this work.

Operation UPSHOT-KNOTHOLE, Nevada Proving Grounds, March-June 1953. Project 6.12. Determination of Height of Burst and Ground Zero

DTIC Science & Technology

1955-05-01

6.2.1 Seismic Height of Burst Determination 6l 6.2.2 Long Range Yield Determination 6l 6.2.3 Lead Sulphide Cells 6l | • APPENDIX A SURVEY...located at M 96 in the vicinity of Array 3* The original metro station was moved to vicinity Camp Mercury Sewage Disposal Plant for Shots 9 and 10. The...JWiRE COMMUNICATION =r=^^l .QBAWIELS ., I foi* 50—ftfejP - -70 CAMP DESERT MICROPHONE 40- 15" (CAMP --’ MERCURY -60 MICROPHONE ARRAY NO I

Passive Wake Acoustics Measurements at Denver International Airport

NASA Technical Reports Server (NTRS)

Wang, Frank Y.; Wassaf, Hadi; Dougherty, Robert P.; Clark, Kevin; Gulsrud, Andrew; Fenichel, Neil; Bryant, Wayne H.

2004-01-01

From August to September 2003, NASA conducted an extensive measurement campaign to characterize the acoustic signal of wake vortices. A large, both spatially as well as in number of elements, phased microphone array was deployed at Denver International Airport for this effort. This paper will briefly describe the program background, the microphone array, as well as the supporting ground-truth and meteorological sensor suite. Sample results to date are then presented and discussed. It is seen that, in the frequency range processed so far, wake noise is generated predominantly from a very confined area around the cores.

Preliminary evaluation of a novel bone-conduction device for single-sided deafness.

PubMed

Popelka, Gerald R; Derebery, Jennifer; Blevins, Nikolas H; Murray, Michael; Moore, Brian C J; Sweetow, Robert W; Wu, Ben; Katsis, Mina

2010-04-01

A new intraoral bone-conduction device has advantages over existing bone-conduction devices for reducing the auditory deficits associated with single-sided deafness (SSD). Existing bone-conduction devices effectively mitigate auditory deficits from single-sided deafness but have suboptimal microphone locations, limited frequency range, and/or require invasive surgery. A new device has been designed to improve microphone placement (in the ear canal of the deaf ear), provide a wider frequency range, and eliminate surgery by delivering bone-conduction signals to the teeth via a removable oral appliance. Forces applied by the oral appliance were compared with forces typically experienced by the teeth from normal functions such as mastication or from other appliances. Tooth surface changes were measured on extracted teeth, and transducer temperature was measured under typical use conditions. Dynamic operating range, including gain, bandwidth, and maximum output limits, were determined from uncomfortable loudness levels and vibrotactile thresholds, and speech recognition scores were measured using normal-hearing subjects. Auditory performance in noise (Hearing in Noise Test) was measured in a limited sample of SSD subjects. Overall comfort, ease of insertion, and removal and visibility of the oral appliance in comparison with traditional hearing aids were measured using a rating scale. The oral appliance produces forces that are far below those experienced by the teeth from normal functions or conventional dental appliances. The bone-conduction signal level can be adjusted to prevent tactile perception yet provide sufficient gain and output at frequencies from 250 to 12,000 Hz. The device does not damage tooth surfaces nor produce heat, can be inserted and removed easily, and is as comfortable to wear as traditional hearing aids. The new microphone location has advantages for reducing the auditory deficits caused by SSD, including the potential to provide spatial cues introduced by reflections from the pinna, compared with microphone locations for existing devices. A new approach for SSD has been proposed that optimizes microphone location and delivers sound by bone conduction through a removable oral appliance. Measures in the laboratory using normal-hearing subjects indicate that the device provides useful gain and output for SSD patients, is comfortable, does not seem to have detrimental effects on oral function or oral health, and has several advantages over existing devices. Specifically, microphone placement is optimized for reducing the auditory deficit caused by SSD, frequency bandwidth is much greater, and the system does not require surgical placement. Auditory performance in a small sample of SSD subjects indicated a substantial advantage compared with not wearing the device. Future studies will involve performance measures on SSD patients wearing the device for longer periods.

Acoustics Reflections of Full-Scale Rotor Noise Measurements in NFAC 40- by 80-Foot Wind Tunnel

NASA Technical Reports Server (NTRS)

Barbely, Natasha Lydia; Kitaplioglu, Cahit; Sim, Ben W.

2012-01-01

The objective of current research is to identify the extent of acoustic time history distortions due to wind tunnel wall reflections. Acoustic measurements from the recent full-scale Boeing-SMART rotor test (Fig. 2) will be used to illustrate the quality of noise measurement in the NFAC 40- by 80-Foot Wind Tunnel test section. Results will be compared to PSU-WOPWOP predictions obtained with and without adjustments due to sound reflections off wind tunnel walls. Present research assumes a rectangular enclosure as shown in Fig. 3a. The Method of Mirror Images7 is used to account for reflection sources and their acoustic paths by introducing mirror images of the rotor (i.e. acoustic source), at each and every wall surface, to enforce a no-flow boundary condition at the position of the physical walls (Fig. 3b). While conventional approach evaluates the "combined" noise from both the source and image rotor at a single microphone position, an alternative approach is used to simplify implementation of PSU-WOPWOP for this reflection analysis. Here, an "equivalent" microphone position is defined with respect to the source rotor for each mirror image that effectively renders the reflection analysis to be a one rotor, multiple microphones problem. This alternative approach has the advantage of allowing each individual "equivalent" microphone, representing the reflection pulse from the associated wall surface, to be adjusted by the panel absorption coefficient illustrated in Fig. 1a. Note that the presence of parallel wall surfaces requires an infinite number of mirror images (Fig. 3c) to satisfy the no-flow boundary conditions. In the present analysis, up to four mirror images (per wall surface) are accounted to achieve convergence in the predicted time histories

Imaging of heart acoustic based on the sub-space methods using a microphone array.

PubMed

Moghaddasi, Hanie; Almasganj, Farshad; Zoroufian, Arezoo

2017-07-01

Heart disease is one of the leading causes of death around the world. Phonocardiogram (PCG) is an important bio-signal which represents the acoustic activity of heart, typically without any spatiotemporal information of the involved acoustic sources. The aim of this study is to analyze the PCG by employing a microphone array by which the heart internal sound sources could be localized, too. In this paper, it is intended to propose a modality by which the locations of the active sources in the heart could also be investigated, during a cardiac cycle. In this way, a microphone array with six microphones is employed as the recording set up to be put on the human chest. In the following, the Group Delay MUSIC algorithm which is a sub-space based localization method is used to estimate the location of the heart sources in different phases of the PCG. We achieved to 0.14cm mean error for the sources of first heart sound (S 1 ) simulator and 0.21cm mean error for the sources of second heart sound (S 2 ) simulator with Group Delay MUSIC algorithm. The acoustical diagrams created for human subjects show distinct patterns in various phases of the cardiac cycles such as the first and second heart sounds. Moreover, the evaluated source locations for the heart valves are matched with the ones that are obtained via the 4-dimensional (4D) echocardiography applied, to a real human case. Imaging of heart acoustic map presents a new outlook to indicate the acoustic properties of cardiovascular system and disorders of valves and thereby, in the future, could be used as a new diagnostic tool. Copyright © 2017. Published by Elsevier B.V.

High-Resolution Time-Frequency Spectrum-Based Lung Function Test from a Smartphone Microphone

PubMed Central

Thap, Tharoeun; Chung, Heewon; Jeong, Changwon; Hwang, Ki-Eun; Kim, Hak-Ryul; Yoon, Kwon-Ha; Lee, Jinseok

2016-01-01

In this paper, a smartphone-based lung function test, developed to estimate lung function parameters using a high-resolution time-frequency spectrum from a smartphone built-in microphone is presented. A method of estimation of the forced expiratory volume in 1 s divided by forced vital capacity (FEV1/FVC) based on the variable frequency complex demodulation method (VFCDM) is first proposed. We evaluated our proposed method on 26 subjects, including 13 healthy subjects and 13 chronic obstructive pulmonary disease (COPD) patients, by comparing with the parameters clinically obtained from pulmonary function tests (PFTs). For the healthy subjects, we found that an absolute error (AE) and a root mean squared error (RMSE) of the FEV1/FVC ratio were 4.49% ± 3.38% and 5.54%, respectively. For the COPD patients, we found that AE and RMSE from COPD patients were 10.30% ± 10.59% and 14.48%, respectively. For both groups, we compared the results using the continuous wavelet transform (CWT) and short-time Fourier transform (STFT), and found that VFCDM was superior to CWT and STFT. Further, to estimate other parameters, including forced vital capacity (FVC), forced expiratory volume in 1 s (FEV1), and peak expiratory flow (PEF), regression analysis was conducted to establish a linear transformation. However, the parameters FVC, FEV1, and PEF had correlation factor r values of 0.323, 0.275, and −0.257, respectively, while FEV1/FVC had an r value of 0.814. The results obtained suggest that only the FEV1/FVC ratio can be accurately estimated from a smartphone built-in microphone. The other parameters, including FVC, FEV1, and PEF, were subjective and dependent on the subject’s familiarization with the test and performance of forced exhalation toward the microphone. PMID:27548164

Active Noise Control Experiments using Sound Energy Flu

NASA Astrophysics Data System (ADS)

Krause, Uli

2015-03-01

This paper reports on the latest results concerning the active noise control approach using net flow of acoustic energy. The test set-up consists of two loudspeakers simulating the engine noise and two smaller loudspeakers which belong to the active noise system. The system is completed by two acceleration sensors and one microphone per loudspeaker. The microphones are located in the near sound field of the loudspeakers. The control algorithm including the update equation of the feed-forward controller is introduced. Numerical simulations are performed with a comparison to a state of the art method minimising the radiated sound power. The proposed approach is experimentally validated.

Maneuver Acoustic Flight Test of the Bell 430 Helicopter

NASA Technical Reports Server (NTRS)

Watts, Michael E.; Snider, Royce; Greenwood, Eric; Baden, Joel

2012-01-01

A cooperative flight test by NASA, Bell Helicopter and the U.S. Army to characterize the steady state acoustics and measure the maneuver noise of a Bell Helicopter 430 aircraft was accomplished. The test occurred during June/July, 2011 at Eglin Air Force Base, Florida. This test gathered a total of 410 data points over 10 test days and compiled an extensive data base of dynamic maneuver measurements. Three microphone configurations with up to 31 microphones in each configuration were used to acquire acoustic data. Aircraft data included DGPS, aircraft state and rotor state information. This paper provides an overview of the test.

Speech intelligibility at high helium-oxygen pressures.

PubMed

Rothman, H B; Gelfand, R; Hollien, H; Lambertsen, C J

1980-12-01

Word-list intelligibility scores of unprocessed speech (mean of 4 subjects) were recorded in helium-oxygen atmospheres at stable pressures equivalent to 1600, 1400, 1200, 1000, 860, 690, 560, 392, and 200 fsw daring Predictive Studies IV-1975 by wide-bandwidth condenser microphones (frequency responses not degraded by increased gas density). Intelligibility scores were substantially lower in helium-oxygen a 200 fsw than in air at l ATA, but there was little difference between 200 fsw and 1600 fsw. A previously documented prominent decrease in intelligibility of speech between 200 or 600 fsw because of helium and pressure was probably due to degradation of microphone frequency response by high gas density.

A two-microphone method for the determination of the mode amplitude distribution in high-frequency ducted broadband sound fields.

PubMed

Joseph, P F

2017-10-01

This paper describes a measurement technique that allows the modal amplitude distribution to be determined in ducts with mean flow and reflections. The method is based only on measurements of the acoustic pressure two-point coherence at the duct wall. The technique is primarily applicable to broadband sound fields in the high frequency limit and whose mode amplitudes are mutually incoherent. The central assumption underlying the technique is that the relative mode amplitude distribution is independent of frequency. The two-microphone method proposed in this paper is also used to determine the transmitted sound power and far field pressure directivity.

Sound Source Localization through 8 MEMS Microphones Array Using a Sand-Scorpion-Inspired Spiking Neural Network.

PubMed

Beck, Christoph; Garreau, Guillaume; Georgiou, Julius

2016-01-01

Sand-scorpions and many other arachnids perceive their environment by using their feet to sense ground waves. They are able to determine amplitudes the size of an atom and locate the acoustic stimuli with an accuracy of within 13° based on their neuronal anatomy. We present here a prototype sound source localization system, inspired from this impressive performance. The system presented utilizes custom-built hardware with eight MEMS microphones, one for each foot, to acquire the acoustic scene, and a spiking neural model to localize the sound source. The current implementation shows smaller localization error than those observed in nature.

Acoustic temperature measurement in a rocket noise field.

PubMed

Giraud, Jarom H; Gee, Kent L; Ellsworth, John E

2010-05-01

A 1 μm diameter platinum wire resistance thermometer has been used to measure temperature fluctuations generated during a static GEM-60 rocket motor test. Exact and small-signal relationships between acoustic pressure and acoustic temperature are derived in order to compare the temperature probe output with that of a 3.18 mm diameter condenser microphone. After preliminary plane wave tests yielded good agreement between the transducers within the temperature probe's ∼2 kHz bandwidth, comparison between the temperature probe and microphone data during the motor firing show that the ±∼3 K acoustic temperature fluctuations are a significant contributor to the total temperature variations.

On spatial spillover in feedforward and feedback noise control

NASA Astrophysics Data System (ADS)

Xie, Antai; Bernstein, Dennis

2017-03-01

Active feedback noise control for rejecting broadband disturbances must contend with the Bode integral constraint, which implies that suppression over some frequency range gives rise to amplification over another range at the performance microphone. This is called spectral spillover. The present paper deals with spatial spillover, which refers to the amplification of noise at locations where no microphone is located. A spatial spillover function is defined, which is valid for both feedforward and feedback control with scalar and vector control inputs. This function is numerically analyzed and measured experimentally. Obstructions are introduced in the acoustic space to investigate their effect on spatial spillover.

The relationship between speech recognition, behavioural listening effort, and subjective ratings.

PubMed

Picou, Erin M; Ricketts, Todd A

2018-06-01

The purpose of this study was to evaluate the reliability and validity of four subjective questions related to listening effort. A secondary purpose of this study was to evaluate the effects of hearing aid beamforming microphone arrays on word recognition and listening effort. Participants answered subjective questions immediately following testing in a dual-task paradigm with three microphone settings in a moderately reverberant laboratory environment in two noise configurations. Participants rated their: (1) mental work, (2) desire to improve the situation, (3) tiredness, and (4) desire to give up. Data were analysed using repeated measures and reliability analyses. Eighteen adults with symmetrical sensorineural hearing loss participated. Beamforming differentially affected word recognition and listening effort. Analysis revealed the same pattern of results for behavioural listening effort and subjective ratings of desire to improve the situation. Conversely, ratings of work revealed the same pattern of results as word recognition performance. Ratings of tiredness and desire to give up were unaffected by hearing aid microphone or noise configuration. Participant ratings of their desire to control the listening situation appear to reliable subjective indicators of listening effort that align with results from a behavioural measure of listening effort.

Sources and levels of background noise in the NASA Ames 40- by 80-foot wind tunnel

NASA Technical Reports Server (NTRS)

Soderman, Paul T.

1988-01-01

Background noise levels are measured in the NASA Ames Research Center 40- by 80-Foot Wind Tunnel following installation of a sound-absorbent lining on the test-section walls. Results show that the fan-drive noise dominated the empty test-section background noise at airspeeds below 120 knots. Above 120 knots, the test-section broadband background noise was dominated by wind-induced dipole noise (except at lower harmonics of fan blade-passage tones) most likely generated at the microphone or microphone support strut. Third-octave band and narrow-band spectra are presented for several fan operating conditions and test-section airspeeds. The background noise levels can be reduced by making improvements to the microphone wind screen or support strut. Empirical equations are presented relating variations of fan noise with fan speed or blade-pitch angle. An empirical expression for typical fan noise spectra is also presented. Fan motor electric power consumption is related to the noise generation. Preliminary measurements of sound absorption by the test-section lining indicate that the 152 mm thick lining will adequately absorb test-section model noise at frequencies above 300 Hz.

Spying on small wildlife sounds using affordable collar-mounted miniature microphones: an innovative method to record individual daylong vocalisations in chipmunks.

PubMed

Couchoux, Charline; Aubert, Maxime; Garant, Dany; Réale, Denis

2015-05-06

Technological advances can greatly benefit the scientific community by making new areas of research accessible. The study of animal vocal communication, in particular, can gain new insights and knowledge from technological improvements in recording equipment. Our comprehension of the acoustic signals emitted by animals would be greatly improved if we could continuously track the daily natural emissions of individuals in the wild, especially in the context of integrating individual variation into evolutionary ecology research questions. We show here how this can be accomplished using an operational tiny audio recorder that can easily be fitted as an on-board acoustic data-logger on small free-ranging animals. The high-quality 24 h acoustic recording logged on the spy microphone device allowed us to very efficiently collect daylong chipmunk vocalisations, giving us much more detailed data than the classical use of a directional microphone over an entire field season. The recordings also allowed us to monitor individual activity patterns and record incredibly long resting heart rates, and to identify self-scratching events and even whining from pre-emerging pups in their maternal burrow.

Spying on small wildlife sounds using affordable collar-mounted miniature microphones: an innovative method to record individual daylong vocalisations in chipmunks

PubMed Central

Couchoux, Charline; Aubert, Maxime; Garant, Dany; Réale, Denis

2015-01-01

Technological advances can greatly benefit the scientific community by making new areas of research accessible. The study of animal vocal communication, in particular, can gain new insights and knowledge from technological improvements in recording equipment. Our comprehension of the acoustic signals emitted by animals would be greatly improved if we could continuously track the daily natural emissions of individuals in the wild, especially in the context of integrating individual variation into evolutionary ecology research questions. We show here how this can be accomplished using an operational tiny audio recorder that can easily be fitted as an on-board acoustic data-logger on small free-ranging animals. The high-quality 24 h acoustic recording logged on the spy microphone device allowed us to very efficiently collect daylong chipmunk vocalisations, giving us much more detailed data than the classical use of a directional microphone over an entire field season. The recordings also allowed us to monitor individual activity patterns and record incredibly long resting heart rates, and to identify self-scratching events and even whining from pre-emerging pups in their maternal burrow. PMID:25944509

Monitoring volcanic activity using correlation patterns between infrasound and ground motion

NASA Astrophysics Data System (ADS)

Ichihara, M.; Takeo, M.; Yokoo, A.; Oikawa, J.; Ohminato, T.

2012-02-01

This paper presents a simple method to distinguish infrasonic signals from wind noise using a cross-correlation function of signals from a microphone and a collocated seismometer. The method makes use of a particular feature of the cross-correlation function of vertical ground motion generated by infrasound, and the infrasound itself. Contribution of wind noise to the correlation function is effectively suppressed by separating the microphone and the seismometer by several meters because the correlation length of wind noise is much shorter than wavelengths of infrasound. The method is applied to data from two recent eruptions of Asama and Shinmoe-dake volcanoes, Japan, and demonstrates that the method effectively detects not only the main eruptions, but also minor activity generating weak infrasound hardly visible in the wave traces. In addition, the correlation function gives more information about volcanic activity than infrasound alone, because it reflects both features of incident infrasonic and seismic waves. Therefore, a graphical presentation of temporal variation in the cross-correlation function enables one to see qualitative changes of eruptive activity at a glance. This method is particularly useful when available sensors are limited, and will extend the utility of a single microphone and seismometer in monitoring volcanic activity.

Studying Room Acoustics using a Monopole-Dipole Microphone Array

NASA Technical Reports Server (NTRS)

Begault, Durand R.; Abel, Jonathan S.; Gills, Stephen R. (Technical Monitor)

1997-01-01

The use of a soundfield microphone for examining the directional nature of a room impulse response was reported recently. By cross-correlating monopole and co-located dipole microphone signals aligned with left-right, up-down, and front-back axes, a sense of signal direction of arrival is revealed. The current study is concerned with the array's ability to detect individual reflections and directions of arrival, as a function of the cross-correlation window duration. If is window is too long, weak reflections are overlooked; if too short, spurious detections result. Guidelines are presented for setting the window width according to perceptual criteria. Formulas are presented describing the accuracy with which direction of arrival can be estimated as a function of room specifics and measurement noise. The direction of arrival of early reflections is more accurately determined than that of later reflections which are quieter and more numerous. The transition from a fairly directional sound field at the beginning of the room impulse response to a uni-directional diffuse field is examined. Finally, it is shown that measurements from additional dipole orientations can significantly improve the ability to detect reflections and estimate their directions of arrival.

Robust active noise control in the loadmaster area of a military transport aircraft.

PubMed

Kochan, Kay; Sachau, Delf; Breitbach, Harald

2011-05-01

The active noise control (ANC) method is based on the superposition of a disturbance noise field with a second anti-noise field using loudspeakers and error microphones. This method can be used to reduce the noise level inside the cabin of a propeller aircraft. However, during the design process of the ANC system, extensive measurements of transfer functions are necessary to optimize the loudspeaker and microphone positions. Sometimes, the transducer positions have to be tailored according to the optimization results to achieve a sufficient noise reduction. The purpose of this paper is to introduce a controller design method for such narrow band ANC systems. The method can be seen as an extension of common transducer placement optimization procedures. In the presented method, individual weighting parameters for the loudspeakers and microphones are used. With this procedure, the tailoring of the transducer positions is replaced by adjustment of controller parameters. Moreover, the ANC system will be robust because of the fact that the uncertainties are considered during the optimization of the controller parameters. The paper describes the necessary theoretic background for the method and demonstrates the efficiency in an acoustical mock-up of a military transport aircraft.

Measurement of Trailing Edge Noise Using Directional Array and Coherent Output Power Methods

NASA Technical Reports Server (NTRS)

Hutcheson, Florence V.; Brooks, Thomas F.

2002-01-01

The use of a directional (or phased) array of microphones for the measurement of trailing edge (TE) noise is described and tested. The capabilities of this method arc evaluated via measurements of TE noise from a NACA 63-215 airfoil model and from a cylindrical rod. This TE noise measurement approach is compared to one that is based on thc cross spectral analysis of output signals from a pair of microphones placed on opposite sides of an airframe model (COP method). Advantages and limitations of both methods arc examined. It is shown that the microphone array can accurately measures TE noise and captures its two-dimensional characteristic over a large frequency range for any TE configuration as long as noise contamination from extraneous sources is within bounds. The COP method is shown to also accurately measure TE noise but over a more limited frequency range that narrows for increased TE thickness. Finally, the applicability and generality of an airfoil self-noise prediction method was evaluated via comparison to the experimental data obtained using the COP and array measurement methods. The predicted and experimental results are shown to agree over large frequency ranges.

Noise of a model high speed counterrotation propeller at simulated takeoff/approach conditions (F7/A7)

NASA Technical Reports Server (NTRS)

Woodward, Richard P.

1987-01-01

A high speed advanced counterrotation propeller, was tested in the NASA-Lewis 9 x 15 foot Anechoic Wind Tunnel at simulated takeoff/approach conditions of 0.2 Mach number. Acoustic measurements were taken with fixed floor microphones, an axially translating microphone probe, and with a polar microphone probe which was fixed to the propeller nacelle and could take both sideline and circumferential acoustic surveys. Aerodynamic measurements were also made to establish the propeller operating conditions. The propeller was run over a range of blade setting angles from 36.4/36.5 to 41.1/39.4 deg, tip speeds from 165 to 259 m/sec, rotor spacings from 1.56 to 3.63 based on forward rotor tip chord to aerodynamic separation, and angles of attack to + or - 16 deg. First order rotor alone tones showed highest directivity levels near the propeller plane, while interaction tone showed high levels throughout sideline directivity, especially toward the propeller rotation axis. Interaction tone levels were sensitive to propeller row spacing while rotor alone tones showed little spacing effect. There is a decreased noise level associated with higher propeller blade numbers for the same overall propeller thrust.

Optimization of actuator arrays for aircraft interior noise control

NASA Technical Reports Server (NTRS)

Cabell, R. H.; Lester, H. C.; Mathur, G. P.; Tran, B. N.

1993-01-01

A numerical procedure for grouping actuators in order to reduce the number of degrees of freedom in an active noise control system is evaluated using experimental data. Piezoceramic actuators for reducing aircraft interior noise are arranged into groups using a nonlinear optimization routine and clustering algorithm. An actuator group is created when two or more actuators are driven with the same control input. This procedure is suitable for active control applications where actuators are already mounted on a structure. The feasibility of this technique is demonstrated using measured data from the aft cabin of a Douglas DC-9 fuselage. The measured data include transfer functions between 34 piezoceramic actuators and 29 interior microphones and microphone responses due to the primary noise produced by external speakers. Control inputs for the grouped actuators were calculated so that a cost function, defined as a quadratic pressure term and a penalty term, was a minimum. The measured transfer functions and microphone responses are checked by comparing calculated noise reductions with measured noise reductions for four frequencies. The grouping procedure is then used to determine actuator groups that improve overall interior noise reductions by 5.3 to 15 dB, compared to the baseline experimental configuration.

An Electromechanical Model for the Cochlear Microphonic

NASA Astrophysics Data System (ADS)

Teal, Paul D.; Lineton, Ben; Elliott, Stephen J.

2011-11-01

The first of the many electrical signals generated in the ear, nerves and brain as a response to a sound incident on the ear is the cochlear microphonic (CM). The CM is generated by the hair cells of the cochlea, primarily the outer hairs cells. The potentials of this signal are a nonlinear filtered version of the acoustic pressure at the tympanic membrane. The CM signal has been used very little in recent years for clinical audiology and audiological research. This is because of uncertainty in interpreting the CM signal as a diagnostic measure, and also because of the difficulty of obtaining the signal, which has usually required the use of a transtympanic electrode. There are however, several potential clinical and research applications for acquisition of the CM. To promote understanding of the CM, and potential clinical application, a model is presented which can account for the generation of the cochlear microphonic signal. The model incorporates micro-mechanical and macro-mechanical aspects of previously published models of the basilar membrane and reticular lamina, as well as cochlear fluid mechanics, piezoelectric activity and capacitance of the outer hair cells. It also models the electrical coupling of signals along the scalae.

The effect of frequency-dependent microphone directionality on horizontal localization performance in hearing-aid users.

PubMed

Keidser, Gitte; O'Brien, Anna; Hain, Jens-Uwe; McLelland, Margot; Yeend, Ingrid

2009-11-01

Frequency-dependent microphone directionality alters the spectral shape of sound as a function of arrival azimuth. The influence of this on horizontal-plane localization performance was investigated. Using a 360 degrees loudspeaker array and five stimuli with different spectral characteristics, localization performance was measured on 21 hearing-impaired listeners when wearing no hearing aids and aided with no directionality, partial (from 1 and 2 kHz) directionality, and full directionality. The test schemes were also evaluated in everyday life. Without hearing aids, localization accuracy was significantly poorer than normative data. Due to inaudibility of high-frequency energy, front/back reversals were prominent. Front/back reversals remained prominent when aided with omnidirectional microphones. For stimuli with low-frequency emphasis, directionality had no further effect on localization. For stimuli with sufficient mid- and high-frequency information, full directionality had a small positive effect on front/back localization but a negative effect on left/right localization. Partial directionality further improved front/back localization and had no significant effect on left/right localization. The field test revealed no significant effects. The alternative spectral cues provided by frequency-dependent directionality improve front/back localization in hearing-aid users.

Broadband implementation of coprime linear microphone arrays for direction of arrival estimation.

PubMed

Bush, Dane; Xiang, Ning

2015-07-01

Coprime arrays represent a form of sparse sensing which can achieve narrow beams using relatively few elements, exceeding the spatial Nyquist sampling limit. The purpose of this paper is to expand on and experimentally validate coprime array theory in an acoustic implementation. Two nested sparse uniform linear subarrays with coprime number of elements ( M and N) each produce grating lobes that overlap with one another completely in just one direction. When the subarray outputs are combined it is possible to retain the shared beam while mostly canceling the other superfluous grating lobes. In this way a small number of microphones ( N+M-1) creates a narrow beam at higher frequencies, comparable to a densely populated uniform linear array of MN microphones. In this work beampatterns are simulated for a range of single frequencies, as well as bands of frequencies. Narrowband experimental beampatterns are shown to correspond with simulated results even at frequencies other than the arrays design frequency. Narrowband side lobe locations are shown to correspond to the theoretical values. Side lobes in the directional pattern are mitigated by increasing bandwidth of analyzed signals. Direction of arrival estimation is also implemented for two simultaneous noise sources in a free field condition.

Observed seismic and infrasonic signals around the Hakone volcano -Discussion based on a finite-difference calculation-

NASA Astrophysics Data System (ADS)

Wakamatu, S.; Kawakata, H.; Hirano, S.

2017-12-01

Observation and analysis of infrasonic waves are important for volcanology because they could be associated with mechanisms of volcanic tremors and earthquakes (Sakai et al., 2000). Around the Hakone volcano area, Japan, infrasonic waves had been observed many times in 2015 (Yukutake et al., 2016, JpGU). In the area, seismometers have been installed more than microphones, so that analysis of seismograms may also contribute to understanding some characteristics of the infrasonic waves. In this study, we focused on the infrasonic waves on July 1, 2015, at the area and discussed their propagation. We analyzed the vertical component of seven seismograms and two infrasound records; instruments for these data have been installed within 5 km from the vent emerged in the June 2015 eruption(HSRI, 2015). We summarized distances of the observation points from the vent and appearance of the signals in the seismograms and the microphone records in Table 1. We confirmed that, when the OWD microphone(Fig1) observed the infrasonic waves, seismometers of the OWD and the KIN surface seismic stations(Fig1) recorded pulse-like signals repeatedly while the other five buried seismometers did not. At the same time, the NNT microphone(Fig1) recorded no more than unclear signals despite the shorter distance to the vent than that of the KIN station. We found that the appearance of pulse-like signals at the KIN seismic station usually 10-11 seconds delay after the appearance at the OWD seismic station. The distance between these two stations is 3.5km, so that the signals in seismograms could represent propagation of the infrasonic waves rather than the seismic waves. If so, however, the infrasound propagation could be influenced by the topography of the area because the signals are unclear in the NNT microphone record.To validate the above interpretation, we simulated the diffraction of the infrasonic waves due to the topography. We executed a 3-D finite-difference calculation by discretizing the air above the area. With the topography of 10m grid, we discussed the diffraction effect on the infrasonic waves propagation. Acknowledgments: We used the records acquired by the Japan Meteorological Agency, the Hot Spring Research Institute of Kanagawa Prefecture (HSRI), and the numerical map published by the Geospatial Information Authority of Japan.

High temperature sensor/microphone development for active noise control

NASA Technical Reports Server (NTRS)

Shrout, Thomas R.

1993-01-01

The industrial and scientific communities have shown genuine interest in electronic systems which can operate at high temperatures, among which are sensors to monitor noise, vibration, and acoustic emissions. Acoustic sensing can be accomplished by a wide variety of commercially available devices, including: simple piezoelectric sensors, accelerometers, strain gauges, proximity sensors, and fiber optics. Of the several sensing mechanisms investigated, piezoelectrics were found to be the most prevalent, because of their simplicity of design and application and, because of their high sensitivity over broad ranges of frequencies and temperature. Numerous piezoelectric materials are used in acoustic sensors today; but maximum use temperatures are imposed by their transition temperatures (T(sub c)) and by their resistivity. Lithium niobate, in single crystal form, has the highest operating temperature of any commercially available material, 650 C; but that is not high enough for future requirements. Only two piezoelectric materials show potential for use at 1000 C; AlN thin film reported to be piezoactive at 1150 C, and perovskite layer structure (PLS) materials, which possess among the highest T(sub c) (greater than 1500 C) reported for ferroelectrics. A ceramic PLS composition was chosen. The solid solution composition, 80% strontium niobate (SN) and 20% strontium tantalate (STa), with a T(sub c) approximately 1160 C, was hot forged, a process which concurrently sinters and renders the plate-like grains into a highly oriented configuration to enhance piezo properties. Poled samples of this composition showed coupling (k33) approximately 6 and piezoelectric strain constant (d33) approximately 3. Piezoactivity was seen at 1125 C, the highest temperature measurement reported for a ferroelectric ceramic. The high temperature piezoelectric responses of this, and similar PLS materials, opens the possibility of their use in electronic devices operating at temperatures up to 1000 C. Concurrent with the materials study was an effort to define issues involved in the development of a microphone capable of operation at temperatures up to 1000 C; important since microphones capable of operation above 260 C are not generally available. The distinguishing feature of a microphone is its diaphragm which receives sound from the atmosphere: whereas, most other acoustic sensors receive sound through the solid structure on which they are installed. In order to gain an understanding of the potential problems involved in designing and testing a high temperature microphone, a prototype was constructed using a commercially available lithium niobate piezoelectric element in a stainless steel structure. The prototype showed excellent frequency response at room temperature, and responded to acoustic stimulation at 670 C, above which temperature the voltage output rapidly diminished because of decreased resistivity in the element. Samples of the PLS material were also evaluated in a simulated microphone configuration, but their voltage output was found to be a few mV compared to the 10 output of the prototype.

Undersafe: Monitoring safety parameters in touristic mines and caves

NASA Astrophysics Data System (ADS)

Parcerisa, David; Sanmiquel, Lluís; Alfonso, Pura; Oliva, Josep

2014-05-01

Tourism is a key sector of the European economy, generating more than 5% of the EU GPD (Gross Domestic Product). Usually, underground touristic sites receive non-expert visitors; nevertheless these activities are poorly regulated or completely deregulated. Nowadays, safety is provided by underground expert professionals whom proceed to regular inspections and by basic safety infrastructures. Even with these measures, some potential personal and environmental dangers are always present and cannot be totally avoided. Therefore, there is a clear need of a new technological product for safety and environmental continuous monitoring of tourist underground attractions. So, the aim of the Undersafe project is to provide underground attractions with a novel and specifically tailored monitoring system, easy to use and maintain. One of the goals of the Undersafe project is to develop a rock falling detection based on a set of cost limited vibration sensors. Based on the technical needs, but with cost constraints, different types of potential sensors are considered: Underground microphone: It is placed in the surface or in the underground. It is based on the consideration that the impact of the stone generates a ground impact vibration which can be understood as a "noise" that is received by a microphone capsule. Airborne sound sensing microphone: It similarly applies to underground use of the microphones, but now the microphone is tested as for its traditional use (I.e. air sound detection). In such case, the microphone detects the environmental noise produced by the impact of the stone falling onto the ground, which will include the impact sound of the stone. Geophone: It is the de facto standard for ground vibrations. Although this technology was initially discarded due to its high cost, recently, low cost geophones have appeared in the market that allows its use inside the underground attractions. Accelerometers: These, can have enough sensibility to act as vibration sensors. Although the costs of the most sensible ones are out of the limits needed for our purposes, but some non-expensive accelerometers will be tested in real environment. All these systems have been tested and it can be concluded that results have been positive for the following technologies: piezoelectric, Electret (airborne and underground) and geophone. On the contrary, accelerometer and movement sensor provided negative results. The most sensible sensor that we have found is Electret that, in turn, is the most sensitive one to out of ground environmental noise (relevant in order to discard surface vibrations effect). All sensors can provide detections in a range of 15m. Low cost rock falling detectors, in cercles of 30 m of diameter are feasible. Also detection for longer distances, up to 80 meters, is feasible, but not advisable for low-cost application. Aknowledgements: Undersafe is an European project under the auspices of the EU 7h Framework Program. EDMA Innovation S.L. deeply contributed to the development of this study.

Effectiveness of the Directional Microphone in the Baha® Divino™

PubMed Central

Oeding, Kristi; Valente, Michael; Kerckhoff, Jessica

2010-01-01

Background Patients with unilateral sensorineural hearing loss (USNHL) experience great difficulty listening to speech in noisy environments. A directional microphone (DM) could potentially improve speech recognition in this difficult listening environment. It is well known that DMs in behind-the-ear (BTE) and custom hearing aids can provide a greater signal-to-noise ratio (SNR) in comparison to an omnidirectional microphone (OM) to improve speech recognition in noise for persons with hearing impairment. Studies examining the DM in bone anchored auditory osseointegrated implants (Baha), however, have been mixed, with little to no benefit reported for the DM compared to an OM. Purpose The primary purpose of this study was to determine if there are statistically significant differences in the mean reception threshold for sentences (RTS in dB) in noise between the OM and DM in the Baha® Divino™. The RTS of these two microphone modes was measured utilizing two loudspeaker arrays (speech from 0° and noise from 180° or a diffuse eight-loudspeaker array) and with the better ear open or closed with an earmold impression and noise attenuating earmuff. Subjective benefit was assessed using the Abbreviated Profile of Hearing Aid Benefit (APHAB) to compare unaided and aided (Divino OM and DM combined) problem scores. Research Design A repeated measures design was utilized, with each subject counterbalanced to each of the eight treatment levels for three independent variables: (1) microphone (OM and DM), (2) loudspeaker array (180° and diffuse), and (3) better ear (open and closed). Study Sample Sixteen subjects with USNHL currently utilizing the Baha were recruited from Washington University’s Center for Advanced Medicine and the surrounding area. Data Collection and Analysis Subjects were tested at the initial visit if they entered the study wearing the Divino or after at least four weeks of acclimatization to a loaner Divino. The RTS was determined utilizing Hearing in Noise Test (HINT) sentences in the R-Space™ system, and subjective benefit was determined utilizing the APHAB. A three-way repeated measures analysis of variance (ANOVA) and a paired samples t-test were utilized to analyze results of the HINT and APHAB, respectively. Results Results revealed statistically significant differences within microphone (p < 0.001; directional advantage of 3.2 dB), loudspeaker array (p = 0.046; 180° advantage of 1.1 dB), and better ear conditions (p < 0.001; open ear advantage of 4.9 dB). Results from the APHAB revealed statistically and clinically significant benefit for the Divino relative to unaided on the subscales of Ease of Communication (EC) (p = 0.037), Background Noise (BN) (p < 0.001), and Reverberation (RV) (p = 0.005). Conclusions The Divino’s DM provides a statistically significant improvement in speech recognition in noise compared to the OM for subjects with USNHL. Therefore, it is recommended that audiologists consider selecting a Baha with a DM to provide improved speech recognition performance in noisy listening environments. PMID:21034701

Effectiveness of the directional microphone in the Baha® Divino™.

PubMed

Oeding, Kristi; Valente, Michael; Kerckhoff, Jessica

2010-09-01

Patients with unilateral sensorineural hearing loss (USNHL) experience great difficulty listening to speech in noisy environments. A directional microphone (DM) could potentially improve speech recognition in this difficult listening environment. It is well known that DMs in behind-the-ear (BTE) and custom hearing aids can provide a greater signal-to-noise ratio (SNR) in comparison to an omnidirectional microphone (OM) to improve speech recognition in noise for persons with hearing impairment. Studies examining the DM in bone anchored auditory osseointegrated implants (Baha), however, have been mixed, with little to no benefit reported for the DM compared to an OM. The primary purpose of this study was to determine if there are statistically significant differences in the mean reception threshold for sentences (RTS in dB) in noise between the OM and DM in the Baha® Divino™. The RTS of these two microphone modes was measured utilizing two loudspeaker arrays (speech from 0° and noise from 180° or a diffuse eight-loudspeaker array) and with the better ear open or closed with an earmold impression and noise attenuating earmuff. Subjective benefit was assessed using the Abbreviated Profile of Hearing Aid Benefit (APHAB) to compare unaided and aided (Divino OM and DM combined) problem scores. A repeated measures design was utilized, with each subject counterbalanced to each of the eight treatment levels for three independent variables: (1) microphone (OM and DM), (2) loudspeaker array (180° and diffuse), and (3) better ear (open and closed). Sixteen subjects with USNHL currently utilizing the Baha were recruited from Washington University's Center for Advanced Medicine and the surrounding area. Subjects were tested at the initial visit if they entered the study wearing the Divino or after at least four weeks of acclimatization to a loaner Divino. The RTS was determined utilizing Hearing in Noise Test (HINT) sentences in the R-Space™ system, and subjective benefit was determined utilizing the APHAB. A three-way repeated measures analysis of variance (ANOVA) and a paired samples t-test were utilized to analyze results of the HINT and APHAB, respectively. Results revealed statistically significant differences within microphone (p < 0.001; directional advantage of 3.2 dB), loudspeaker array (p = 0.046; 180° advantage of 1.1 dB), and better ear conditions (p < 0.001; open ear advantage of 4.9 dB). Results from the APHAB revealed statistically and clinically significant benefit for the Divino relative to unaided on the subscales of Ease of Communication (EC) (p = 0.037), Background Noise (BN) (p < 0.001), and Reverberation (RV) (p = 0.005). The Divino's DM provides a statistically significant improvement in speech recognition in noise compared to the OM for subjects with USNHL. Therefore, it is recommended that audiologists consider selecting a Baha with a DM to provide improved speech recognition performance in noisy listening environments. American Academy of Audiology.

Laser microphone

DOEpatents

Veligdan, James T.

2000-11-14

A microphone for detecting sound pressure waves includes a laser resonator having a laser gain material aligned coaxially between a pair of first and second mirrors for producing a laser beam. A reference cell is disposed between the laser material and one of the mirrors for transmitting a reference portion of the laser beam between the mirrors. A sensing cell is disposed between the laser material and one of the mirrors, and is laterally displaced from the reference cell for transmitting a signal portion of the laser beam, with the sensing cell being open for receiving the sound waves. A photodetector is disposed in optical communication with the first mirror for receiving the laser beam, and produces an acoustic signal therefrom for the sound waves.

Evaluation of Speech Perception via the Use of Hearing Loops and Telecoils

PubMed Central

Holmes, Alice E.; Kricos, Patricia B.; Gaeta, Laura; Martin, Sheridan

2015-01-01

A cross-sectional, experimental, and randomized repeated-measures design study was used to examine the objective and subjective value of telecoil and hearing loop systems. Word recognition and speech perception were tested in 12 older adult hearing aid users using the telecoil and microphone inputs in quiet and noise conditions. Participants were asked to subjectively rate cognitive listening effort and self-confidence for each condition. Significant improvement in speech perception with the telecoil over microphone input in both quiet and noise was found along with significantly less reported cognitive listening effort and high self-confidence. The use of telecoils with hearing aids should be recommended for older adults with hearing loss. PMID:28138458

A Blanik L-23 glider carrying a microphone and a pressure transducer flies near a BADS sensor following flight under the path of the F-5E SSBE aircraft

NASA Image and Video Library

2004-01-13

A United States Air Force Test Pilot School Blanik L-23 glider carrying a microphone and a pressure transducer flies near a BADS (Boom Amplitudes Direction System) sensor following flight at an altitude of 10 thousand feet under the path of the F-5E SSBE aircraft. The SSBE (Shaped Sonic Boom Experiment) was formerly known as the Shaped Sonic Boom Demonstration, or SSBD, and is part of DARPA's Quiet Supersonic Platform (QSP) program. On August 27, 2003, the F-5E SSBD aircraft demonstrated a method to reduce the intensity of sonic booms.

Assessment of Microphone Phased Array for Measuring Launch Vehicle Lift-off Acoustics

NASA Technical Reports Server (NTRS)

Garcia, Roberto

2012-01-01

The specific purpose of the present work was to demonstrate the suitability of a microphone phased array for launch acoustics applications via participation in selected firings of the Ares I Scale Model Acoustics Test. The Ares I Scale Model Acoustics Test is a part of the discontinued Constellation Program Ares I Project, but the basic understanding gained from this test is expected to help development of the Space Launch System vehicles. Correct identification of sources not only improves the predictive ability, but provides guidance for a quieter design of the launch pad and optimization of the water suppression system. This document contains the results of the NASA Engineering and Safety Center assessment.

Maneuver Acoustic Flight Test of the Bell 430 Helicopter Data Report

NASA Technical Reports Server (NTRS)

Watts, Michael E.; Greenwood, Eric; Smith, Charles D.; Snider, Royce; Conner, David A.

2014-01-01

A cooperative ight test by NASA, Bell Helicopter and the U.S. Army to characterize the steady state acoustics and measure the maneuver noise of a Bell Helicopter 430 aircraft was accomplished. The test occurred during June/July 2011 at Eglin Air Force Base, Florida. This test gathered a total of 410 test points over 10 test days and compiled an extensive database of dynamic maneuver measurements. Three microphone arrays with up to 31 microphon. es in each were used to acquire acoustic data. Aircraft data included Differential Global Positioning System, aircraft state and rotor state information. This paper provides an overview of the test and documents the data acquired.

Assessment of the pseudo-tracking approach for the calculation of material acceleration and pressure fields from time-resolved PIV: part II. Spatio-temporal filtering

NASA Astrophysics Data System (ADS)

van Gent, P. L.; Schrijer, F. F. J.; van Oudheusden, B. W.

2018-04-01

The present study characterises the spatio-temporal filtering associated with pseudo-tracking. A combined theoretical and numerical assessment is performed that uses the relatively simple flow case of a two-dimensional Taylor vortex as analytical test case. An additional experimental assessment considers the more complex flow of a low-speed axisymmetric base flow, for which time-resolved tomographic PIV measurements and microphone measurements were obtained. The results of these assessments show how filtering along Lagrangian tracks leads to amplitude modulation of flow structures. A cut-off track length and spatial resolution are specified to support future applications of the pseudo-tracking approach. The experimental results show a fair agreement between PIV and microphone pressure data in terms of fluctuation levels and pressure frequency spectra. The coherence and correlation between microphone and PIV pressure measurements were found to be substantial and almost independent of the track length, indicating that the low-frequency behaviour of the flow could be reproduced regardless of the track length. It is suggested that a spectral analysis can be used inform the selection of a suitable track length and to estimate the local error margin of reconstructed pressure values.

Measurements of Infrared and Acoustic Source Distributions in Jet Plumes

NASA Technical Reports Server (NTRS)

Agboola, Femi A.; Bridges, James; Saiyed, Naseem

2004-01-01

The aim of this investigation was to use the linear phased array (LPA) microphones and infrared (IR) imaging to study the effects of advanced nozzle-mixing techniques on jet noise reduction. Several full-scale engine nozzles were tested at varying power cycles with the linear phased array setup parallel to the jet axis. The array consisted of 16 sparsely distributed microphones. The phased array microphone measurements were taken at a distance of 51.0 ft (15.5 m) from the jet axis, and the results were used to obtain relative overall sound pressure levels from one nozzle design to the other. The IR imaging system was used to acquire real-time dynamic thermal patterns of the exhaust jet from the nozzles tested. The IR camera measured the IR radiation from the nozzle exit to a distance of six fan diameters (X/D(sub FAN) = 6), along the jet plume axis. The images confirmed the expected jet plume mixing intensity, and the phased array results showed the differences in sound pressure level with respect to nozzle configurations. The results show the effects of changes in configurations to the exit nozzles on both the flows mixing patterns and radiant energy dissipation patterns. By comparing the results from these two measurements, a relationship between noise reduction and core/bypass flow mixing is demonstrated.

Improvement of plastic optical fiber microphone based on moisture pattern sensing in devoiced breath

NASA Astrophysics Data System (ADS)

Taki, Tomohito; Honma, Satoshi; Morisawa, Masayuki; Muto, Shinzo

2008-03-01

Conversation is the most practical and common form in communication. However, people with a verbal handicap feel a difficulty to produce words due to variations in vocal chords. This research leads to develop a new devoiced microphone system based on distinguishes between the moisture patterns for each devoiced breaths, using a plastic optical fiber (POF) moisture sensor. In the experiment, five POF-type moisture sensors with fast response were fabricated by coating swell polymer with a slightly larger refractive index than that of fiber core and were set in front of mouth. When these sensors are exposed into humid air produced by devoiced breath, refractive index in cladding layer decreases by swelling and then the POF sensor heads change to guided type. Based on the above operation principle, the output light intensities from the five sensors set in front of mouth change each other. Using above mentioned output light intensity patterns, discernment of devoiced vowels in Japanese (a,i,u,e,o) was tried by means of DynamicProgramming-Matching (DP-matching) method. As the result, distinction rate over 90% was obtained to Japanese devoiced vowels. Therefore, using this system and a voice synthesizer, development of new microphone for the person with a functional disorder in the vocal chords seems to be possible.

Estimation of Temporal Gait Parameters Using a Wearable Microphone-Sensor-Based System

PubMed Central

Wang, Cheng; Wang, Xiangdong; Long, Zhou; Yuan, Jing; Qian, Yueliang; Li, Jintao

2016-01-01

Most existing wearable gait analysis methods focus on the analysis of data obtained from inertial sensors. This paper proposes a novel, low-cost, wireless and wearable gait analysis system which uses microphone sensors to collect footstep sound signals during walking. This is the first time a microphone sensor is used as a wearable gait analysis device as far as we know. Based on this system, a gait analysis algorithm for estimating the temporal parameters of gait is presented. The algorithm fully uses the fusion of two feet footstep sound signals and includes three stages: footstep detection, heel-strike event and toe-on event detection, and calculation of gait temporal parameters. Experimental results show that with a total of 240 data sequences and 1732 steps collected using three different gait data collection strategies from 15 healthy subjects, the proposed system achieves an average 0.955 F1-measure for footstep detection, an average 94.52% accuracy rate for heel-strike detection and 94.25% accuracy rate for toe-on detection. Using these detection results, nine temporal related gait parameters are calculated and these parameters are consistent with their corresponding normal gait temporal parameters and labeled data calculation results. The results verify the effectiveness of our proposed system and algorithm for temporal gait parameter estimation. PMID:27999321

Mission-Oriented Sensor Arrays and UAVs - a Case Study on Environmental Monitoring

NASA Astrophysics Data System (ADS)

Figueira, N. M.; Freire, I. L.; Trindade, O.; Simões, E.

2015-08-01

This paper presents a new concept of UAV mission design in geomatics, applied to the generation of thematic maps for a multitude of civilian and military applications. We discuss the architecture of Mission-Oriented Sensors Arrays (MOSA), proposed in Figueira et Al. (2013), aimed at splitting and decoupling the mission-oriented part of the system (non safety-critical hardware and software) from the aircraft control systems (safety-critical). As a case study, we present an environmental monitoring application for the automatic generation of thematic maps to track gunshot activity in conservation areas. The MOSA modeled for this application integrates information from a thermal camera and an on-the-ground microphone array. The use of microphone arrays technology is of particular interest in this paper. These arrays allow estimation of the direction-of-arrival (DOA) of the incoming sound waves. Information about events of interest is obtained by the fusion of the data provided by the microphone array, captured by the UAV, fused with information from the termal image processing. Preliminary results show the feasibility of the on-the-ground sound processing array and the simulation of the main processing module, to be embedded into an UAV in a future work. The main contributions of this paper are the proposed MOSA system, including concepts, models and architecture.

Fly-ear inspired acoustic sensors for gunshot localization

NASA Astrophysics Data System (ADS)

Liu, Haijun; Currano, Luke; Gee, Danny; Yang, Benjamin; Yu, Miao

2009-05-01

The supersensitive ears of the parasitoid fly Ormia ochracea have inspired researchers to develop bio-inspired directional microphone for sound localization. Although the fly ear is optimized for localizing the narrow-band calling song of crickets at 5 kHz, experiments and simulation have shown that it can amplify directional cues for a wide frequency range. In this article, a theoretical investigation is presented to study the use of fly-ear inspired directional microphones for gunshot localization. Using an equivalent 2-DOF model of the fly ear, the time responses of the fly ear structure to a typical shock wave are obtained and the associated time delay is estimated by using cross-correlation. Both near-field and far-field scenarios are considered. The simulation shows that the fly ear can greatly amplify the time delay by ~20 times, which indicates that with an interaural distance of only 1.2 mm the fly ear is able to generate a time delay comparable to that obtained by a conventional microphone pair with a separation as large as 24 mm. Since the parameters of the fly ear structure can also be tuned for muzzle blast and other impulse stimulus, fly-ear inspired acoustic sensors offers great potential for developing portable gunshot localization systems.

A self-teaching image processing and voice-recognition-based, intelligent and interactive system to educate visually impaired children

NASA Astrophysics Data System (ADS)

Iqbal, Asim; Farooq, Umar; Mahmood, Hassan; Asad, Muhammad Usman; Khan, Akrama; Atiq, Hafiz Muhammad

2010-02-01

A self teaching image processing and voice recognition based system is developed to educate visually impaired children, chiefly in their primary education. System comprises of a computer, a vision camera, an ear speaker and a microphone. Camera, attached with the computer system is mounted on the ceiling opposite (on the required angle) to the desk on which the book is placed. Sample images and voices in the form of instructions and commands of English, Urdu alphabets, Numeric Digits, Operators and Shapes are already stored in the database. A blind child first reads the embossed character (object) with the help of fingers than he speaks the answer, name of the character, shape etc into the microphone. With the voice command of a blind child received by the microphone, image is taken by the camera which is processed by MATLAB® program developed with the help of Image Acquisition and Image processing toolbox and generates a response or required set of instructions to child via ear speaker, resulting in self education of a visually impaired child. Speech recognition program is also developed in MATLAB® with the help of Data Acquisition and Signal Processing toolbox which records and process the command of the blind child.

Comparison of Psychophysical and Physical Measurements of Real Ear to Coupler Differences.

PubMed

Koning, Raphael; Wouters, Jan; Francart, Tom

2015-01-01

The purpose of the study is to compare real ear to coupler difference (RECD) curves based on physical and psychophysical measures. For the physically measured RECD, the RECD was measured with real ear and coupler measurements for the ear simulator and HA1- and HA2 2-cc couplers. The psychophysically measured RECDs were derived from audiogram measures. RECDs were measured in 19 normally hearing subjects. The coupler measurement was done with the probe microphone and the coupler microphone itself. Psychophysically measured RECDs were derived for all subjects by measuring the audiogram in sound field and with an ER-3A insert phone. Reference data were obtained for the three coupler types. It was possible to derive the RECD curve with psychophysical methods. There was no overall statistical difference between the physically and psychophysically measured RECD curves for the HA2 2-cc coupler and the ear simulator. The standard deviation was, however, much higher for the psychophysically derived RECD, indicating that physically measured RECDs are more precise than psychophysically derived RECDs. For the physical RECD measurements, the coupler microphone should be used for the coupler measurement. Physically measured RECDs were validated on group level by the reliable derivation of the RECD curve from audiogram measures.

Tutorial and Guidelines on Measurement of Sound Pressure Level in Voice and Speech.

PubMed

Švec, Jan G; Granqvist, Svante

2018-03-15

Sound pressure level (SPL) measurement of voice and speech is often considered a trivial matter, but the measured levels are often reported incorrectly or incompletely, making them difficult to compare among various studies. This article aims at explaining the fundamental principles behind these measurements and providing guidelines to improve their accuracy and reproducibility. Basic information is put together from standards, technical, voice and speech literature, and practical experience of the authors and is explained for nontechnical readers. Variation of SPL with distance, sound level meters and their accuracy, frequency and time weightings, and background noise topics are reviewed. Several calibration procedures for SPL measurements are described for stand-mounted and head-mounted microphones. SPL of voice and speech should be reported together with the mouth-to-microphone distance so that the levels can be related to vocal power. Sound level measurement settings (i.e., frequency weighting and time weighting/averaging) should always be specified. Classified sound level meters should be used to assure measurement accuracy. Head-mounted microphones placed at the proximity of the mouth improve signal-to-noise ratio and can be taken advantage of for voice SPL measurements when calibrated. Background noise levels should be reported besides the sound levels of voice and speech.

Calibration of Clinical Audio Recording and Analysis Systems for Sound Intensity Measurement.

PubMed

Maryn, Youri; Zarowski, Andrzej

2015-11-01

Sound intensity is an important acoustic feature of voice/speech signals. Yet recordings are performed with different microphone, amplifier, and computer configurations, and it is therefore crucial to calibrate sound intensity measures of clinical audio recording and analysis systems on the basis of output of a sound-level meter. This study was designed to evaluate feasibility, validity, and accuracy of calibration methods, including audiometric speech noise signals and human voice signals under typical speech conditions. Calibration consisted of 3 comparisons between data from 29 measurement microphone-and-computer systems and data from the sound-level meter: signal-specific comparison with audiometric speech noise at 5 levels, signal-specific comparison with natural voice at 3 levels, and cross-signal comparison with natural voice at 3 levels. Intensity measures from recording systems were then linearly converted into calibrated data on the basis of these comparisons, and validity and accuracy of calibrated sound intensity were investigated. Very strong correlations and quasisimilarity were found between calibrated data and sound-level meter data across calibration methods and recording systems. Calibration of clinical sound intensity measures according to this method is feasible, valid, accurate, and representative for a heterogeneous set of microphones and data acquisition systems in real-life circumstances with distinct noise contexts.

Parametric Investigation of Laser Doppler Microphones

NASA Astrophysics Data System (ADS)

Daoud, M.; Naguib, A.

2002-11-01

The concept of a Laser Doppler Microphone (LDM) is based on utilizing the Doppler frequency shift of a focused laser beam to measure the unsteady velocity of the center point of a flexible polymer diaphragm that is mounted on top of a hole and subjected to the unsteady pressure. Time integration of the velocity signal yields a time series of the diaphragm displacement, which can be converted to pressure from knowledge of the sensor's deflection sensitivity. In our APS/DFD presentation last year, the stringent frequency resolution requirement of these new sensors and methods to meet this requirement were discussed. Here, the dependence of the sensor characteristics (sensitivity, bandwidth, and noise floor) on various significant parameters is investigated in detail by calibrating the sensor in a plane wave tube in the frequency range of 50 - 5000 Hz. Parameters investigated include sensor diaphragm material and thickness, sensor size, damping of the diaphragm motion and laser beam spot size. The results shed light on the operating limits of the new sensor and demonstrate its ability to conduct high-spatial-resolution measurements in typical high-Reynolds-number test facilities. Moreover, calibrated LDM sensors were used to conduct measurements in a separating/reattaching flow and the results are compared to classical electret-type microphones with a similar sensing diameter.

Development and Technical Validation of the Mobile Based Assistive Listening System: A Smartphone-Based Remote Microphone.

PubMed

Lopez, Esteban Alejandro; Costa, Orozimbo Alves; Ferrari, Deborah Viviane

2016-10-01

The purpose of this research note is to describe the development and technical validation of the Mobile Based Assistive Listening System (MoBALS), a free-of-charge smartphone-based remote microphone application. MoBALS Version 1.0 was developed for Android (Version 2.1 or higher) and was coded with Java using Eclipse Indigo with the Android Software Development Kit. A Wi-Fi router with background traffic and 2 affordable smartphones were used for debugging and technical validation comprising, among other things, multicasting capability, data packet loss, and battery consumption. MoBALS requires at least 2 smartphones connected to the same Wi-Fi router for signal transmission and reception. Subscriber identity module cards or Internet connections are not needed. MoBALS can be used alone or connected to a hearing aid or cochlear implant via direct audio input. Maximum data packet loss was 99.28%, and minimum battery life was 5 hr. Other relevant design specifications and their implementation are described. MoBALS performed as a remote microphone with enhanced accessibility features and avoids overhead expenses by using already-available and affordable technology. The further development and technical revalidation of MoBALS will be followed by clinical evaluation with persons with hearing impairment.

Theory and investigation of acoustic multiple-input multiple-output systems based on spherical arrays in a room.

PubMed

Morgenstern, Hai; Rafaely, Boaz; Zotter, Franz

2015-11-01

Spatial attributes of room acoustics have been widely studied using microphone and loudspeaker arrays. However, systems that combine both arrays, referred to as multiple-input multiple-output (MIMO) systems, have only been studied to a limited degree in this context. These systems can potentially provide a powerful tool for room acoustics analysis due to the ability to simultaneously control both arrays. This paper offers a theoretical framework for the spatial analysis of enclosed sound fields using a MIMO system comprising spherical loudspeaker and microphone arrays. A system transfer function is formulated in matrix form for free-field conditions, and its properties are studied using tools from linear algebra. The system is shown to have unit-rank, regardless of the array types, and its singular vectors are related to the directions of arrival and radiation at the microphone and loudspeaker arrays, respectively. The formulation is then generalized to apply to rooms, using an image source method. In this case, the rank of the system is related to the number of significant reflections. The paper ends with simulation studies, which support the developed theory, and with an extensive reflection analysis of a room impulse response, using the platform of a MIMO system.

Generation of Higher Order Modes in a Rectangular Duct

NASA Technical Reports Server (NTRS)

Gerhold, Carl H.; Cabell, Randolph H.; Brown, Donald E.

2004-01-01

Advanced noise control methodologies to reduce sound emission from aircraft engines take advantage of the modal structure of the noise in the duct. This noise is caused by the interaction of rotor wakes with downstream obstructions such as exit guide vanes. Mode synthesis has been accomplished in circular ducts and current active noise control work has made use of this capability to cancel fan noise. The goal of the current effort is to examine the fundamental process of higher order mode propagation through an acoustically treated, curved duct. The duct cross-section is rectangular to permit greater flexibility in representation of a range of duct curvatures. The work presented is the development of a feedforward control system to generate a user-specified modal pattern in the duct. The multiple-error, filtered-x LMS algorithm is used to determine the magnitude and phase of signal input to the loudspeakers to produce a desired modal pattern at a set of error microphones. Implementation issues, including loudspeaker placement and error microphone placement, are discussed. Preliminary results from a 9-3/8 inch by 21 inch duct, using 12 loudspeakers and 24 microphones, are presented. These results demonstrate the ability of the control system to generate a user-specified mode while suppressing undesired modes.

Microphone array measurement system for analysis of directional and spatial variations of sound fields.

PubMed

Gover, Bradford N; Ryan, James G; Stinson, Michael R

2002-11-01

A measurement system has been developed that is capable of analyzing the directional and spatial variations in a reverberant sound field. A spherical, 32-element array of microphones is used to generate a narrow beam that is steered in 60 directions. Using an omnidirectional loudspeaker as excitation, the sound pressure arriving from each steering direction is measured as a function of time, in the form of pressure impulse responses. By subsequent analysis of these responses, the variation of arriving energy with direction is studied. The directional diffusion and directivity index of the arriving sound can be computed, as can the energy decay rate in each direction. An analysis of the 32 microphone responses themselves allows computation of the point-to-point variation of reverberation time and of sound pressure level, as well as the spatial cross-correlation coefficient, over the extent of the array. The system has been validated in simple sound fields in an anechoic chamber and in a reverberation chamber. The system characterizes these sound fields as expected, both quantitatively from the measures and qualitatively from plots of the arriving energy versus direction. It is anticipated that the system will be of value in evaluating the directional distribution of arriving energy and the degree and diffuseness of sound fields in rooms.

Implementation issues of the nearfield equivalent source imaging microphone array

NASA Astrophysics Data System (ADS)

Bai, Mingsian R.; Lin, Jia-Hong; Tseng, Chih-Wen

2011-01-01

This paper revisits a nearfield microphone array technique termed nearfield equivalent source imaging (NESI) proposed previously. In particular, various issues concerning the implementation of the NESI algorithm are examined. The NESI can be implemented in both the time domain and the frequency domain. Acoustical variables including sound pressure, particle velocity, active intensity and sound power are calculated by using multichannel inverse filters. Issues concerning sensor deployment are also investigated for the nearfield array. The uniform array outperformed a random array previously optimized for far-field imaging, which contradicts the conventional wisdom in far-field arrays. For applications in which only a patch array with scarce sensors is available, a virtual microphone approach is employed to ameliorate edge effects using extrapolation and to improve imaging resolution using interpolation. To enhance the processing efficiency of the time-domain NESI, an eigensystem realization algorithm (ERA) is developed. Several filtering methods are compared in terms of computational complexity. Significant saving on computations can be achieved using ERA and the frequency-domain NESI, as compared to the traditional method. The NESI technique was also experimentally validated using practical sources including a 125 cc scooter and a wooden box model with a loudspeaker fitted inside. The NESI technique proved effective in identifying broadband and non-stationary sources produced by the sources.

Modal smoothing for analysis of room reflections measured with spherical microphone and loudspeaker arrays.

PubMed

Morgenstern, Hai; Rafaely, Boaz

2018-02-01

Spatial analysis of room acoustics is an ongoing research topic. Microphone arrays have been employed for spatial analyses with an important objective being the estimation of the direction-of-arrival (DOA) of direct sound and early room reflections using room impulse responses (RIRs). An optimal method for DOA estimation is the multiple signal classification algorithm. When RIRs are considered, this method typically fails due to the correlation of room reflections, which leads to rank deficiency of the cross-spectrum matrix. Preprocessing methods for rank restoration, which may involve averaging over frequency, for example, have been proposed exclusively for spherical arrays. However, these methods fail in the case of reflections with equal time delays, which may arise in practice and could be of interest. In this paper, a method is proposed for systems that combine a spherical microphone array and a spherical loudspeaker array, referred to as multiple-input multiple-output systems. This method, referred to as modal smoothing, exploits the additional spatial diversity for rank restoration and succeeds where previous methods fail, as demonstrated in a simulation study. Finally, combining modal smoothing with a preprocessing method is proposed in order to increase the number of DOAs that can be estimated using low-order spherical loudspeaker arrays.

Impact of a Moving Noise Masker on Speech Perception in Cochlear Implant Users

PubMed Central

Weissgerber, Tobias; Rader, Tobias; Baumann, Uwe

2015-01-01

Objectives Previous studies investigating speech perception in noise have typically been conducted with static masker positions. The aim of this study was to investigate the effect of spatial separation of source and masker (spatial release from masking, SRM) in a moving masker setup and to evaluate the impact of adaptive beamforming in comparison with fixed directional microphones in cochlear implant (CI) users. Design Speech reception thresholds (SRT) were measured in S0N0 and in a moving masker setup (S0Nmove) in 12 normal hearing participants and 14 CI users (7 subjects bilateral, 7 bimodal with a hearing aid in the contralateral ear). Speech processor settings were a moderately directional microphone, a fixed beamformer, or an adaptive beamformer. The moving noise source was generated by means of wave field synthesis and was smoothly moved in a shape of a half-circle from one ear to the contralateral ear. Noise was presented in either of two conditions: continuous or modulated. Results SRTs in the S0Nmove setup were significantly improved compared to the S0N0 setup for both the normal hearing control group and the bilateral group in continuous noise, and for the control group in modulated noise. There was no effect of subject group. A significant effect of directional sensitivity was found in the S0Nmove setup. In the bilateral group, the adaptive beamformer achieved lower SRTs than the fixed beamformer setting. Adaptive beamforming improved SRT in both CI user groups substantially by about 3 dB (bimodal group) and 8 dB (bilateral group) depending on masker type. Conclusions CI users showed SRM that was comparable to normal hearing subjects. In listening situations of everyday life with spatial separation of source and masker, directional microphones significantly improved speech perception with individual improvements of up to 15 dB SNR. Users of bilateral speech processors with both directional microphones obtained the highest benefit. PMID:25970594

Effects of venting on wind noise levels measured at the eardrum.

PubMed

Chung, King

2013-01-01

Wind noise can be a nuisance to hearing aid users. With the advent of sophisticated feedback reduction algorithms, people with higher degrees of hearing loss are fit with larger vents than previously allowed, and more people with lesser degrees of hearing loss are fit with open hearing aids. The purpose of this study was to examine the effects of venting on wind noise levels in the ear canal for hearing aids with omnidirectional and directional microphones. Two behind-the-ear hearing aids were programmed when they were worn on a Knowles Electronics Manikin for Acoustic Research. The hearing aid worn on the right ear was programmed to the omnidirectional microphone mode and the one on the left to the directional microphone mode. The hearing aids were adjusted to linear amplification with flat frequency response in an anechoic chamber. Gains below 10 dB were used to avoid output limiting of wind noise levels at low input levels. Wind noise samples were recorded at the eardrum location in a wind tunnel at wind velocities ranging from a gentle to a strong breeze. The hearing aids were coupled to #13 tubings (i.e., open vent), or conventional skeleton earmolds with no vent, pressure vents, or 3mm vents. Polar and spectral characteristics of wind noise were analyzed off-line using MatLab programs. Wind noise levels in the ear canals were mostly predicted by vent-induced frequency response changes in the conventional earmold conditions for both omnidirectional and directional hearing aids. The open vent condition, however, yielded the lowest levels, which could not be entirely predicted by the frequency response changes of the hearing aids. This indicated that a wind-related vent effect permitted an additional amount of sound reduction in the ear canal, which could not be explained by known vent effects. For the microphone location, form factor, and gain settings tested, open fit hearing aids yielded lower noise levels at the eardrum location than conventional behind-the-ear hearing aids.

Acoustic Location of Lightning Using Interferometric Techniques

NASA Astrophysics Data System (ADS)

Erives, H.; Arechiga, R. O.; Stock, M.; Lapierre, J. L.; Edens, H. E.; Stringer, A.; Rison, W.; Thomas, R. J.

2013-12-01

Acoustic arrays have been used to accurately locate thunder sources in lightning flashes. The acoustic arrays located around the Magdalena mountains of central New Mexico produce locations which compare quite well with source locations provided by the New Mexico Tech Lightning Mapping Array. These arrays utilize 3 outer microphones surrounding a 4th microphone located at the center, The location is computed by band-passing the signal to remove noise, and then computing the cross correlating the outer 3 microphones with respect the center reference microphone. While this method works very well, it works best on signals with high signal to noise ratios; weaker signals are not as well located. Therefore, methods are being explored to improve the location accuracy and detection efficiency of the acoustic location systems. The signal received by acoustic arrays is strikingly similar to th signal received by radio frequency interferometers. Both acoustic location systems and radio frequency interferometers make coherent measurements of a signal arriving at a number of closely spaced antennas. And both acoustic and interferometric systems then correlate these signals between pairs of receivers to determine the direction to the source of the received signal. The primary difference between the two systems is the velocity of propagation of the emission, which is much slower for sound. Therefore, the same frequency based techniques that have been used quite successfully with radio interferometers should be applicable to acoustic based measurements as well. The results presented here are comparisons between the location results obtained with current cross correlation method and techniques developed for radio frequency interferometers applied to acoustic signals. The data were obtained during the summer 2013 storm season using multiple arrays sensitive to both infrasonic frequency and audio frequency acoustic emissions from lightning. Preliminary results show that interferometric techniques have good potential for improving the lightning location accuracy and detection efficiency of acoustic arrays.

Relating hearing loss and executive functions to hearing aid users' preference for, and speech recognition with, different combinations of binaural noise reduction and microphone directionality

PubMed Central

Neher, Tobias

2014-01-01

Knowledge of how executive functions relate to preferred hearing aid (HA) processing is sparse and seemingly inconsistent with related knowledge for speech recognition outcomes. This study thus aimed to find out if (1) performance on a measure of reading span (RS) is related to preferred binaural noise reduction (NR) strength, (2) similar relations exist for two different, non-verbal measures of executive function, (3) pure-tone average hearing loss (PTA), signal-to-noise ratio (SNR), and microphone directionality (DIR) also influence preferred NR strength, and (4) preference and speech recognition outcomes are similar. Sixty elderly HA users took part. Six HA conditions consisting of omnidirectional or cardioid microphones followed by inactive, moderate, or strong binaural NR as well as linear amplification were tested. Outcome was assessed at fixed SNRs using headphone simulations of a frontal target talker in a busy cafeteria. Analyses showed positive effects of active NR and DIR on preference, and negative and positive effects of, respectively, strong NR and DIR on speech recognition. Also, while moderate NR was the most preferred NR setting overall, preference for strong NR increased with SNR. No relation between RS and preference was found. However, larger PTA was related to weaker preference for inactive NR and stronger preference for strong NR for both microphone modes. Equivalent (but weaker) relations between worse performance on one non-verbal measure of executive function and the HA conditions without DIR were found. For speech recognition, there were relations between HA condition, PTA, and RS, but their pattern differed from that for preference. Altogether, these results indicate that, while moderate NR works well in general, a notable proportion of HA users prefer stronger NR. Furthermore, PTA and executive functions can account for some of the variability in preference for, and speech recognition with, different binaural NR and DIR settings. PMID:25538547

Piezoelectric Film.

ERIC Educational Resources Information Center

Garrison, Steve

1992-01-01

Presents activities that utilize piezoelectric film to familiarize students with fundamental principles of electricity. Describes classroom projects involving chemical sensors, microbalances, microphones, switches, infrared sensors, and power generation. (MDH)

Sound vibration signal processing for detection and identification detonation (knock) to optimize performance Otto engine

NASA Astrophysics Data System (ADS)

Sujono, A.; Santoso, B.; Juwana, W. E.

2016-03-01

Problems of detonation (knock) on Otto engine (petrol engine) is completely unresolved problem until now, especially if want to improve the performance. This research did sound vibration signal processing engine with a microphone sensor, for the detection and identification of detonation. A microphone that can be mounted is not attached to the cylinder block, that's high temperature, so that its performance will be more stable, durable and inexpensive. However, the method of analysis is not very easy, because a lot of noise (interference). Therefore the use of new methods of pattern recognition, through filtration, and the regression function normalized envelope. The result is quite good, can achieve a success rate of about 95%.

An active drop counting device using condenser microphone for superheated emulsion detector

DOE Office of Scientific and Technical Information (OSTI.GOV)

Das, Mala; Marick, C.; Kanjilal, D.

2008-11-15

An active device for superheated emulsion detector is described. A capacitive diaphragm sensor or condenser microphone is used to convert the acoustic pulse of drop nucleation to electrical signal. An active peak detector is included in the circuit to avoid multiple triggering of the counter. The counts are finally recorded by a microprocessor based data acquisition system. Genuine triggers, missed by the sensor, were studied using a simulated clock pulse. The neutron energy spectrum of {sup 252}Cf fission neutron source was measured using the device with R114 as the sensitive liquid and compared with the calculated fission neutron energy spectrummore » of {sup 252}Cf. Frequency analysis of the detected signals was also carried out.« less

Wearable knee health rehabilitation assessment using acoustical emissions

NASA Astrophysics Data System (ADS)

Teague, Caitlin N.; Hersek, Sinan; Conant, Jordan L.; Gilliland, Scott M.; Inan, Omer T.

2017-02-01

We have developed a novel, wearable sensing system based on miniature piezoelectric contact microphones for measuring the acoustical emissions from the knee during movement. The system consists of two contact microphones, positioned on the medial and lateral sides of the patella, connected to custom, analog pre-amplifier circuits and a microcontroller for digitization and data storage on a secure digital card. Tn addition to the acoustical sensing, the system includes two integrated inertial measurement sensors including accelerometer and gyroscope modalities to enable joint angle calculations; these sensors, with digital outputs, are connected directly to the same microcontroller. The system provides low noise, accurate joint acoustical emission and angle measurements in a wearable form factor and has several hours of battery life.

An active drop counting device using condenser microphone for superheated emulsion detector

NASA Astrophysics Data System (ADS)

Das, Mala; Arya, A. S.; Marick, C.; Kanjilal, D.; Saha, S.

2008-11-01

An active device for superheated emulsion detector is described. A capacitive diaphragm sensor or condenser microphone is used to convert the acoustic pulse of drop nucleation to electrical signal. An active peak detector is included in the circuit to avoid multiple triggering of the counter. The counts are finally recorded by a microprocessor based data acquisition system. Genuine triggers, missed by the sensor, were studied using a simulated clock pulse. The neutron energy spectrum of C252f fission neutron source was measured using the device with R114 as the sensitive liquid and compared with the calculated fission neutron energy spectrum of C252f. Frequency analysis of the detected signals was also carried out.

Probe-tube microphone measures in hearing-impaired children and adults.

PubMed

Barlow, N L; Auslander, M C; Rines, D; Stelmachowicz, P G

1988-10-01

This study was designed to investigate the reliability of real-ear measurements of sound pressure level (SPL) and to compare these values with two coupler measures of SPL. A commercially available probe tube microphone system was used to measure real ear SPL in both children and adults. Test-retest reliability decreased as a function of frequency for both groups and, in general, was slightly poorer for the children. For both groups, coupler to real ear differences were larger for the 2 cm3 coupler than for the reduced volume coupler; however, no significant differences were observed between groups. In addition, a measure of ear canal volume was not found to be a good predictor of coupler to real ear discrepancies.

Evaluating the Acoustic Effect of Over-the-Rotor Foam-Metal Liner Installed on a Low Speed Fan Using Virtual Rotating Microphone Imaging

NASA Technical Reports Server (NTRS)

Sutliff, Daniel L.; Dougherty, Robert P.; Walker, Bruce E.

2010-01-01

An in-duct beamforming technique for imaging rotating broadband fan sources has been used to evaluate the acoustic characteristics of a Foam-Metal Liner installed over-the-rotor of a low-speed fan. The NASA Glenn Research Center s Advanced Noise Control Fan was used as a test bed. A duct wall-mounted phased array consisting of several rings of microphones was employed. The data are mathematically resampled in the fan rotating reference frame and subsequently used in a conventional beamforming technique. The steering vectors for the beamforming technique are derived from annular duct modes, so that effects of reflections from the duct walls are reduced.

Photoacoustic and Photothermal Effects in Particulate Suspensions

DOE Office of Scientific and Technical Information (OSTI.GOV)

Diebold, Gerald, J.

A summary of the research areas investigated by the author during the grant period is given. Experiments and theory have been carried out on the photoacoustic effect arising from a number of physical and chemical processes. A number of studies of the photoacoustic effect as it occurs in transient grating experiments have been completed. The research done with the Ludwig-Soret effect on the generation of shock waves is reported. Other research, such as that carried out on interferometric and beam deflection microphones, the use of microphones in vacuum as momentum flux detectors, and chemical generation of sonoluminescence is listed. Amore » list of published research including selected publications, a complete list of journal articles, books, review articles, and reviews are given.« less

Report on key comparison COOMET.AUV.A-K5: pressure calibration of laboratory standard microphones in the frequency range 2 Hz to 10 kHz

NASA Astrophysics Data System (ADS)

Dobrowolska, D.; Kosterov, A.

2016-01-01

This is the final report for regional key comparison COOMET.AUV.A-K5 on the pressure calibration of laboratory standard microphones in the frequency range from 2 Hz to 10 kHz. Two laboratories—Central Office of Measures (GUM)—the national metrology institute for Poland and the State Enterprise Scientific-Research Institute for Metrology of Measurement and Control Systems (DP NDI Systema)— the designated institute for acoustics in Ukraine took part in this comparison with the GUM as a pilot. One travelling type LS1P microphone was circulated to the participants and results in the form of regular calibration certificates were collected. The results of the DP NDI Systema obtained in this comparison were linked to the CCAUV.A-K5 key comparison through the joint participation of the GUM. The degrees of equivalence were computed for DP NDI Systema with respect to the CCAUV.A-K5 key comparison reference value. Main text To reach the main text of this paper, click on Final Report. Note that this text is that which appears in Appendix B of the BIPM key comparison database kcdb.bipm.org/. The final report has been peer-reviewed and approved for publication by the CCAUV, according to the provisions of the CIPM Mutual Recognition Arrangement (CIPM MRA).

Azimuthal sound localization in the European starling (Sturnus vulgaris): I. Physical binaural cues.

PubMed

Klump, G M; Larsen, O N

1992-02-01

The physical measurements reported here test whether the European starling (Sturnus vulgaris) evaluates the azimuth direction of a sound source with a peripheral auditory system composed of two acoustically coupled pressure-difference receivers (1) or of two decoupled pressure receivers (2). A directional pattern of sound intensity in the free-field was measured at the entrance of the auditory meatus using a probe microphone, and at the tympanum using laser vibrometry. The maximum differences in the sound-pressure level measured with the microphone between various speaker positions and the frontal speaker position were 2.4 dB at 1 and 2 kHz, 7.3 dB at 4 kHz, 9.2 dB at 6 kHz, and 10.9 dB at 8 kHz. The directional amplitude pattern measured by laser vibrometry did not differ from that measured with the microphone. Neither did the directional pattern of travel times to the ear. Measurements of the amplitude and phase transfer function of the starling's interaural pathway using a closed sound system were in accord with the results of the free-field measurements. In conclusion, although some sound transmission via the interaural canal occurred, the present experiments support the hypothesis 2 above that the starling's peripheral auditory system is best described as consisting of two functionally decoupled pressure receivers.

Head-mounted active noise control system with virtual sensing technique

NASA Astrophysics Data System (ADS)

Miyazaki, Nobuhiro; Kajikawa, Yoshinobu

2015-03-01

In this paper, we apply a virtual sensing technique to a head-mounted active noise control (ANC) system we have already proposed. The proposed ANC system can reduce narrowband noise while improving the noise reduction ability at the desired locations. A head-mounted ANC system based on an adaptive feedback structure can reduce noise with periodicity or narrowband components. However, since quiet zones are formed only at the locations of error microphones, an adequate noise reduction cannot be achieved at the locations where error microphones cannot be placed such as near the eardrums. A solution to this problem is to apply a virtual sensing technique. A virtual sensing ANC system can achieve higher noise reduction at the desired locations by measuring the system models from physical sensors to virtual sensors, which will be used in the online operation of the virtual sensing ANC algorithm. Hence, we attempt to achieve the maximum noise reduction near the eardrums by applying the virtual sensing technique to the head-mounted ANC system. However, it is impossible to place the microphone near the eardrums. Therefore, the system models from physical sensors to virtual sensors are estimated using the Head And Torso Simulator (HATS) instead of human ears. Some simulation, experimental, and subjective assessment results demonstrate that the head-mounted ANC system with virtual sensing is superior to that without virtual sensing in terms of the noise reduction ability at the desired locations.

Electret accelerometers: physics and dynamic characterization.

PubMed

Hillenbrand, J; Haberzettl, S; Motz, T; Sessler, G M

2011-06-01

Electret microphones are produced in numbers that significantly exceed those for all other microphone types. This is due to the fact that air-borne electret sensors are of simple and low-cost design but have very good acoustical properties. In contrast, most of the discrete structure-borne sound sensors (or accelerometers) are based on the piezoelectric effect. In the present work, capacitive accelerometers utilizing the electret principle were constructed, built, and characterized. These electret accelerometers comprise a metallic seismic mass, covered by an electret film, a ring of a soft cellular polymer supplying the restoring force, and a metallic backplate. These components replace membrane, spacer, and back electrode, respectively, of the electret microphone. An adjustable static pressure to the seismic mass is generated by two metal springs. The dynamic characterization of the accelerometers was carried out by using an electrodynamic shaker and an external charge or voltage amplifier. Sensors with various seismic masses, air gap distances, and electret voltages were investigated. Charge sensitivities from 10 to 40 pC/g, voltage sensitivities from 600 to 2000 mV/g, and resonance frequencies from 3 to 1.5 kHz were measured. A model describing both the charge and the voltage sensitivity is presented. Good agreement of experimental and calculated values is found. The experimental results show that sensitive, lightweight, and inexpensive electret accelerometers can be built. © 2011 Acoustical Society of America

Sound pressure distribution and power flow within the gerbil ear canal from 100 Hz to 80 kHz

PubMed Central

Ravicz, Michael E.; Olson, Elizabeth S.; Rosowski, John J.

2008-01-01

Sound pressure was mapped in the bony ear canal of gerbils during closed-field sound stimulation at frequencies from 0.1 to 80 kHz. A 1.27-mm-diam probe-tube microphone or a 0.17-mm-diam fiber-optic miniature microphone was positioned along approximately longitudinal trajectories within the 2.3-mm-diam ear canal. Substantial spatial variations in sound pressure, sharp minima in magnitude, and half-cycle phase changes occurred at frequencies >30 kHz. The sound frequencies of these transitions increased with decreasing distance from the tympanic membrane (TM). Sound pressure measured orthogonally across the surface of the TM showed only small variations at frequencies below 60 kHz. Hence, the ear canal sound field can be described fairly well as a one-dimensional standing wave pattern. Ear-canal power reflectance estimated from longitudinal spatial variations was roughly constant at 0.2–0.5 at frequencies between 30 and 45 kHz. In contrast, reflectance increased at higher frequencies to at least 0.8 above 60 kHz. Sound pressure was also mapped in a microphone-terminated uniform tube—an “artificial ear.” Comparison with ear canal sound fields suggests that an artificial ear or “artificial cavity calibration” technique may underestimate the in situ sound pressure by 5–15 dB between 40 and 60 kHz. PMID:17902852

Simulation of Thin-Film Damping and Thermal Mechanical Noise Spectra for Advanced Micromachined Microphone Structures.

PubMed

Hall, Neal A; Okandan, Murat; Littrell, Robert; Bicen, Baris; Degertekin, F Levent

2008-06-01

In many micromachined sensors the thin (2-10 μm thick) air film between a compliant diaphragm and backplate electrode plays a dominant role in shaping both the dynamic and thermal noise characteristics of the device. Silicon microphone structures used in grating-based optical-interference microphones have recently been introduced that employ backplates with minimal area to achieve low damping and low thermal noise levels. Finite-element based modeling procedures based on 2-D discretization of the governing Reynolds equation are ideally suited for studying thin-film dynamics in such structures which utilize relatively complex backplate geometries. In this paper, the dynamic properties of both the diaphragm and thin air film are studied using a modal projection procedure in a commonly used finite element software and the results are used to simulate the dynamic frequency response of the coupled structure to internally generated electrostatic actuation pressure. The model is also extended to simulate thermal mechanical noise spectra of these advanced sensing structures. In all cases simulations are compared with measured data and show excellent agreement-demonstrating 0.8 pN/√Hz and 1.8 μPa/√Hz thermal force and thermal pressure noise levels, respectively, for the 1.5 mm diameter structures under study which have a fundamental diaphragm resonance-limited bandwidth near 20 kHz.

Measurement of Phased Array Point Spread Functions for Use with Beamforming

NASA Technical Reports Server (NTRS)

Bahr, Chris; Zawodny, Nikolas S.; Bertolucci, Brandon; Woolwine, Kyle; Liu, Fei; Li, Juan; Sheplak, Mark; Cattafesta, Louis

2011-01-01

Microphone arrays can be used to localize and estimate the strengths of acoustic sources present in a region of interest. However, the array measurement of a region, or beam map, is not an accurate representation of the acoustic field in that region. The true acoustic field is convolved with the array s sampling response, or point spread function (PSF). Many techniques exist to remove the PSF's effect on the beam map via deconvolution. Currently these methods use a theoretical estimate of the array point spread function and perhaps account for installation offsets via determination of the microphone locations. This methodology fails to account for any reflections or scattering in the measurement setup and still requires both microphone magnitude and phase calibration, as well as a separate shear layer correction in an open-jet facility. The research presented seeks to investigate direct measurement of the array's PSF using a non-intrusive acoustic point source generated by a pulsed laser system. Experimental PSFs of the array are computed for different conditions to evaluate features such as shift-invariance, shear layers and model presence. Results show that experimental measurements trend with theory with regard to source offset. The source shows expected behavior due to shear layer refraction when observed in a flow, and application of a measured PSF to NACA 0012 aeroacoustic trailing-edge noise data shows a promising alternative to a classic shear layer correction method.

Identification of Noise Sources During Rocket Engine Test Firings and a Rocket Launch Using a Microphone Phased-Array

NASA Technical Reports Server (NTRS)

Panda, Jayanta; Mosher, Robert N.; Porter, Barry J.

2013-01-01

A 70 microphone, 10-foot by 10-foot, microphone phased array was built for use in the harsh environment of rocket launches. The array was setup at NASA Wallops launch pad 0A during a static test firing of Orbital Sciences' Antares engines, and again during the first launch of the Antares vehicle. It was placed 400 feet away from the pad, and was hoisted on a scissor lift 40 feet above ground. The data sets provided unprecedented insight into rocket noise sources. The duct exit was found to be the primary source during the static test firing; the large amount of water injected beneath the nozzle exit and inside the plume duct quenched all other sources. The maps of the noise sources during launch were found to be time-dependent. As the engines came to full power and became louder, the primary source switched from the duct inlet to the duct exit. Further elevation of the vehicle caused spilling of the hot plume, resulting in a distributed noise map covering most of the pad. As the entire plume emerged from the duct, and the ondeck water system came to full power, the plume itself became the loudest noise source. These maps of the noise sources provide vital insight for optimization of sound suppression systems for future Antares launches.

36 CFR 1193.51 - Compatibility.

Code of Federal Regulations, 2010 CFR

2010-07-01

... BOARD TELECOMMUNICATIONS ACT ACCESSIBILITY GUIDELINES Requirements for Compatibility With Peripheral... user to easily turn any microphone on and off to allow the user to intermix speech with TTY use. (e...

36 CFR 1193.51 - Compatibility.

Code of Federal Regulations, 2011 CFR

2011-07-01

... BOARD TELECOMMUNICATIONS ACT ACCESSIBILITY GUIDELINES Requirements for Compatibility With Peripheral... user to easily turn any microphone on and off to allow the user to intermix speech with TTY use. (e...

Reaching Out.

ERIC Educational Resources Information Center

Porter, Randall C.

1999-01-01

Discusses technology and equipment requirements for developing an effective distance-learning classroom. Areas covered include cabling, the control booth, microphones, acoustics, lighting, heating and air conditioning, cameras, video monitors, staffing, and power requirements. (GR)

Wheel/rail noise generated by a high-speed train investigated with a line array of microphones

NASA Astrophysics Data System (ADS)

Barsikow, B.; King, W. F.; Pfizenmaier, E.

1987-10-01

Radiated noise generated by a high-speed electric train travelling at speeds up to 250 km/h has been measured with a line array of microphones mounted along the wayside in two different orientations. The test train comprised a 103 electric locomotive, four Intercity coaches, and a dynamo coach. Some of the wheels were fitted with experimental wheel-noise absorbers. By using the directional capabilities of the array, the locations of the dominant sources of wheel/rail radiated noise were identified on the wheels. For conventional wheels, these sources lie near or on the rim at an average height of about 0·2 m above the railhead. The effect of wheel-noise absorbers and freshly turned treads on radiated noise were also investigated.

Moving microphone arrays to reduce spatial aliasing in the beamforming technique: theoretical background and numerical investigation.

PubMed

Cigada, Alfredo; Lurati, Massimiliano; Ripamonti, Francesco; Vanali, Marcello

2008-12-01

This paper introduces a measurement technique aimed at reducing or possibly eliminating the spatial aliasing problem in the beamforming technique. Beamforming main disadvantages are a poor spatial resolution, at low frequency, and the spatial aliasing problem, at higher frequency, leading to the identification of false sources. The idea is to move the microphone array during the measurement operation. In this paper, the proposed approach is theoretically and numerically investigated by means of simple sound propagation models, proving its efficiency in reducing the spatial aliasing. A number of different array configurations are numerically investigated together with the most important parameters governing this measurement technique. A set of numerical results concerning the case of a planar rotating array is shown, together with a first experimental validation of the method.

Characteristics of propeller noise on an aircraft fuselage related to interior noise transmission

NASA Technical Reports Server (NTRS)

Mixson, J. S.; Barton, C. K.; Piersol, A. G.; Wilby, J. F.

1979-01-01

Exterior noise was measured on the fuselage of a twin-engine, light aircraft at four values of engine rpm in ground static tests and at forward speeds up to 36 m/s in taxi tests. Propeller noise levels, spectra, and correlations were determined using a horizontal array of seven flush-mounted microphones and a vertical array of four flush-mounted microphones in the propeller plane. The measured levels and spectra are compared with predictions based on empirical and analytical methods for static and taxi conditions. Trace wavelengths of the propeller noise field, obtained from point-to-point correlations, are compared with the aircraft sidewall structural dimensions, and some analytical results are presented that suggest the sensitivity of interior noise transmission to variations of the propeller noise characteristics.

Locating and Quantifying Broadband Fan Sources Using In-Duct Microphones

NASA Technical Reports Server (NTRS)

Dougherty, Robert P.; Walker, Bruce E.; Sutliff, Daniel L.

2010-01-01

In-duct beamforming techniques have been developed for locating broadband noise sources on a low-speed fan and quantifying the acoustic power in the inlet and aft fan ducts. The NASA Glenn Research Center's Advanced Noise Control Fan was used as a test bed. Several of the blades were modified to provide a broadband source to evaluate the efficacy of the in-duct beamforming technique. Phased arrays consisting of rings and line arrays of microphones were employed. For the imaging, the data were mathematically resampled in the frame of reference of the rotating fan. For both the imaging and power measurement steps, array steering vectors were computed using annular duct modal expansions, selected subsets of the cross spectral matrix elements were used, and the DAMAS and CLEAN-SC deconvolution algorithms were applied.

49 CFR 229.129 - Locomotive horn.

Code of Federal Regulations, 2011 CFR

2011-10-01

..., such as barriers, hills, billboards, tractor trailers or other large vehicles, locomotives or rail cars.... The observer shall not stand between the microphone and the horn. (8) Background noise shall be...

Implantable digital hearing aid

NASA Technical Reports Server (NTRS)

Kissiah, A. M., Jr.

1979-01-01

Hearing aid converts analog output of microphone into digital pulses in about 10 channels of audiofrequencies. Each pulse band could be directly connected to portion of auditory nerve most sensitive to that range.

Wireless Microphone Users Interference Protection Act

THOMAS, 111th Congress

Rep. Rush, Bobby L. [D-IL-1

2009-12-16

House - 12/19/2009 Referred to the Subcommittee on Communications, Technology, and the Internet. (All Actions) Tracker: This bill has the status IntroducedHere are the steps for Status of Legislation:

Types of Hearing Aids

MedlinePlus

... hearing impairment. Most hearing aids share several similar electronic components, including a microphone that picks up sound; ... the ear canal; and batteries that power the electronic parts. Hearing aids differ by: design technology used ...

Design and Evaluation of Modifications to the NASA Langley Flow Impedance Tube

NASA Technical Reports Server (NTRS)

Jones, Michael G.; Watson, Willie R.; Parrott, Tony L.; Smith, Charles D.

2004-01-01

The need to minimize fan noise radiation from commercial aircraft engine nacelles continues to provide an impetus for developing new acoustic liner concepts. If the full value of such concepts is to be attained, an understanding of grazing flow effects is crucial. Because of this need for improved understanding of grazing flow effects, the NASA Langley Research Center Liner Physics Group has invested a large effort over the past decade into the development of a 2-D finite element method that characterizes wave propagation through a lined duct. The original test section in the Langley Grazing IncidenceTube was used to acquire data needed for implementation of this finite element method. This test section employed a stepper motor-driven axial-traversing bar, embedded in the wall opposite the test liner, to position a flush-mounted microphone at pre-selected locations. Complex acoustic pressure data acquired with this traversing microphone were used to educe the acoustic impedance of test liners using this 2-D finite element method and a local optimization technique. Results acquired in this facility have been extensively reported, and were compared with corresponding results from various U.S. aeroacoustics laboratories in the late 1990 s. Impedance data comparisons acquired from this multi-laboratory study suggested that it would be valuable to incorporate more realistic 3-D aeroacoustic effects into the impedance eduction methodology. This paper provides a description of modifications that have been implemented to facilitate studies of 3-D effects. The two key features of the modified test section are (1) the replacement of the traversing bar and its flush-mounted microphone with an array of 95 fixed-location microphones that are flush-mounted in all four walls of the duct, and (2) the inclusion of a suction device to modify the boundary layer upstream of the lined portion of the duct. The initial results achieved with the modified test section are provided in this report, and a comparison of these results with those achieved using the original test section is used to demonstrate that the data acquisition and analysis with the new test section can be confidently used for impedance eduction.

Spatial acoustic signal processing for immersive communication

NASA Astrophysics Data System (ADS)

Atkins, Joshua

Computing is rapidly becoming ubiquitous as users expect devices that can augment and interact naturally with the world around them. In these systems it is necessary to have an acoustic front-end that is able to capture and reproduce natural human communication. Whether the end point is a speech recognizer or another human listener, the reduction of noise, reverberation, and acoustic echoes are all necessary and complex challenges. The focus of this dissertation is to provide a general method for approaching these problems using spherical microphone and loudspeaker arrays.. In this work, a theory of capturing and reproducing three-dimensional acoustic fields is introduced from a signal processing perspective. In particular, the decomposition of the spatial part of the acoustic field into an orthogonal basis of spherical harmonics provides not only a general framework for analysis, but also many processing advantages. The spatial sampling error limits the upper frequency range with which a sound field can be accurately captured or reproduced. In broadband arrays, the cost and complexity of using multiple transducers is an issue. This work provides a flexible optimization method for determining the location of array elements to minimize the spatial aliasing error. The low frequency array processing ability is also limited by the SNR, mismatch, and placement error of transducers. To address this, a robust processing method is introduced and used to design a reproduction system for rendering over arbitrary loudspeaker arrays or binaurally over headphones. In addition to the beamforming problem, the multichannel acoustic echo cancellation (MCAEC) issue is also addressed. A MCAEC must adaptively estimate and track the constantly changing loudspeaker-room-microphone response to remove the sound field presented over the loudspeakers from that captured by the microphones. In the multichannel case, the system is overdetermined and many adaptive schemes fail to converge to the true impulse response. This forces the need to track both the near and far end room responses. A transform domain method that mitigates this problem is derived and implemented. Results with a real system using a 16-channel loudspeaker array and 32-channel microphone array are presented.

47 CFR 74.801 - Definitions.

Code of Federal Regulations, 2011 CFR

2011-10-01

... Definitions. Cable television system operator. A cable television operator is defined in § 76.5(cc) of the... transmit over distances of approximately 100 meters for uses such as wireless microphones, cue and control...

47 CFR 74.801 - Definitions.

Code of Federal Regulations, 2012 CFR

2012-10-01

... Definitions. Cable television system operator. A cable television operator is defined in § 76.5(cc) of the... transmit over distances of approximately 100 meters for uses such as wireless microphones, cue and control...

47 CFR 74.801 - Definitions.

Code of Federal Regulations, 2010 CFR

2010-10-01

... Definitions. Cable television system operator. A cable television operator is defined in § 76.5(cc) of the... transmit over distances of approximately 100 meters for uses such as wireless microphones, cue and control...

Identification and tracking of particular speaker in noisy environment

NASA Astrophysics Data System (ADS)

Sawada, Hideyuki; Ohkado, Minoru

2004-10-01

Human is able to exchange information smoothly using voice under different situations such as noisy environment in a crowd and with the existence of plural speakers. We are able to detect the position of a source sound in 3D space, extract a particular sound from mixed sounds, and recognize who is talking. By realizing this mechanism with a computer, new applications will be presented for recording a sound with high quality by reducing noise, presenting a clarified sound, and realizing a microphone-free speech recognition by extracting particular sound. The paper will introduce a realtime detection and identification of particular speaker in noisy environment using a microphone array based on the location of a speaker and the individual voice characteristics. The study will be applied to develop an adaptive auditory system of a mobile robot which collaborates with a factory worker.

Motion measurement of acoustically levitated object

NASA Technical Reports Server (NTRS)

Watkins, John L. (Inventor); Barmatz, Martin B. (Inventor)

1993-01-01

A system is described for determining motion of an object that is acoustically positioned in a standing wave field in a chamber. Sonic energy in the chamber is sensed, and variation in the amplitude of the sonic energy is detected, which is caused by linear motion, rotational motion, or drop shape oscillation of the object. Apparatus for detecting object motion can include a microphone coupled to the chamber and a low pass filter connected to the output of the microphone, which passes only frequencies below the frequency of sound produced by a transducer that maintains the acoustic standing wave field. Knowledge about object motion can be useful by itself, can be useful to determine surface tension, viscosity, and other information about the object, and can be useful to determine the pressure and other characteristics of the acoustic field.

Speech enhancement on smartphone voice recording

NASA Astrophysics Data System (ADS)

Tris Atmaja, Bagus; Nur Farid, Mifta; Arifianto, Dhany

2016-11-01

Speech enhancement is challenging task in audio signal processing to enhance the quality of targeted speech signal while suppress other noises. In the beginning, the speech enhancement algorithm growth rapidly from spectral subtraction, Wiener filtering, spectral amplitude MMSE estimator to Non-negative Matrix Factorization (NMF). Smartphone as revolutionary device now is being used in all aspect of life including journalism; personally and professionally. Although many smartphones have two microphones (main and rear) the only main microphone is widely used for voice recording. This is why the NMF algorithm widely used for this purpose of speech enhancement. This paper evaluate speech enhancement on smartphone voice recording by using some algorithms mentioned previously. We also extend the NMF algorithm to Kulback-Leibler NMF with supervised separation. The last algorithm shows improved result compared to others by spectrogram and PESQ score evaluation.

Measurement of sound emitted by flying projectiles with aeroacoustic sources

NASA Technical Reports Server (NTRS)

Cho, Y. I.; Shakkottai, P.; Harstad, K. G.; Back, L. H.

1988-01-01

Training projectiles with axisymmetric ring cavities that produce intense tones in an airstream were shot in a straight-line trajectory. A ground-based microphone was used to obtain the angular distribution of sound intensity produced from the flying projectile. Data reduction required calculation of Doppler and attenuation factors. Also, the directional sensitivity of the ground-mounted microphone was measured and used in the data reduction. A rapid angular variation of sound intensity produced from the projectile was found that can be used to plot an intensity contour map on the ground. A full-scale field test confirmed the validity of the aeroacoustic concept of producing a relatively intense whistle from the projectile, and the usefulness of short-range flight tests that yield acoustic data free of uncertainties associated with diffraction, reflection, and refraction at jet boundaries in free-jet tests.

A Sparsity-Based Approach to 3D Binaural Sound Synthesis Using Time-Frequency Array Processing

NASA Astrophysics Data System (ADS)

Cobos, Maximo; Lopez, JoseJ; Spors, Sascha

2010-12-01

Localization of sounds in physical space plays a very important role in multiple audio-related disciplines, such as music, telecommunications, and audiovisual productions. Binaural recording is the most commonly used method to provide an immersive sound experience by means of headphone reproduction. However, it requires a very specific recording setup using high-fidelity microphones mounted in a dummy head. In this paper, we present a novel processing framework for binaural sound recording and reproduction that avoids the use of dummy heads, which is specially suitable for immersive teleconferencing applications. The method is based on a time-frequency analysis of the spatial properties of the sound picked up by a simple tetrahedral microphone array, assuming source sparseness. The experiments carried out using simulations and a real-time prototype confirm the validity of the proposed approach.

Vehicle Counting and Moving Direction Identification Based on Small-Aperture Microphone Array.

PubMed

Zu, Xingshui; Zhang, Shaojie; Guo, Feng; Zhao, Qin; Zhang, Xin; You, Xing; Liu, Huawei; Li, Baoqing; Yuan, Xiaobing

2017-05-10

The varying trend of a moving vehicle's angles provides much important intelligence for an unattended ground sensor (UGS) monitoring system. The present study investigates the capabilities of a small-aperture microphone array (SAMA) based system to identify the number and moving direction of vehicles travelling on a previously established route. In this paper, a SAMA-based acoustic monitoring system, including the system hardware architecture and algorithm mechanism, is designed as a single node sensor for the application of UGS. The algorithm is built on the varying trend of a vehicle's bearing angles around the closest point of approach (CPA). We demonstrate the effectiveness of our proposed method with our designed SAMA-based monitoring system in various experimental sites. The experimental results in harsh conditions validate the usefulness of our proposed UGS monitoring system.

Selective Listening Point Audio Based on Blind Signal Separation and Stereophonic Technology

NASA Astrophysics Data System (ADS)

Niwa, Kenta; Nishino, Takanori; Takeda, Kazuya

A sound field reproduction method is proposed that uses blind source separation and a head-related transfer function. In the proposed system, multichannel acoustic signals captured at distant microphones are decomposed to a set of location/signal pairs of virtual sound sources based on frequency-domain independent component analysis. After estimating the locations and the signals of the virtual sources by convolving the controlled acoustic transfer functions with each signal, the spatial sound is constructed at the selected point. In experiments, a sound field made by six sound sources is captured using 48 distant microphones and decomposed into sets of virtual sound sources. Since subjective evaluation shows no significant difference between natural and reconstructed sound when six virtual sources and are used, the effectiveness of the decomposing algorithm as well as the virtual source representation are confirmed.

Design of an Acoustic Target Intrusion Detection System Based on Small-Aperture Microphone Array.

PubMed

Zu, Xingshui; Guo, Feng; Huang, Jingchang; Zhao, Qin; Liu, Huawei; Li, Baoqing; Yuan, Xiaobing

2017-03-04

Automated surveillance of remote locations in a wireless sensor network is dominated by the detection algorithm because actual intrusions in such locations are a rare event. Therefore, a detection method with low power consumption is crucial for persistent surveillance to ensure longevity of the sensor networks. A simple and effective two-stage algorithm composed of energy detector (ED) and delay detector (DD) with all its operations in time-domain using small-aperture microphone array (SAMA) is proposed. The algorithm analyzes the quite different velocities between wind noise and sound waves to improve the detection capability of ED in the surveillance area. Experiments in four different fields with three types of vehicles show that the algorithm is robust to wind noise and the probability of detection and false alarm are 96.67% and 2.857%, respectively.

An experimental investigation of the interior noise control effects of propeller synchrophasing

NASA Technical Reports Server (NTRS)

Jones, J. D.; Fuller, C. R.

1986-01-01

A simplified cylindrical model of an aircraft fuselage is used to investigate the mechanisms of interior noise suppression using synchrophasing techniques. This investigation allows isolation of important parameters to define the characteristics of synchrophasing. The optimum synchrophase angle for maximum noise reduction is found for several interior microphone positions with pure tone source excitation. Noise reductions of up to 30 dB are shown for some microphone positions, however, overall reductions are less. A computer algorithm is developed to decompose the cylinder vibration into modal components over a wide range of synchrophase angles. The circumferential modal response of the shell vibration is shown to govern the transmission of sound into the cylinder rather than localized transmission. As well as investigating synchrophasing, the interior sound field due to sources typical of propellers has been measured and discussed.

Neurodynamic evaluation of hearing aid features using EEG correlates of listening effort.

PubMed

Bernarding, Corinna; Strauss, Daniel J; Hannemann, Ronny; Seidler, Harald; Corona-Strauss, Farah I

2017-06-01

In this study, we propose a novel estimate of listening effort using electroencephalographic data. This method is a translation of our past findings, gained from the evoked electroencephalographic activity, to the oscillatory EEG activity. To test this technique, electroencephalographic data from experienced hearing aid users with moderate hearing loss were recorded, wearing hearing aids. The investigated hearing aid settings were: a directional microphone combined with a noise reduction algorithm in a medium and a strong setting, the noise reduction setting turned off, and a setting using omnidirectional microphones without any noise reduction. The results suggest that the electroencephalographic estimate of listening effort seems to be a useful tool to map the exerted effort of the participants. In addition, the results indicate that a directional processing mode can reduce the listening effort in multitalker listening situations.

Acoustic change detection algorithm using an FM radio

NASA Astrophysics Data System (ADS)

Goldman, Geoffrey H.; Wolfe, Owen

2012-06-01

The U.S. Army is interested in developing low-cost, low-power, non-line-of-sight sensors for monitoring human activity. One modality that is often overlooked is active acoustics using sources of opportunity such as speech or music. Active acoustics can be used to detect human activity by generating acoustic images of an area at different times, then testing for changes among the imagery. A change detection algorithm was developed to detect physical changes in a building, such as a door changing positions or a large box being moved using acoustics sources of opportunity. The algorithm is based on cross correlating the acoustic signal measured from two microphones. The performance of the algorithm was shown using data generated with a hand-held FM radio as a sound source and two microphones. The algorithm could detect a door being opened in a hallway.

Comparison of acoustic shock waves generated by micro and nanosecond lasers for a smart laser surgery system

NASA Astrophysics Data System (ADS)

Nguendon Kenhagho, Hervé K.; Rauter, Georg; Guzman, Raphael; C. Cattin, Philippe; Zam, Azhar

2018-02-01

Characterization of acoustic shock wave will guarantee efficient tissue differentiation as feedback to reduce the probability of undesirable damaging (i.e. cutting) of tissues in laser surgery applications. We ablated hard (bone) and soft (muscle) tissues using a nanosecond pulsed Nd:YAG laser at 532 nm and a microsecond pulsed Er:YAG laser at 2.94 μm. When the intense short ns-pulsed laser is applied to material, the energy gain causes locally a plasma at the ablated spot that expands and propagates as an acoustic shock wave with a rarefaction wave behind the shock front. However, when using a μs-pulsed Er:YAG laser for material ablation, the acoustic shock wave is generated during the explosion of the ablated material. We measured and compared the emitted acoustic shock wave generated by a ns-pulsed Nd:YAG laser and a μs-pulsed Er:YAG laser measured by a calibrated microphone. As the acoustic shock wave attenuates as it propagates through air, the distance between ablation spots and a calibrated microphone was at 5 cm. We present the measurements on the propagation characteristics of the laser generated acoustic shock wave by measuring the arrival time-of-flight with a calibrated microphone and the energy-dependent evolution of acoustic parameters such as peak-topeak pressure, the ratio of the peak-to-peak pressures for the laser induced breakdown in air, the ablated muscle and the bone, and the spectral energy.

Cochlear microphonic responses to acoustic clicks in guinea pig and their relation with microphonic responses to pure tones.

PubMed

Echeverría, E L; Robles, L W

1983-02-01

Cochlear microphonic (CM) responses to acoustic transient stimuli were studied at the three more basal turns of the cochlea in the guinea pig. The responses to rarefaction and condensation pressure pulses of less than 100-mus duration were recorded using the differential electrode technique. In some animals the CM response to pure tones was recorded at the same position at which the transient response was obtained. The transient responses recorded at the three turns of the cochlea displayed a damped oscillation at a frequency consistent with the values of cutoff frequency already known for the electrode positions. Some of the responses were significantly less damped than click responses previously reported. There was a good correlation between the cutoff frequency in the frequency response curve and the frequency of oscillation in the transient response for recordings obtained at the same position in the cochlea. A nonlinear effect was observed for changes in stimulus intensity. There was a less than proportional decrease in amplitude of the initial part of the damped oscillation for a decrease of the stimulus intensity, while the late part of the response behaved almost linearly. This nonlinearity observed in the CM transient response could not be explained by a nonlinear characteristic of the sort reported in the basilar membrane of the squirrel monkey by Robles et al. [J. Acoust. Soc. Am. 59, 926-939 (1976)]; rather it seems to be a saturation nonlinearity similar to the one known for sinusoidal stimulation.

In situ Probe Microphone Measurement for Testing the Direct Acoustical Cochlear Stimulator.

PubMed

Stieger, Christof; Alnufaily, Yasser H; Candreia, Claudia; Caversaccio, Marco D; Arnold, Andreas M

2017-01-01

Hypothesis: Acoustical measurements can be used for functional control of a direct acoustic cochlear stimulator (DACS). Background: The DACS is a recently released active hearing implant that works on the principle of a conventional piston prosthesis driven by the rod of an electromagnetic actuator. An inherent part of the DACS actuator is a thin titanium diaphragm that allows for movement of the stimulation rod while hermetically sealing the housing. In addition to mechanical stimulation, the actuator emits sound into the mastoid cavity because of the motion of the diaphragm. Methods: We investigated the use of the sound emission of a DACS for intra-operative testing. We measured sound emission in the external auditory canal (P EAC ) and velocity of the actuators stimulation rod (V act ) in five implanted ears of whole-head specimens. We tested the influence various positions of the loudspeaker and a probe microphone on P EAC and simulated implant malfunction in one example. Results: Sound emission of the DACS with a signal-to-noise ratio >10 dB was observed between 0.5 and 5 kHz. Simulated implant misplacement or malfunction could be detected by the absence or shift in the characteristic resonance frequency of the actuator. P EAC changed by <6 dB for variations of the microphone and loudspeaker position. Conclusion: Our data support the feasibility of acoustical measurements for in situ testing of the DACS implant in the mastoid cavity as well as for post-operative monitoring of actuator function.

Fast PSP measurements of wall-pressure fluctuation in low-speed flows: improvements using proper orthogonal decomposition

NASA Astrophysics Data System (ADS)

Peng, Di; Wang, Shaofei; Liu, Yingzheng

2016-04-01

Fast pressure-sensitive paint (PSP) is very useful in flow diagnostics due to its fast response and high spatial resolution, but its applications in low-speed flows are usually challenging due to limitations of paint's pressure sensitivity and the capability of high-speed imagers. The poor signal-to-noise ratio in low-speed cases makes it very difficult to extract useful information from the PSP data. In this study, unsteady PSP measurements were made on a flat plate behind a cylinder in a low-speed wind tunnel (flow speed from 10 to 17 m/s). Pressure fluctuations (Δ P) on the plate caused by vortex-plate interaction were recorded continuously by fast PSP (using a high-speed camera) and a microphone array. Power spectrum of pressure fluctuations and phase-averaged Δ P obtained from PSP and microphone were compared, showing good agreement in general. Proper orthogonal decomposition (POD) was used to reduce noise in PSP data and extract the dominant pressure features. The PSP results reconstructed from selected POD modes were then compared to the pressure data obtained simultaneously with microphone sensors. Based on the comparison of both instantaneous Δ P and root-mean-square of Δ P, it was confirmed that POD analysis could effectively remove noise while preserving the instantaneous pressure information with good fidelity, especially for flows with strong periodicity. This technique extends the application range of fast PSP and can be a powerful tool for fundamental fluid mechanics research at low speed.

A comparison between swallowing sounds and vibrations in patients with dysphagia

PubMed Central

Movahedi, Faezeh; Kurosu, Atsuko; Coyle, James L.; Perera, Subashan

2017-01-01

The cervical auscultation refers to the observation and analysis of sounds or vibrations captured during swallowing using either a stethoscope or acoustic/vibratory detectors. Microphones and accelerometers have recently become two common sensors used in modern cervical auscultation methods. There are open questions about whether swallowing signals recorded by these two sensors provide unique or complementary information about swallowing function; or whether they present interchangeable information. The aim of this study is to present a broad comparison of swallowing signals recorded by a microphone and a tri-axial accelerometer from 72 patients (mean age 63.94 ± 12.58 years, 42 male, 30 female), who underwent videofluoroscopic examination. The participants swallowed one or more boluses of thickened liquids of different consistencies, including thin liquids, nectar-thick liquids, and pudding. A comfortable self-selected volume from a cup or a controlled volume by the examiner from a 5ml spoon was given to the participants. A comprehensive set of features was extracted in time, information-theoretic, and frequency domains from each of 881 swallows presented in this study. The swallowing sounds exhibited significantly higher frequency content and kurtosis values than the swallowing vibrations. In addition, the Lempel-Ziv complexity was lower for swallowing sounds than those for swallowing vibrations. To conclude, information provided by microphones and accelerometers about swallowing function are unique and these two transducers are not interchangeable. Consequently, the selection of transducer would be a vital step in future studies. PMID:28495001

Modeling of influencing parameters in active noise control on an enclosure wall

NASA Astrophysics Data System (ADS)

Tarabini, Marco; Roure, Alain

2008-04-01

This paper investigates, by means of a numerical model, the possibility of using an active noise barrier to virtually reduce the acoustic transparency of a partition wall inside an enclosure. The room is modeled with the image method as a rectangular enclosure with a stationary point source; the active barrier is set up by an array of loudspeakers and error microphones and is meant to minimize the squared sound pressure on a wall with the use of a decentralized control. Simulations investigate the effects of the enclosure characteristics and of the barrier geometric parameters on the sound pressure attenuation on the controlled partition, on the whole enclosure potential energy and on the diagonal control stability. Performances are analyzed in a frequency range of 25-300 Hz at discrete 25 Hz steps. Influencing parameters and their effects on the system performances are identified with a statistical inference procedure. Simulation results have shown that it is possible to averagely reduce the sound pressure on the controlled partition. In the investigated configuration, the surface attenuation and the diagonal control stability are mainly driven by the distance between the loudspeakers and the error microphones and by the loudspeakers directivity; minor effects are due to the distance between the error microphones and the wall, by the wall reflectivity and by the active barrier grid meshing. Room dimensions and source position have negligible effects. Experimental results point out the validity of the model and the efficiency of the barrier in the reduction of the wall acoustic transparency.

A novel automated detection system for swallowing sounds during eating and speech under everyday conditions.

PubMed

Fukuike, C; Kodama, N; Manda, Y; Hashimoto, Y; Sugimoto, K; Hirata, A; Pan, Q; Maeda, N; Minagi, S

2015-05-01

The wave analysis of swallowing sounds has been receiving attention because the recording process is easy and non-invasive. However, up until now, an expert has been needed to visually examine the entire recorded wave to distinguish swallowing from other sounds. The purpose of this study was to establish a methodology to automatically distinguish the sound of swallowing from sound data recorded during a meal in the presence of everyday ambient sound. Seven healthy participants (mean age: 26·7 ± 1·3 years) participated in this study. A laryngeal microphone and a condenser microphone attached to the nostril were used for simultaneous recording. Recoding took place while participants were taking a meal and talking with a conversational partner. Participants were instructed to step on a foot pedal trigger switch when they swallowed, representing self-enumeration of swallowing, and also to achieve six additional noise-making tasks during the meal in a randomised manner. The automated analysis system correctly detected 342 out of the 352 self-enumerated swallowing events (sensitivity: 97·2%) and 479 out of the 503 semblable wave periods of swallowing (specificity: 95·2%). In this study, the automated detection system for swallowing sounds using a nostril microphone was able to detect the swallowing event with high sensitivity and specificity even under the conditions of daily life, thus showing potential utility in the diagnosis or screening of dysphagic patients in future studies. © 2014 John Wiley & Sons Ltd.

J-FLiC UAS Flights for Acoustic Testing Research

NASA Technical Reports Server (NTRS)

Motter, Mark A.; High, James W.

2016-01-01

The jet-powered flying testbed (J-FLiC) unmanned aircraft system (UAS) successfully completed twenty-six flights at Fort AP Hill, VA, from 27 August until September 3 2015, supporting tests of a microphone array system for aircraft noise measurement. The test vehicles, J-FLiC NAVY2 (N508NU), and J-FLiC 4 (N509NU), were flown under manual and autopiloted control in a variety of test conditions: clean at speeds ranging from 80 to 150 knots; and full landing configuration at speeds ranging from 50 to 95 knots. During the test campaign, autopilot capability was incrementally improved to ultimately provide a high degree of accuracy and repeatability of the critical test requirements for airspeed, altitude, runway alignment and position over the microphone array. Manual flights were performed for test conditions at the both ends of the speed envelope where autopiloted flight would have required flight beyond visual range and more extensive developmental work. The research objectives of the campaign were fully achieved. The ARMD Integrated Systems Research Program (ISRP) Environmentally Responsible Aviation (ERA) Project aims to develop the enabling capabilities/technologies that will allow prediction/reduction of aircraft noise. A primary measurement tool for ascertaining and characterizing empirically the effectiveness of various noise reduction technologies is a microphone phased array system. Such array systems need to be vetted and certified for operational use via field deployments and overflights of the array with test aircraft, in this case with sUAS aircraft such as J-FLiC.

Hearing Aids: How to Choose the Right One

MedlinePlus

... and all are powered with a hearing aid battery. Small microphones collect sounds from the environment. A ... to pick up wind noise Uses very small batteries, which have shorter life and can be difficult ...

Animation of MARDI Instrument

NASA Image and Video Library

2008-09-30

This frame from an animation shows a zoom into the Mars Descent Imager MARDI instrument onboard NASA Phoenix Mars Lander. The Phoenix team will soon attempt to use a microphone on the MARDI instrument to capture sounds of Mars.

Mission Impossible? Never. or Beep...Beep...Beep

ERIC Educational Resources Information Center

Grady, William F.

1971-01-01

Described and recommended is a method for preventing members of travelling study groups from missing any of their guides' remarks. Basically the system consists of a wireless microphone for the guide and earplugs for each traveler. (LS)

Fiber-optic interferometric sensors for measurements of pressure fluctuations: Experimental evaluation

NASA Technical Reports Server (NTRS)

Cho, Y. C.; Soderman, P. T.

1993-01-01

This paper addresses an anechoic chamber evaluation of a fiber-optic interferometric sensor (fiber-optic microphone), which is being developed at NASA Ames Research Center for measurements of pressure fluctuations in wind tunnels.

Acoustic imaging of aircraft wake vortex dynamics

DOT National Transportation Integrated Search

2005-06-01

The experience in utilizing a phased microphone array to passively image aircraft wake : vortices is highlighted. It is demonstrated that the array can provide visualization of wake : dynamics similar to smoke release or natural condensation of vorti...

Method for Determining Artillery Position

NASA Technical Reports Server (NTRS)

Fischer, Johannes; Meuser, Wilfried

1988-01-01

A method is described for determinig artillery positions. Two groups of four closely spaced microphones are placed at known positions, and radio bearings are determined by projectile flight time differences of muzzle blasts. The advantages of the method are discussed.

Sensing of Particular Speakers for the Construction of Voice Interface Utilized in Noisy Environment

NASA Astrophysics Data System (ADS)

Sawada, Hideyuki; Ohkado, Minoru

Human is able to exchange information smoothly using voice under different situations such as noisy environment in a crowd and with the existence of plural speakers. We are able to detect the position of a source sound in 3D space, extract a particular sound from mixed sounds, and recognize who is talking. By realizing this mechanism with a computer, new applications will be presented for recording a sound with high quality by reducing noise, presenting a clarified sound, and realizing a microphone-free speech recognition by extracting particular sound. The paper will introduce a realtime detection and identification of particular speaker in noisy environment using a microphone array based on the location of a speaker and the individual voice characteristics. The study will be applied to develop an adaptive auditory system of a mobile robot which collaborates with a factory worker.

Background Noise Reduction Using Adaptive Noise Cancellation Determined by the Cross-Correlation

NASA Technical Reports Server (NTRS)

Spalt, Taylor B.; Brooks, Thomas F.; Fuller, Christopher R.

2012-01-01

Background noise due to flow in wind tunnels contaminates desired data by decreasing the Signal-to-Noise Ratio. The use of Adaptive Noise Cancellation to remove background noise at measurement microphones is compromised when the reference sensor measures both background and desired noise. The technique proposed modifies the classical processing configuration based on the cross-correlation between the reference and primary microphone. Background noise attenuation is achieved using a cross-correlation sample width that encompasses only the background noise and a matched delay for the adaptive processing. A present limitation of the method is that a minimum time delay between the background noise and desired signal must exist in order for the correlated parts of the desired signal to be separated from the background noise in the crosscorrelation. A simulation yields primary signal recovery which can be predicted from the coherence of the background noise between the channels. Results are compared with two existing methods.

Automated Cough Assessment on a Mobile Platform

PubMed Central

2014-01-01

The development of an Automated System for Asthma Monitoring (ADAM) is described. This consists of a consumer electronics mobile platform running a custom application. The application acquires an audio signal from an external user-worn microphone connected to the device analog-to-digital converter (microphone input). This signal is processed to determine the presence or absence of cough sounds. Symptom tallies and raw audio waveforms are recorded and made easily accessible for later review by a healthcare provider. The symptom detection algorithm is based upon standard speech recognition and machine learning paradigms and consists of an audio feature extraction step followed by a Hidden Markov Model based Viterbi decoder that has been trained on a large database of audio examples from a variety of subjects. Multiple Hidden Markov Model topologies and orders are studied. Performance of the recognizer is presented in terms of the sensitivity and the rate of false alarm as determined in a cross-validation test. PMID:25506590

Measurement of the acoustic response of a wind instrument with application to bore reconstruction

NASA Astrophysics Data System (ADS)

van Walstijn, Maarten; Campbell, Murray

2002-11-01

Reconstruction of a bore from measured acoustic response data has been shown to be very useful in studying wind instruments. Such data may be obtained in different ways; directly measuring the frequency-domain response of an acoustic bore has some distinct advantages over directly measuring time-domain data (for example, by pulse reflectometry), but so far has been unsuitable for producing input data for deterministic bore reconstruction algorithms, due to the limited accuracy at high frequencies. In this paper a method is presented for large-bandwidth measurement of the input impedance of a wind instrument using a cylindrical measurement head with multiple wall-mounted microphones. The influence of the number of microphones and the types of calibration impedance on the accuracy will be discussed, and bore reconstructions derived using this technique will be compared with reconstructions obtained using pulse reflectometry. [Work supported by EPSRC.

Apparatus and Methods for Photoacoustic Measurement of Light Absorption of Particulate and Gaseous Species

NASA Technical Reports Server (NTRS)

Brown, William (Inventor); Yu, Zhenhong (Inventor); Kebabian, Paul L. (Inventor); Assif, James (Inventor)

2017-01-01

In one embodiment, a photoacoustic effect measurement instrument for measuring a species (e.g., a species of PM) in a gas employs a pair of differential acoustic cells including a sample cell that receives sample gas including the species, and a reference cell that receives a filtered version of the sample gas from which the species has been substantially removed. An excitation light source provides an amplitude modulated beam to each of the acoustic cells. An array of multiple microphones is mounted to each of the differential acoustic cells, and measures an acoustic wave generated in the respective acoustic cell by absorption of light by sample gas therein to produce a respective signal. The microphones are isolated from sample gas internal to the acoustic cell by a film. A preamplifier determines a differential signal and a controller calculates concentration of the species based on the differential signal.

Evaluation of a Sensor System for Detecting Humans Trapped under Rubble: A Pilot Study

PubMed Central

Kasai, Ritaro; Cosentino, Sarah; Giacomo, Cimarelli; Mochida, Yasuaki; Yamada, Hiroya; Guarnieri, Michele; Takanishi, Atsuo

2018-01-01

Rapid localization of injured survivors by rescue teams to prevent death is a major issue. In this paper, a sensor system for human rescue including three different types of sensors, a CO2 sensor, a thermal camera, and a microphone, is proposed. The performance of this system in detecting living victims under the rubble has been tested in a high-fidelity simulated disaster area. Results show that the CO2 sensor is useful to effectively reduce the possible concerned area, while the thermal camera can confirm the correct position of the victim. Moreover, it is believed that the use of microphones in connection with other sensors would be of great benefit for the detection of casualties. In this work, an algorithm to recognize voices or suspected human noise under rubble has also been developed and tested. PMID:29534055

Hybrid Feedforward-Feedback Noise Control Using Virtual Sensors

NASA Technical Reports Server (NTRS)

Bean, Jacob; Fuller, Chris; Schiller, Noah

2016-01-01

Several approaches to active noise control using virtual sensors are evaluated for eventual use in an active headrest. Specifically, adaptive feedforward, feedback, and hybrid control structures are compared. Each controller incorporates the traditional filtered-x least mean squares algorithm. The feedback controller is arranged in an internal model configuration to draw comparisons with standard feedforward control theory results. Simulation and experimental results are presented that illustrate each controllers ability to minimize the pressure at both physical and virtual microphone locations. The remote microphone technique is used to obtain pressure estimates at the virtual locations. It is shown that a hybrid controller offers performance benefits over the traditional feedforward and feedback controllers. Stability issues associated with feedback and hybrid controllers are also addressed. Experimental results show that 15-20 dB reduction in broadband disturbances can be achieved by minimizing the measured pressure, whereas 10-15 dB reduction is obtained when minimizing the estimated pressure at a virtual location.

A SOUND SOURCE LOCALIZATION TECHNIQUE TO SUPPORT SEARCH AND RESCUE IN LOUD NOISE ENVIRONMENTS

NASA Astrophysics Data System (ADS)

Yoshinaga, Hiroshi; Mizutani, Koichi; Wakatsuki, Naoto

At some sites of earthquakes and other disasters, rescuers search for people buried under rubble by listening for the sounds which they make. Thus developing a technique to localize sound sources amidst loud noise will support such search and rescue operations. In this paper, we discuss an experiment performed to test an array signal processing technique which searches for unperceivable sound in loud noise environments. Two speakers simultaneously played a noise of a generator and a voice decreased by 20 dB (= 1/100 of power) from the generator noise at an outdoor space where cicadas were making noise. The sound signal was received by a horizontally set linear microphone array 1.05 m in length and consisting of 15 microphones. The direction and the distance of the voice were computed and the sound of the voice was extracted and played back as an audible sound by array signal processing.

Plane-wave decomposition by spherical-convolution microphone array

NASA Astrophysics Data System (ADS)

Rafaely, Boaz; Park, Munhum

2004-05-01

Reverberant sound fields are widely studied, as they have a significant influence on the acoustic performance of enclosures in a variety of applications. For example, the intelligibility of speech in lecture rooms, the quality of music in auditoria, the noise level in offices, and the production of 3D sound in living rooms are all affected by the enclosed sound field. These sound fields are typically studied through frequency response measurements or statistical measures such as reverberation time, which do not provide detailed spatial information. The aim of the work presented in this seminar is the detailed analysis of reverberant sound fields. A measurement and analysis system based on acoustic theory and signal processing, designed around a spherical microphone array, is presented. Detailed analysis is achieved by decomposition of the sound field into waves, using spherical Fourier transform and spherical convolution. The presentation will include theoretical review, simulation studies, and initial experimental results.

Antarctic atmospheric infrasound. Final technical report, 1 July 1981-30 September 1984

DOE Office of Scientific and Technical Information (OSTI.GOV)

Wilson, C.R.; McKibben, B.N.

1986-11-01

In order to monitor atmospheric infrasonic waves in the passband from 0.1 to 0.01 Hz a digital infrasonic detection system was installed in Antarctica on the Ross Ice shelf near McMurdo Station on McMurdo Sound. An array of seven infrasonic microphones subtending an area of about 35 sg km was operated in Windless Bight. The analog microphone data were telemetered to McMurdo station where the infrasonic date were digitized and subjected to on-line real-time analysis to detect traveling infrasonic waves with periods from 10 to 100 seconds. During the period of operation of the Antartic infrasonic observatory, hundreds of infrasonicmore » signals were detected in association with many natural sources such as the aurora australis, marine storm sea-air interactions, volcanic eruptions, mountain generated lee-wave effects, large meteors and auroral electrojet supersonic motions.« less

Dynamic tire pressure sensor for measuring ground vibration.

PubMed

Wang, Qi; McDaniel, James Gregory; Wang, Ming L

2012-11-07

This work presents a convenient and non-contact acoustic sensing approach for measuring ground vibration. This approach, which uses an instantaneous dynamic tire pressure sensor (DTPS), possesses the capability to replace the accelerometer or directional microphone currently being used for inspecting pavement conditions. By measuring dynamic pressure changes inside the tire, ground vibration can be amplified and isolated from environmental noise. In this work, verifications of the DTPS concept of sensing inside the tire have been carried out. In addition, comparisons between a DTPS, ground-mounted accelerometer, and directional microphone are made. A data analysis algorithm has been developed and optimized to reconstruct ground acceleration from DTPS data. Numerical and experimental studies of this DTPS reveal a strong potential for measuring ground vibration caused by a moving vehicle. A calibration of transfer function between dynamic tire pressure change and ground acceleration may be needed for different tire system or for more accurate application.