Sample records for ultrasensitive directional microphone
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Practical considerations for a second-order directional hearing aid microphone system
NASA Astrophysics Data System (ADS)
Thompson, Stephen C.
2003-04-01
First-order directional microphone systems for hearing aids have been available for several years. Such a system uses two microphones and has a theoretical maximum free-field directivity index (DI) of 6.0 dB. A second-order microphone system using three microphones could provide a theoretical increase in free-field DI to 9.5 dB. These theoretical maximum DI values assume that the microphones have exactly matched sensitivities at all frequencies of interest. In practice, the individual microphones in the hearing aid always have slightly different sensitivities. For the small microphone separation necessary to fit in a hearing aid, these sensitivity matching errors degrade the directivity from the theoretical values, especially at low frequencies. This paper shows that, for first-order systems the directivity degradation due to sensitivity errors is relatively small. However, for second-order systems with practical microphone sensitivity matching specifications, the directivity degradation below 1 kHz is not tolerable. A hybrid order directive system is proposed that uses first-order processing at low frequencies and second-order directive processing at higher frequencies. This hybrid system is suggested as an alternative that could provide improved directivity index in the frequency regions that are important to speech intelligibility.
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Ruscetta, Melissa N; Palmer, Catherine V; Durrant, John D; Grayhack, Judith; Ryan, Carey
2007-10-01
The chief complaint of individuals with hearing impairment is difficulty hearing in noise, with directional microphones emerging as the most capable remediation. Our purpose was to determine the impact of directional microphones on localization disability and concurrent handicap. Fifty-seven individuals participated unaided and then in groups of 19, using omni-directional microphones, directional-microphones, or toggle-switch equipped amplification. The outcome measure was a localization disabilities and handicaps questionnaire. Comparisons between the unaided group versus the aided groups, and the directional-microphone groups versus the other two aided groups revealed no significant differences. None of the microphone schemes either increased or decreased self-perceived localization disability or handicap. Objective measures of localization ability are warranted and if significance is noted, clinicians should caution patients when moving in their environment. If no significant objective differences exist, in light of the subjective findings in this investigation concern over decreases in quality of life and safety with directional microphones need not be considered.
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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.
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Factors influencing individual variation in perceptual directional microphone benefit.
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.
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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.
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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.
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Chen, Hanchi; Abhayapala, Thushara D; Zhang, Wen
2015-11-01
Soundfield analysis based on spherical harmonic decomposition has been widely used in various applications; however, a drawback is the three-dimensional geometry of the microphone arrays. In this paper, a method to design two-dimensional planar microphone arrays that are capable of capturing three-dimensional (3D) spatial soundfields is proposed. Through the utilization of both omni-directional and first order microphones, the proposed microphone array is capable of measuring soundfield components that are undetectable to conventional planar omni-directional microphone arrays, thus providing the same functionality as 3D arrays designed for the same purpose. Simulations show that the accuracy of the planar microphone array is comparable to traditional spherical microphone arrays. Due to its compact shape, the proposed microphone array greatly increases the feasibility of 3D soundfield analysis techniques in real-world applications.
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Optimization of Microphone Locations for Acoustic Liner Impedance Eduction
NASA Technical Reports Server (NTRS)
Jones, M. G.; Watson, W. R.; June, J. C.
2015-01-01
Two impedance eduction methods are explored for use with data acquired in the NASA Langley Grazing Flow Impedance Tube. The first is an indirect method based on the convected Helmholtz equation, and the second is a direct method based on the Kumaresan and Tufts algorithm. Synthesized no-flow data, with random jitter to represent measurement error, are used to evaluate a number of possible microphone locations. Statistical approaches are used to evaluate the suitability of each set of microphone locations. Given the computational resources required, small sample statistics are employed for the indirect method. Since the direct method is much less computationally intensive, a Monte Carlo approach is employed to gather its statistics. A comparison of results achieved with full and reduced sets of microphone locations is used to determine which sets of microphone locations are acceptable. For the indirect method, each array that includes microphones in all three regions (upstream and downstream hard wall sections, and liner test section) provides acceptable results, even when as few as eight microphones are employed. The best arrays employ microphones well away from the leading and trailing edges of the liner. The direct method is constrained to use microphones opposite the liner. Although a number of arrays are acceptable, the optimum set employs 14 microphones positioned well away from the leading and trailing edges of the liner. The selected sets of microphone locations are also evaluated with data measured for ceramic tubular and perforate-over-honeycomb liners at three flow conditions (Mach 0.0, 0.3, and 0.5). They compare favorably with results attained using all 53 microphone locations. Although different optimum microphone locations are selected for the two impedance eduction methods, there is significant overlap. Thus, the union of these two microphone arrays is preferred, as it supports usage of both methods. This array contains 3 microphones in the upstream hard wall section, 14 microphones opposite the liner, and 3 microphones in the downstream hard wall section.
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Microphone directionality, pre-emphasis filter, and wind noise in cochlear implants.
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.
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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
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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…
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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.
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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.
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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.
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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.
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Bai, Lijuan; Yuan, Ruo; Chai, Yaqin; Yuan, Yali; Wang, Yan; Xie, Shunbi
2012-11-18
For the first time, a glucose oxidase-functionalized bioconjugate was prepared and served as a new trace label through its direct electrochemistry and electrocatalysis in a sandwich-type electrochemical aptasensor for ultrasensitive detection of thrombin.
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The Effect of Microphone Type on Acoustical Measures of Synthesized Vowels.
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.
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Calibration of High Frequency MEMS Microphones
NASA Technical Reports Server (NTRS)
Shams, Qamar A.; Humphreys, William M.; Bartram, Scott M.; Zuckewar, Allan J.
2007-01-01
Understanding and controlling aircraft noise is one of the major research topics of the NASA Fundamental Aeronautics Program. One of the measurement technologies used to acquire noise data is the microphone directional array (DA). Traditional direction array hardware, consisting of commercially available condenser microphones and preamplifiers can be too expensive and their installation in hard-walled wind tunnel test sections too complicated. An emerging micro-machining technology coupled with the latest cutting edge technologies for smaller and faster systems have opened the way for development of MEMS microphones. The MEMS microphone devices are available in the market but suffer from certain important shortcomings. Based on early experiments with array prototypes, it has been found that both the bandwidth and the sound pressure level dynamic range of the microphones should be increased significantly to improve the performance and flexibility of the overall array. Thus, in collaboration with an outside MEMS design vendor, NASA Langley modified commercially available MEMS microphone as shown in Figure 1 to meet the 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. Over the years, several methods have been used for microphone calibration. Some of the common methods of microphone calibration are Coupler (Reciprocity, Substitution, and Simultaneous), Pistonphone, Electrostatic actuator, and Free-field calibration (Reciprocity, Substitution, and Simultaneous). Traditionally, electrostatic actuators (EA) have been used to characterize air-condenser microphones for wideband frequency ranges; however, MEMS microphones are not adaptable to the EA method due to their construction and very small diaphragm size. 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. 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.
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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.
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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.
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Assessment of a directional microphone array for hearing-impaired listeners.
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.
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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.
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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.
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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.
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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.
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The benefits of remote microphone technology for adults with cochlear implants.
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.
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Vibrational Profiling of Brain Tumors and Cells
Nelson, Sultan L; Proctor, Dustin T; Ghasemloonia, Ahmad; Lama, Sanju; Zareinia, Kourosh; Ahn, Younghee; Al-Saiedy, Mustafa R; Green, Francis HY; Amrein, Matthias W; Sutherland, Garnette R
2017-01-01
This study reports vibration profiles of neuronal cells and tissues as well as brain tumor and neocortical specimens. A contact-free method and analysis protocol was designed to convert an atomic force microscope into an ultra-sensitive microphone with capacity to record and listen to live biological samples. A frequency of 3.4 Hz was observed for both cultured rat hippocampal neurons and tissues and vibration could be modulated pharmacologically. Malignant astrocytoma tissue samples obtained from operating room, transported in artificial cerebrospinal fluid, and tested within an hour, vibrated with a much different frequency profile and amplitude, compared to meningioma or lateral temporal cortex providing a quantifiable measurement to accurately distinguish the three tissues in real-time. Vibration signals were converted to audible sound waves by frequency modulation, thus demonstrating, acoustic patterns unique to meningioma, malignant astrocytoma and neocortex. PMID:28744324
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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.
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Design of Small MEMS Microphone Array Systems for Direction Finding of Outdoors Moving Vehicles
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
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Design of small MEMS microphone array systems for direction finding of outdoors moving vehicles.
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.
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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.
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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.
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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.
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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.
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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.
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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.
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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.
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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…
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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.
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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.
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The effect of hearing aid technologies on listening in an automobile.
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.
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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.
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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.
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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.
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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.
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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...
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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.
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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.
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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.
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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.
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The effect of hearing aid technologies on listening in an automobile
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
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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.…
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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.
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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…
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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.
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Spatial acoustic radiation of respiratory sounds for sleep evaluation.
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.
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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.
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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
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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.
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Use of a Parabolic Microphone to Detect Hidden Subjects in Search and Rescue.
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.
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Ultrasensitive Detection of Shigella Species in Blood and Stool.
Luo, Jieling; Wang, Jiapeng; Mathew, Anup S; Yau, Siu-Tung
2016-02-16
A modified immunosensing system with voltage-controlled signal amplification was used to detect Shigella in stool and blood matrixes at the single-digit CFU level. Inactivated Shigella was spiked in these matrixes and detected directly. The detection was completed in 78 min. Detection limits of 21 CFU/mL and 18 CFU/mL were achieved in stool and blood, respectively, corresponding to 2-7 CFUs immobilized on the detecting electrode. The outcome of the detection of extremely low bacterium concentration, i.e., below 100 CFU/mL, blood samples show a random nature. An analysis of the detection probabilities indicates the correlation between the sample volume and the success of detection and suggests that sample volume is critical for ultrasensitive detection of bacteria. The calculated detection limit is qualitatively in agreement with the empirically determined detection limit. The demonstrated ultrasensitive detection of Shigella on the single-digit CFU level suggests the feasibility of the direct detection of the bacterium in the samples without performing a culture.
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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…
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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.
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Broadband implementation of coprime linear microphone arrays for direction of arrival estimation.
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.
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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.
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Signal Processing and Interpretation Using Multilevel Signal Abstractions.
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
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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.
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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.
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Wenga, G; Jacques, E; Salaün, A-C; Rogel, R; Pichon, L; Geneste, F
2013-02-15
Currently, detection of DNA hybridization using fluorescence-based detection technique requires expensive optical systems and complex bioinformatics tools. Hence, the development of new low cost devices that enable direct and highly sensitive detection stimulates a lot of research efforts. Particularly, devices based on silicon nanowires are emerging as ultrasensitive electrical sensors for the direct detection of biological species thanks to their high surface to volume ratio. In this study, we propose innovative devices using step-gate polycrystalline silicon nanowire FET (poly-Si NW FETs), achieved with simple and low cost fabrication process, and used as ultrasensitive electronic sensor for DNA hybridization. The poly-SiNWs are synthesized using the sidewall spacer formation technique. The detailed fabrication procedure for a step-gate NWFET sensor is described in this paper. No-complementary and complementary DNA sequences were clearly discriminated and detection limit to 1 fM range is observed. This first result using this nano-device is promising for the development of low cost and ultrasensitive polysilicon nanowires based DNA sensors compatible with the CMOS technology. Copyright © 2012 Elsevier B.V. All rights reserved.
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Huang, Lin; Wu, Jingjing; Zheng, Lei; Qian, Haisheng; Xue, Feng; Wu, Yucheng; Pan, Daodong; Adeloju, Samuel B; Chen, Wei
2013-11-19
A novel electrochemical aptasensor is described for rapid and ultrasensitive detection of ochratoxin A (OTA) based on signal enhancement with rolling circle amplification (RCA). The primer for RCA was designed to compose of a two-part sequence, one part of the aptamer sequence directed against OTA while the other part was complementary to the capture probe on the electrode surface. In the presence of target OTA, the primer, originally hybridized with the RCA padlock, is replaced to combine with OTA. This induces the inhibition of RCA and decreases the OTA sensing signal obtained with the electrochemical aptasensor. Under the optimized conditions, ultrasensitive detection of OTA was achieved with a limit of detection (LOD) of 0.065 ppt (pg/mL), which is much lower than previously reported. The electrochemical aptasensor was also successfully applied to the determination of OTA in wine samples. This ultrasensitive electrochemical aptasensor is of great practical importance in food safety and could be widely extended to the detection of other toxins by replacing the sequence of the recognition aptamer.
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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.
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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.
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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.
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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.
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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.
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Yan, Yurong; Ding, Shijia; Zhao, Dan; Yuan, Rui; Zhang, Yuhong; Cheng, Wei
2016-01-01
Sensitive and specific methodologies for detection of pathogenic gene at the point-of-care are still urgent demands in rapid diagnosis of infectious diseases. This work develops a simple and pragmatic electrochemical biosensing strategy for ultrasensitive and specific detection of pathogenic nucleic acids directly by integrating homogeneous target-initiated transcription amplification (HTITA) with interfacial sensing process in single analysis system. The homogeneous recognition and specific binding of target DNA with the designed hairpin probe triggered circular primer extension reaction to form DNA double-strands which contained T7 RNA polymerase promoter and served as templates for in vitro transcription amplification. The HTITA protocol resulted in numerous single-stranded RNA products which could synchronously hybridized with the detection probes and immobilized capture probes for enzyme-amplified electrochemical detection on the biosensor surface. The proposed electrochemical biosensing strategy showed very high sensitivity and selectivity for target DNA with a dynamic response range from 1 fM to 100 pM. Using salmonella as a model, the established strategy was successfully applied to directly detect invA gene from genomic DNA extract. This proposed strategy presented a simple, pragmatic platform toward ultrasensitive nucleic acids detection and would become a versatile and powerful tool for point-of-care pathogen identification. PMID:26729209
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NASA Astrophysics Data System (ADS)
Yan, Yurong; Ding, Shijia; Zhao, Dan; Yuan, Rui; Zhang, Yuhong; Cheng, Wei
2016-01-01
Sensitive and specific methodologies for detection of pathogenic gene at the point-of-care are still urgent demands in rapid diagnosis of infectious diseases. This work develops a simple and pragmatic electrochemical biosensing strategy for ultrasensitive and specific detection of pathogenic nucleic acids directly by integrating homogeneous target-initiated transcription amplification (HTITA) with interfacial sensing process in single analysis system. The homogeneous recognition and specific binding of target DNA with the designed hairpin probe triggered circular primer extension reaction to form DNA double-strands which contained T7 RNA polymerase promoter and served as templates for in vitro transcription amplification. The HTITA protocol resulted in numerous single-stranded RNA products which could synchronously hybridized with the detection probes and immobilized capture probes for enzyme-amplified electrochemical detection on the biosensor surface. The proposed electrochemical biosensing strategy showed very high sensitivity and selectivity for target DNA with a dynamic response range from 1 fM to 100 pM. Using salmonella as a model, the established strategy was successfully applied to directly detect invA gene from genomic DNA extract. This proposed strategy presented a simple, pragmatic platform toward ultrasensitive nucleic acids detection and would become a versatile and powerful tool for point-of-care pathogen identification.
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Yan, Yurong; Ding, Shijia; Zhao, Dan; Yuan, Rui; Zhang, Yuhong; Cheng, Wei
2016-01-05
Sensitive and specific methodologies for detection of pathogenic gene at the point-of-care are still urgent demands in rapid diagnosis of infectious diseases. This work develops a simple and pragmatic electrochemical biosensing strategy for ultrasensitive and specific detection of pathogenic nucleic acids directly by integrating homogeneous target-initiated transcription amplification (HTITA) with interfacial sensing process in single analysis system. The homogeneous recognition and specific binding of target DNA with the designed hairpin probe triggered circular primer extension reaction to form DNA double-strands which contained T7 RNA polymerase promoter and served as templates for in vitro transcription amplification. The HTITA protocol resulted in numerous single-stranded RNA products which could synchronously hybridized with the detection probes and immobilized capture probes for enzyme-amplified electrochemical detection on the biosensor surface. The proposed electrochemical biosensing strategy showed very high sensitivity and selectivity for target DNA with a dynamic response range from 1 fM to 100 pM. Using salmonella as a model, the established strategy was successfully applied to directly detect invA gene from genomic DNA extract. This proposed strategy presented a simple, pragmatic platform toward ultrasensitive nucleic acids detection and would become a versatile and powerful tool for point-of-care pathogen identification.
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Hydrogel microphones for stealthy underwater listening
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
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Analysis of the cochlear microphonic to a low-frequency tone embedded in filtered noise
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
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Leak locating microphone, method and system for locating fluid leaks in pipes
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.
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Impact of a Moving Noise Masker on Speech Perception in Cochlear Implant Users
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
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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.
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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.
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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.
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Furthur remarks on atmospheric probing by ultrasensitive radar
NASA Technical Reports Server (NTRS)
Atlas, D.
1969-01-01
This paper is supplementary to that of Hardy and Katz. It emphasizes the meteorological value of the various capabilities of ultrasensitive radar, highlights the points of agreement and disagreement, and focuses upon the directions of promising research. The theory of backscatter from a refractively turbulent region is said to be confirmed by the radar observations both with respect to magnitude and wavelength dependence. A reason for the apparent discrepancy between the results of some of the forwardscatter experiments and theory is suggested. Disagreement still exists with respect to the origin of clear air sea breeze echoes; the author does not agree with Hardy and Katz that they are due to insects. However, it is agreed that some unusually widespread echo displays on clear days are indeed due to insects. The meteorological value of ultrasensitive radars demonstrated by Hardy and Katz, here, and by others is so profound as to demand their use in remote atmospheric probing.
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Cobalt oxide nanosheets wrapped onto nickel foam for non-enzymatic detection of glucose
NASA Astrophysics Data System (ADS)
Meng, Shangjun; Wu, Meiyan; Wang, Qian; Dai, Ziyang; Si, Weili; Huang, Wei; Dong, Xiaochen
2016-08-01
Ultra-sensitive and highly selective detection of glucose is essential for the clinical diagnosis of diabetes. In this paper, an ultra-sensitive glucose sensor was successfully fabricated based on cobalt oxide (Co3O4) nanosheets directly grown on nickel foam through a simple hydrothermal method. Characterizations indicated that the Co3O4 nanosheets are completely and uniformly wrapped onto the surface of nickel foam to form a three-dimensional heterostructure. The resulting self-standing electrochemical electrode presents a high performance for the non-enzymatic detection of glucose, including short response time (<10 s), ultra-sensitivity (12.97 mA mM-1 cm-2), excellent selectivity and low detection limit (0.058 μM, S/N = 3). These results indicate that Co3O4 nanosheets wrapped onto nickel foam are a low-cost, practical, and high performance electrochemical electrode for bio sensing.
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Single and Multiple Microphone Noise Reduction Strategies in Cochlear Implants
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
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Noise Spectra and Directivity For a Scale-Model Landing Gear
NASA Technical Reports Server (NTRS)
Humphreys, William M., Jr.; Brooks, Thomas F.
2007-01-01
An extensive experimental study has been conducted to acquire detailed noise spectra and directivity data for a high-fidelity, 6.3%-scale, Boeing 777 main landing gear. The measurements were conducted in the NASA Langley Quiet Flow Facility using a 41-microphone directional array system positioned at a range of polar and azimuthal observer angles with respect to the model. DAMAS (Deconvolution Approach for the Mapping of Acoustic Sources) array processing as well as straightforward individual microphone processing were employed to compile unique flyover and sideline directivity databases for a range of freestream Mach numbers (0.11 - 0.17) covering typical approach conditions. Comprehensive corrections were applied to the test data to account for shear layer ray path and amplitude variations. This allowed proper beamforming at different measurement orientations, as well as directivity presentation in free-field emission coordinates. Four different configurations of the landing gear were tested: a baseline configuration with and without an attached side door, and a noise reduction concept "toboggan" truck fairing with and without side door. DAMAS noise source distributions were determined. Spectral analyses demonstrated that individual microphones could establish model spectra. This finding permitted the determination of unique, spatially-detailed directivity contours of spectral band levels over a hemispherical surface. Spectral scaling for the baseline model confirmed that the acoustic intensity scaled with the expected sixth-power of the Mach number. Finally, comparison of spectra and directivity between the baseline gear and the gear with an attached toboggan indicated that the toboggan fairing may be of some value in reducing gear noise over particular frequency ranges.
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Real-time distributed fiber microphone based on phase-OTDR.
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.
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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.
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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.
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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.
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Azimuthal sound localization in the European starling (Sturnus vulgaris): I. Physical binaural cues.
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.
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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.
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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.
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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
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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.
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Graphene electrostatic microphone and ultrasonic radio
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
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Altszyler, Edgar; Ventura, Alejandra C; Colman-Lerner, Alejandro; Chernomoretz, Ariel
2017-01-01
Ultrasensitive response motifs, capable of converting graded stimuli into binary responses, are well-conserved in signal transduction networks. Although it has been shown that a cascade arrangement of multiple ultrasensitive modules can enhance the system's ultrasensitivity, how a given combination of layers affects a cascade's ultrasensitivity remains an open question for the general case. Here, we introduce a methodology that allows us to determine the presence of sequestration effects and to quantify the relative contribution of each module to the overall cascade's ultrasensitivity. The proposed analysis framework provides a natural link between global and local ultrasensitivity descriptors and it is particularly well-suited to characterize and understand mathematical models used to study real biological systems. As a case study, we have considered three mathematical models introduced by O'Shaughnessy et al. to study a tunable synthetic MAPK cascade, and we show how our methodology can help modelers better understand alternative models.
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Altszyler, Edgar; Ventura, Alejandra C.; Colman-Lerner, Alejandro; Chernomoretz, Ariel
2017-01-01
Ultrasensitive response motifs, capable of converting graded stimuli into binary responses, are well-conserved in signal transduction networks. Although it has been shown that a cascade arrangement of multiple ultrasensitive modules can enhance the system’s ultrasensitivity, how a given combination of layers affects a cascade’s ultrasensitivity remains an open question for the general case. Here, we introduce a methodology that allows us to determine the presence of sequestration effects and to quantify the relative contribution of each module to the overall cascade’s ultrasensitivity. The proposed analysis framework provides a natural link between global and local ultrasensitivity descriptors and it is particularly well-suited to characterize and understand mathematical models used to study real biological systems. As a case study, we have considered three mathematical models introduced by O’Shaughnessy et al. to study a tunable synthetic MAPK cascade, and we show how our methodology can help modelers better understand alternative models. PMID:28662096
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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.
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Microphones and Educational Media.
ERIC Educational Resources Information Center
Page, Marilyn
This paper describes the types of microphones that are available for use in media production. Definitions of 16 words and phrases used to describe microphones are followed by detailed descriptions of the two kinds of microphones as classified by mode of operation, i.e., velocity, or ribbon microphones, and pressure operated microphones, which…
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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.
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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.
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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.
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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.
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Multimodal physiological sensor for motion artefact rejection.
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.
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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.
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Ultrasensitivity of the Bacillus subtilis sporulation decision.
Narula, Jatin; Devi, Seram N; Fujita, Masaya; Igoshin, Oleg A
2012-12-11
Starving Bacillus subtilis cells execute a gene expression program resulting in the formation of stress-resistant spores. Sporulation master regulator, Spo0A, is activated by a phosphorelay and controls the expression of a multitude of genes, including the forespore-specific sigma factor σ(F) and the mother cell-specific sigma factor σ(E). Identification of the system-level mechanism of the sporulation decision is hindered by a lack of direct control over Spo0A activity. This limitation can be overcome by using a synthetic system in which Spo0A activation is controlled by inducing expression of phosphorelay kinase KinA. This induction results in a switch-like increase in the number of sporulating cells at a threshold of KinA. Using a combination of mathematical modeling and single-cell microscopy, we investigate the origin and physiological significance of this ultrasensitive threshold. The results indicate that the phosphorelay is unable to achieve a sufficiently fast and ultrasensitive response via its positive feedback architecture, suggesting that the sporulation decision is made downstream. In contrast, activation of σ(F) in the forespore and of σ(E) in the mother cell compartments occurs via a cascade of coherent feed-forward loops, and thereby can produce fast and ultrasensitive responses as a result of KinA induction. Unlike σ(F) activation, σ(E) activation in the mother cell compartment only occurs above the KinA threshold, resulting in completion of sporulation. Thus, ultrasensitive σ(E) activation explains the KinA threshold for sporulation induction. We therefore infer that under uncertain conditions, cells initiate sporulation but postpone making the sporulation decision to average stochastic fluctuations and to achieve a robust population response.
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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.
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Effects of venting on wind noise levels measured at the eardrum.
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.
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High sensitivity capacitive MEMS microphone with spring supported diaphragm
NASA Astrophysics Data System (ADS)
Mohamad, Norizan; Iovenitti, Pio; Vinay, Thurai
2007-12-01
Capacitive microphones (condenser microphones) work on a principle of variable capacitance and voltage by the movement of its electrically charged diaphragm and back plate in response to sound pressure. There has been considerable research carried out to increase the sensing performance of microphones while reducing their size to cater for various modern applications such as mobile communication and hearing aid devices. This paper reviews the development and current performance of several condenser MEMS microphone designs, and introduces a microphone with spring supported diaphragm to further improve condenser microphone performance. The numerical analysis using Coventor FEM software shows that this new microphone design has a higher mechanical sensitivity compared to the existing edge clamped flat diaphragm condenser MEMS microphone. The spring supported diaphragm is shown to have a flat frequency response up to 7 kHz and more stable under the variations of the diaphragm residual stress. The microphone is designed to be easily fabricated using the existing silicon fabrication technology and the stability against the residual stress increases its reproducibility.
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Background noise levels measured in the NASA Lewis 9- by 15-foot low-speed wind tunnel
NASA Technical Reports Server (NTRS)
Woodward, Richard P.; Dittmar, James H.; Hall, David G.; Kee-Bowling, Bonnie
1994-01-01
The acoustic capability of the NASA Lewis 9 by 15 Foot Low Speed Wind Tunnel has been significantly improved by reducing the background noise levels measured by in-flow microphones. This was accomplished by incorporating streamlined microphone holders having a profile developed by researchers at the NASA Ames Research Center. These new holders were fabricated for fixed mounting on the tunnel wall and for an axially traversing microphone probe which was mounted to the tunnel floor. Measured in-flow noise levels in the tunnel test section were reduced by about 10 dB with the new microphone holders compared with those measured with the older, less refined microphone holders. Wake interference patterns between fixed wall microphones were measured and resulted in preferred placement patterns for these microphones to minimize these effects. Acoustic data from a model turbofan operating in the tunnel test section showed that results for the fixed and translating microphones were equivalent for common azimuthal angles, suggesting that the translating microphone probe, with its significantly greater angular resolution, is preferred for sideline noise measurements. Fixed microphones can provide a local check on the traversing microphone data quality, and record acoustic performance at other azimuthal angles.
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Speech intelligibility in noise using throat and acoustic microphones.
Acker-Mills, Barbara E; Houtsma, Adrianus J M; Ahroon, William A
2006-01-01
Helicopter cockpits are very noisy and this noise must be reduced for effective communication. The standard U.S. Army aviation helmet is equipped with a noise-canceling acoustic microphone, but some ambient noise still is transmitted. Throat microphones are not sensitive to air molecule vibrations and thus, transmittal of ambient noise is reduced. It is possible that throat microphones could enhance speech communication in helicopters, but speech intelligibility with the devices must first be assessed. In the current study, speech intelligibility of signals generated by an acoustic microphone, a throat microphone, and by the combined output of the two microphones was assessed using the Modified Rhyme Test (MRT). Stimulus words were recorded in a reverberant chamber with ambient broadband noise intensity at 90 and 106 dBA. Listeners completed the MRT task in the same settings, thus simulating the typical environment of a rotary-wing aircraft. Results show that speech intelligibility is significantly worse for the throat microphone (average percent correct = 55.97) than for the acoustic microphone (average percent correct = 69.70), particularly for the higher noise level. In addition, no benefit is gained by simultaneously using both microphones. A follow-up experiment evaluated different consonants using the Diagnostic Rhyme Test and replicated the MRT results. The current results show that intelligibility using throat microphones is poorer than with the use of boom microphones in noisy and in quiet environments. Therefore, throat microphones are not recommended for use in any situation where fast and accurate speech intelligibility is essential.
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Chen, L G; Zhang, C; Zhang, R; Zhang, X L; Dong, Z C
2013-06-01
We report the development of a custom scanning tunneling microscope equipped with photon collection and detection systems. The optical optimization includes the comprehensive design of aspherical lens for light collimation and condensing, the sophisticated piezo stages for in situ lens adjustment inside ultrahigh vacuum, and the fiber-free coupling of collected photons directly onto the ultrasensitive single-photon detectors. We also demonstrate submolecular photon mapping for the molecular islands of porphyrin on Ag(111) under small tunneling currents down to 10 pA and short exposure time down to 1.2 ms/pixel. A high quantum efficiency up to 10(-2) was also observed.
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Ultrasensitive detection and characterization of molecules with infrared plasmonic metamaterials
Cheng, Fei; Yang, Xiaodong; Gao, Jie
2015-01-01
Infrared vibrational spectroscopy is an effective technique which enables the direct probe of molecular fingerprints, and such detection can be further enhanced by the emerging engineered plasmonic metamaterials. Here we experimentally demonstrate ultrasensitive detection and characterization of polymer molecules based on an asymmetric infrared plasmonic metamaterial, and quantitatively analyze the molecule detection sensitivity and molecule-structure interactions. A sharp, non-radiative Fano resonance supported by the plasmonic metamaterial exhibits strongly enhanced near-field, and the resonance frequency is tailored to match the vibrational fingerprint of the target molecule. By utilizing the near-field nature of the plasmonic excitation, significantly enhanced absorption signal of molecules in the infrared spectroscopy are obtained, enabling ultrasensitive detection of only minute quantities of organic molecules. The enhancement of molecular absorption up to 105 fold is obtained, and sensitive detection of molecules at zeptomole levels (corresponding to a few tens of molecules within a unit cell) is achieved with high signal-to-noise ratio in our experiment. The demonstrated infrared plasmonic metamaterial sensing platform offers great potential for improving the specificity and sensitivity of label-free, biochemical detection. PMID:26388404
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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.
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Vehicle Counting and Moving Direction Identification Based on Small-Aperture Microphone Array.
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.
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Neurodynamic evaluation of hearing aid features using EEG correlates of listening effort.
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.
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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.
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A combined microphone and camera calibration technique with application to acoustic imaging.
Legg, Mathew; Bradley, Stuart
2013-10-01
We present a calibration technique for an acoustic imaging microphone array, combined with a digital camera. Computer vision and acoustic time of arrival data are used to obtain microphone coordinates in the camera reference frame. Our new method allows acoustic maps to be plotted onto the camera images without the need for additional camera alignment or calibration. Microphones and cameras may be placed in an ad-hoc arrangement and, after calibration, the coordinates of the microphones are known in the reference frame of a camera in the array. No prior knowledge of microphone positions, inter-microphone spacings, or air temperature is required. This technique is applied to a spherical microphone array and a mean difference of 3 mm was obtained between the coordinates obtained with this calibration technique and those measured using a precision mechanical method.
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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.
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Theoretical and experimental study of a fiber optic microphone
NASA Technical Reports Server (NTRS)
Hu, Andong; Cuomo, Frank W.; Zuckerwar, Allan J.
1992-01-01
Modifications to condenser microphone theory yield new expressions for the membrane deflections at its center, which provide the basic theory for the fiber optic microphone. The theoretical analysis for the membrane amplitude and the phase response of the fiber optic microphone is given in detail in terms of its basic geometrical quantities. A relevant extension to the original concepts of the optical microphone includes the addition of a backplate with holes similar in design to present condenser microphone technology. This approach generates improved damping characteristics and extended frequency response that were not previously considered. The construction and testing of the improved optical fiber microphone provide experimental data that are in good agreement with the theoretical analysis.
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Towards a sub 15-dBA optical micromachined microphone
Kim, Donghwan; Hall, Neal A.
2014-01-01
Micromachined microphones with grating-based optical-interferometric readout have been demonstrated previously. These microphones are similar in construction to bottom-inlet capacitive microelectromechanical-system (MEMS) microphones, with the exception that optoelectronic emitters and detectors are placed inside the microphone's front or back cavity. A potential advantage of optical microphones in designing for low noise level is the use of highly-perforated microphone backplates to enable low-damping and low thermal-mechanical noise levels. This work presents an experimental study of a microphone diaphragm and backplate designed for optical readout and low thermal-mechanical noise. The backplate is 1 mm × 1 mm and is fabricated in a 2-μm-thick epitaxial silicon layer of a silicon-on-insulator wafer and contains a diffraction grating with 4-μm pitch etched at the center. The presented system has a measured thermal-mechanical noise level equal to 22.6 dBA. Through measurement of the electrostatic frequency response and measured noise spectra, a device model for the microphone system is verified. The model is in-turn used to identify design paths towards MEMS microphones with sub 15-dBA noise floors. PMID:24815250
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Benefits of the fiber optic versus the electret microphone in voice amplification.
Kyriakou, Kyriaki; Fisher, Hélène R
2013-01-01
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 with amplification systems is the electret microphone. One alternate form of microphone is the fiber optic microphone. To examine the benefits of the fiber optic (1190S) versus the electret (M04) microphone as measured by objective and subjective parameters in the amplification of a patient's voice with reduced loudness caused by neurological and/or respiratory-based problems. Eighteen patients with vocal fold paralysis, Parkinson's disease and/or chronic obstructive pulmonary disease (COPD) participated in the study. The study contained a measurement of intensity, amplitude perturbation and signal-to-noise ratio during a sustained vowel production and a measurement of intensity during conversation with the use of the two microphones simultaneously. It also included the completion of a questionnaire indicating the patient's satisfaction with each microphone. The fiber optic (1190S) microphone had better objective acoustic performance (i.e. lower amplitude perturbation, higher signal-to-noise ratio and higher intensity) than the electret (M04) microphone. It also had better patient subjective satisfaction (i.e. less conspicuousness, more voice clarity, less acoustic feedback, more loudness and more utilization) than the electret microphone. Patients with neurological and/or respiratory-based voice problems may more confidently and frequently use the fiber optic microphone to communicate, socialize and participate in occupational activities more easily. Speech-language pathologists may more confidently use or recommend the fiber optic microphone with amplification systems. © 2012 Royal College of Speech and Language Therapists.
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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.
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Truck acoustic data analyzer system
Haynes, Howard D.; Akerman, Alfred; Ayers, Curtis W.
2006-07-04
A passive vehicle acoustic data analyzer system having at least one microphone disposed in the acoustic field of a moving vehicle and a computer in electronic communication the microphone(s). The computer detects and measures the frequency shift in the acoustic signature emitted by the vehicle as it approaches and passes the microphone(s). The acoustic signature of a truck driving by a microphone can provide enough information to estimate the truck speed in miles-per-hour (mph), engine speed in rotations-per-minute (RPM), turbocharger speed in RPM, and vehicle weight.
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NASA Astrophysics Data System (ADS)
Dittberner, Andrew; Bentler, Ruth
2005-09-01
The relationship between various directivity measures and subject performance with directional microphone hearing aids was determined. Test devices included first- and second-order directional microphones. Recordings of sentences and noise (Hearing in Noise Test, HINT) were made through each test device in simple, complex, and anisotropic background noise conditions. Twenty-six subjects, with normal hearing, were administered the HINT test recordings, and directional benefit was computed. These measures were correlated to theoretical, free-field, and KEMAR DI values, as well as front-to-back ratios, in situ SNRs, and a newly proposed Db-SNR, wherein a predictive value of the SNR improvement is calculated as a function of the noise source incidence. The different predictive scores showed high correlation to the measured directional benefit scores in the complex (diffuse-like) background noise condition (r=0.89-0.97, p<0.05) but not across all background noise conditions (r=0.45-0.97, p<0.05). The Db-SNR approach and the in situ SNR measures provided excellent prediction of subject performance in all background noise conditions (0.85-0.97, p<0.05) None of the predictive measures could account for the effects of reverberation on the speech signal (r=0.35-0.40, p<0.05).
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NASA Technical Reports Server (NTRS)
Radcliffe, Eliott (Inventor); Naguib, Ahmed (Inventor); Humphreys, Jr., William M. (Inventor)
2014-01-01
A feedback-controlled microphone includes a microphone body and a membrane operatively connected to the body. The membrane is configured to be initially deflected by acoustic pressure such that the initial deflection is characterized by a frequency response. The microphone also includes a sensor configured to detect the frequency response of the initial deflection and generate an output voltage indicative thereof. The microphone additionally includes a compensator in electric communication with the sensor and configured to establish a regulated voltage in response to the output voltage. Furthermore, the microphone includes an actuator in electric communication with the compensator, wherein the actuator is configured to secondarily deflect the membrane in opposition to the initial deflection such that the frequency response is adjusted. An acoustic beam forming microphone array including a plurality of the above feedback-controlled microphones is also disclosed.
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On the ability of consumer electronics microphones for environmental noise monitoring.
Van Renterghem, Timothy; Thomas, Pieter; Dominguez, Frederico; Dauwe, Samuel; Touhafi, Abdellah; Dhoedt, Bart; Botteldooren, Dick
2011-03-01
The massive production of microphones for consumer electronics, and the shift from dedicated processing hardware to PC-based systems, opens the way to build affordable, extensive noise measurement networks. Applications include e.g. noise limit and urban soundscape monitoring, and validation of calculated noise maps. Microphones are the critical components of such a network. Therefore, in a first step, some basic characteristics of 8 microphones, distributed over a wide range of price classes, were measured in a standardized way in an anechoic chamber. In a next step, a thorough evaluation was made of the ability of these microphones to be used for environmental noise monitoring. This was done during a continuous, half-year lasting outdoor experiment, characterized by a wide variety of meteorological conditions. While some microphones failed during the course of this test, it was shown that it is possible to identify cheap microphones that highly correlate to the reference microphone during the full test period. When the deviations are expressed in total A-weighted (road traffic) noise levels, values of less than 1 dBA are obtained, in excess to the deviation amongst reference microphones themselves.
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Effectiveness of the Directional Microphone in the Baha® Divino™
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
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Effectiveness of the directional microphone in the Baha® Divino™.
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.
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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.
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Dynamic tire pressure sensor for measuring ground vibration.
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.
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Dynamic Tire Pressure Sensor for Measuring Ground Vibration
Wang, Qi; McDaniel, James Gregory; Wang, Ming L.
2012-01-01
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. PMID:23202206
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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.
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Probe Microphone Measurements: 20 Years of Progress
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
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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 microphone location program is presented. The Microphone Location Program determines microphone locations that ensure accurate and stable results from the equation system used to calculate modal structures. As part of the computational procedure for the Microphone Location Program, a first-order measure of the stability of the equation system was indicated by a matrix 'conditioning' number.
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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.
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Wagner, Randall P.; Guthrie, William F.
2015-01-01
The devices calibrated most frequently by the acoustical measurement services at the National Institute of Standards and Technology (NIST) over the 50-year period from 1963 to 20121 were one-inch condenser microphones of three specific standard types: LS1Pn, LS1Po, and WS1P. Due to its long history of providing calibrations of such microphones to customers, NIST is in a unique position to analyze data concerning the long-term stability of these devices. This long history has enabled NIST to acquire and aggregate a substantial amount of repeat calibration data for a large number of microphones that belong to various other standards and calibration laboratories. In addition to determining microphone sensitivities at the time of calibration, it is important to have confidence that the microphones do not typically undergo significant drift as compared to the calibration uncertainty during the periods between calibrations. For each of the three microphone types, an average drift rate and approximate 95 % confidence interval were computed by two different statistical methods, and the results from the two methods were found to differ insignificantly in each case. These results apply to typical microphones of these types that are used in a suitable environment and handled with care. The average drift rate for Type LS1Pn microphones was −0.004 dB/year to 0.003 dB/year. The average drift rate for Type LS1Po microphones was −0.016 dB/year to 0.008 dB/year. The average drift rate for Type WS1P microphones was −0.004 dB/year to 0.018 dB/year. For each of these microphone types, the average drift rate is not significantly different from zero. This result is consistent with the performance expected of condenser microphones designed for use as transfer standards. In addition, the values that bound the confidence intervals are well within the limits specified for long-term stability in international standards. Even though these results show very good long-term stability historically for these microphone types, it is expected that periodic calibrations will always be done to track the calibration history of individual microphones and check for anomalies indicative of shifts in sensitivity. PMID:26958445
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Acoustic characterization of wake vortices in ground effect
DOT National Transportation Integrated Search
2005-01-01
The experience and findings of an exploratory effort to characterize the sound emitted by : aircraft wake vortices near the ground are presented. A line array of four directional : microphones was deployed and recorded the wakes of several commercial...
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Speech Intelligibility in Various Noise Conditions with the Nucleus® 5 CP810 Sound Processor.
Dillier, Norbert; Lai, Wai Kong
2015-06-11
The Nucleus(®) 5 System Sound Processor (CP810, Cochlear™, Macquarie University, NSW, Australia) contains two omnidirectional microphones. They can be configured as a fixed directional microphone combination (called Zoom) or as an adaptive beamformer (called Beam), which adjusts the directivity continuously to maximally reduce the interfering noise. Initial evaluation studies with the CP810 had compared performance and usability of the new processor in comparison with the Freedom™ Sound Processor (Cochlear™) for speech in quiet and noise for a subset of the processing options. This study compares the two processing options suggested to be used in noisy environments, Zoom and Beam, for various sound field conditions using a standardized speech in noise matrix test (Oldenburg sentences test). Nine German-speaking subjects who previously had been using the Freedom speech processor and subsequently were upgraded to the CP810 device participated in this series of additional evaluation tests. The speech reception threshold (SRT for 50% speech intelligibility in noise) was determined using sentences presented via loudspeaker at 65 dB SPL in front of the listener and noise presented either via the same loudspeaker (S0N0) or at 90 degrees at either the ear with the sound processor (S0NCI+) or the opposite unaided ear (S0NCI-). The fourth noise condition consisted of three uncorrelated noise sources placed at 90, 180 and 270 degrees. The noise level was adjusted through an adaptive procedure to yield a signal to noise ratio where 50% of the words in the sentences were correctly understood. In spatially separated speech and noise conditions both Zoom and Beam could improve the SRT significantly. For single noise sources, either ipsilateral or contralateral to the cochlear implant sound processor, average improvements with Beam of 12.9 and 7.9 dB in SRT were found. The average SRT of -8 dB for Beam in the diffuse noise condition (uncorrelated noise from both sides and back) is truly remarkable and comparable to the performance of normal hearing listeners in the same test environment. The static directivity (Zoom) option in the diffuse noise condition still provides a significant benefit of 5.9 dB in comparison with the standard omnidirectional microphone setting. These results indicate that CI recipients may improve their speech recognition in noisy environments significantly using these directional microphone-processing options.
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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.
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Ultrasensitivity and sharp threshold theorems for multisite systems
NASA Astrophysics Data System (ADS)
Dougoud, M.; Mazza, C.; Vinckenbosch, L.
2017-02-01
This work studies the ultrasensitivity of multisite binding processes where ligand molecules can bind to several binding sites. It considers more particularly recent models involving complex chemical reactions in allosteric phosphorylation processes and for transcription factors and nucleosomes competing for binding on DNA. New statistics-based formulas for the Hill coefficient and the effective Hill coefficient are provided and necessary conditions for a system to be ultrasensitive are exhibited. It is first shown that the ultrasensitivity of binding processes can be approached using sharp-threshold theorems which have been developed in applied probability theory and statistical mechanics for studying sharp threshold phenomena in reliability theory, random graph theory and percolation theory. Special classes of binding process are then introduced and are described as density dependent birth and death process. New precise large deviation results for the steady state distribution of the process are obtained, which permits to show that switch-like ultrasensitive responses are strongly related to the multi-modality of the steady state distribution. Ultrasensitivity occurs if and only if the entropy of the dynamical system has more than one global minimum for some critical ligand concentration. In this case, the Hill coefficient is proportional to the number of binding sites, and the system is highly ultrasensitive. The classical effective Hill coefficient I is extended to a new cooperativity index I q , for which we recommend the computation of a broad range of values of q instead of just the standard one I = I 0.9 corresponding to the 10%-90% variation in the dose-response. It is shown that this single choice can sometimes mislead the conclusion by not detecting ultrasensitivity. This new approach allows a better understanding of multisite ultrasensitive systems and provides new tools for the design of such systems.
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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.
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Acoustic centering of sources measured by surrounding spherical microphone arrays.
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
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Response Identification in the Extremely Low Frequency Region of an Electret Condenser Microphone
Jeng, Yih-Nen; Yang, Tzung-Ming; Lee, Shang-Yin
2011-01-01
This study shows that a small electret condenser microphone connected to a notebook or a personal computer (PC) has a prominent response in the extremely low frequency region in a specific environment. It confines most acoustic waves within a tiny air cell as follows. The air cell is constructed by drilling a small hole in a digital versatile disk (DVD) plate. A small speaker and an electret condenser microphone are attached to the two sides of the hole. Thus, the acoustic energy emitted by the speaker and reaching the microphone is strong enough to actuate the diaphragm of the latter. The experiments showed that, once small air leakages are allowed on the margin of the speaker, the microphone captured the signal in the range of 0.5 to 20 Hz. Moreover, by removing the plastic cover of the microphone and attaching the microphone head to the vibration surface, the low frequency signal can be effectively captured too. Two examples are included to show the convenience of applying the microphone to pick up the low frequency vibration information of practical systems. PMID:22346594
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Response identification in the extremely low frequency region of an electret condenser microphone.
Jeng, Yih-Nen; Yang, Tzung-Ming; Lee, Shang-Yin
2011-01-01
This study shows that a small electret condenser microphone connected to a notebook or a personal computer (PC) has a prominent response in the extremely low frequency region in a specific environment. It confines most acoustic waves within a tiny air cell as follows. The air cell is constructed by drilling a small hole in a digital versatile disk (DVD) plate. A small speaker and an electret condenser microphone are attached to the two sides of the hole. Thus, the acoustic energy emitted by the speaker and reaching the microphone is strong enough to actuate the diaphragm of the latter. The experiments showed that, once small air leakages are allowed on the margin of the speaker, the microphone captured the signal in the range of 0.5 to 20 Hz. Moreover, by removing the plastic cover of the microphone and attaching the microphone head to the vibration surface, the low frequency signal can be effectively captured too. Two examples are included to show the convenience of applying the microphone to pick up the low frequency vibration information of practical systems.
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Demonstrating Sound Impulses in Pipes.
ERIC Educational Resources Information Center
Raymer, M. G.; Micklavzina, Stan
1995-01-01
Describes a simple, direct method to demonstrate the effects of the boundary conditions on sound impulse reflections in pipes. A graphical display of the results can be made using a pipe, cork, small hammer, microphone, and fast recording electronics. Explains the principles involved. (LZ)
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[An implantable microphone for electronic hearing aids].
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).
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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.
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Goldsworthy, Raymond L.; Delhorne, Lorraine A.; Desloge, Joseph G.; Braida, Louis D.
2014-01-01
This article introduces and provides an assessment of a spatial-filtering algorithm based on two closely-spaced (∼1 cm) microphones in a behind-the-ear shell. The evaluated spatial-filtering algorithm used fast (∼10 ms) temporal-spectral analysis to determine the location of incoming sounds and to enhance sounds arriving from straight ahead of the listener. Speech reception thresholds (SRTs) were measured for eight cochlear implant (CI) users using consonant and vowel materials under three processing conditions: An omni-directional response, a dipole-directional response, and the spatial-filtering algorithm. The background noise condition used three simultaneous time-reversed speech signals as interferers located at 90°, 180°, and 270°. Results indicated that the spatial-filtering algorithm can provide speech reception benefits of 5.8 to 10.7 dB SRT compared to an omni-directional response in a reverberant room with multiple noise sources. Given the observed SRT benefits, coupled with an efficient design, the proposed algorithm is promising as a CI noise-reduction solution. PMID:25096120
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In-flight measurement of propeller noise on the fuselage of an airplane
NASA Technical Reports Server (NTRS)
Pla, Frederic G.; Ranaudo, Richard; Woodward, Richard P.
1989-01-01
In-flight measurements of propeller noise on the fuselage of an OV-10A aircraft were obtained using a horizontal and a vertical microphone array. A wide range of flight conditions were tested including changes in angle of attack, sideslip angle, power coefficient, helical tip Mach number and advance ratio, and propeller direction of rotation. Results show a dependence of the level and directivity of the tones on the angle of attack and on the sideslip angle with the propeller direction of rotation, which is similar to results obtained in wind tunnel tests with advanced propeller designs. The level of the tones at each microphone increases with increasing angle of attack for inboard-down propeller rotation and decreases for inboard-up rotation. The level also increases with increasing slideslip angle for both propeller directions of rotation. Increasing the power coefficient results in a slight increase in the level of the tones. A strong shock wave is generated by the propeller blades even at relatively low helical tip Mach numbers resulting in high harmonic levels. As the helical tip Mach number and the advance ratio are increased, the level of the higher harmonics increases much faster than the level of the blade passage frequency.
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Prey pursuit strategy of Japanese horseshoe bats during an in-flight target-selection task.
Kinoshita, Yuki; Ogata, Daiki; Watanabe, Yoshiaki; Riquimaroux, Hiroshi; Ohta, Tetsuo; Hiryu, Shizuko
2014-09-01
The prey pursuit behavior of Japanese horseshoe bats (Rhinolophus ferrumequinum nippon) was investigated by tasking bats during flight with choosing between two tethered fluttering moths. Echolocation pulses were recorded using a telemetry microphone mounted on the bat combined with a 17-channel horizontal microphone array to measure pulse directions. Flight paths of the bat and moths were monitored using two high-speed video cameras. Acoustical measurements of returning echoes from fluttering moths were first collected using an ultrasonic loudspeaker, turning the head direction of the moth relative to the loudspeaker from 0° (front) to 180° (back) in the horizontal plane. The amount of acoustical glints caused by moth fluttering varied with the sound direction, reaching a maximum at 70°-100° in the horizontal plane. In the flight experiment, moths chosen by the bat fluttered within or moved across these angles relative to the bat's pulse direction, which would cause maximum dynamic changes in the frequency and amplitude of acoustical glints during flight. These results suggest that echoes with acoustical glints containing the strongest frequency and amplitude modulations appear to attract bats for prey selection.
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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
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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)
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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.
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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.
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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.
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Method for Determining Artillery Position
NASA Technical Reports Server (NTRS)
Meuser, Wilfried
1988-01-01
A method is described for determining the position of artillery in which a circle of four closely spaced microphones is located at two measurement sites for acoustic radio direction finding of muzzle blasts. A method for determining the position of artillery using this procedure is discussed.
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The impact of the microphone position on the frequency analysis of snoring sounds.
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.
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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...
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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...
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Near-Field Sound Localization Based on the Small Profile Monaural Structure
Kim, Youngwoong; Kim, Keonwook
2015-01-01
The acoustic wave around a sound source in the near-field area presents unconventional properties in the temporal, spectral, and spatial domains due to the propagation mechanism. This paper investigates a near-field sound localizer in a small profile structure with a single microphone. The asymmetric structure around the microphone provides a distinctive spectral variation that can be recognized by the dedicated algorithm for directional localization. The physical structure consists of ten pipes of different lengths in a vertical fashion and rectangular wings positioned between the pipes in radial directions. The sound from an individual direction travels through the nearest open pipe, which generates the particular fundamental frequency according to the acoustic resonance. The Cepstral parameter is modified to evaluate the fundamental frequency. Once the system estimates the fundamental frequency of the received signal, the length of arrival and angle of arrival (AoA) are derived by the designed model. From an azimuthal distance of 3–15 cm from the outer body of the pipes, the extensive acoustic experiments with a 3D-printed structure show that the direct and side directions deliver average hit rates of 89% and 73%, respectively. The closer positions to the system demonstrate higher accuracy, and the overall hit rate performance is 78% up to 15 cm away from the structure body. PMID:26580618
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NASA Astrophysics Data System (ADS)
Singh, Manpreet; Alabanza, Anginelle; Gonzalez, Lorelis E.; Wang, Weiwei; Reeves, W. Brian; Hahm, Jong-In
2016-02-01
Determining ultratrace amounts of protein biomarkers in patient samples in a straightforward and quantitative manner is extremely important for early disease diagnosis and treatment. Here, we successfully demonstrate the novel use of zinc oxide nanorods (ZnO NRs) in the ultrasensitive and quantitative detection of two acute kidney injury (AKI)-related protein biomarkers, tumor necrosis factor (TNF)-α and interleukin (IL)-8, directly from patient samples. We first validate the ZnO NRs-based IL-8 results via comparison with those obtained from using a conventional enzyme-linked immunosorbent method in samples from 38 individuals. We further assess the full detection capability of the ZnO NRs-based technique by quantifying TNF-α, whose levels in human urine are often below the detection limits of conventional methods. Using the ZnO NR platforms, we determine the TNF-α concentrations of all 46 patient samples tested, down to the fg per mL level. Subsequently, we screen for TNF-α levels in approximately 50 additional samples collected from different patient groups in order to demonstrate a potential use of the ZnO NRs-based assay in assessing cytokine levels useful for further clinical monitoring. Our research efforts demonstrate that ZnO NRs can be straightforwardly employed in the rapid, ultrasensitive, quantitative, and simultaneous detection of multiple AKI-related biomarkers directly in patient urine samples, providing an unparalleled detection capability beyond those of conventional analysis methods. Additional key advantages of the ZnO NRs-based approach include a fast detection speed, low-volume assay condition, multiplexing ability, and easy automation/integration capability to existing fluorescence instrumentation. Therefore, we anticipate that our ZnO NRs-based detection method will be highly beneficial for overcoming the frequent challenges in early biomarker development and treatment assessment, pertaining to the facile and ultrasensitive quantification of hard-to-trace biomolecules.Determining ultratrace amounts of protein biomarkers in patient samples in a straightforward and quantitative manner is extremely important for early disease diagnosis and treatment. Here, we successfully demonstrate the novel use of zinc oxide nanorods (ZnO NRs) in the ultrasensitive and quantitative detection of two acute kidney injury (AKI)-related protein biomarkers, tumor necrosis factor (TNF)-α and interleukin (IL)-8, directly from patient samples. We first validate the ZnO NRs-based IL-8 results via comparison with those obtained from using a conventional enzyme-linked immunosorbent method in samples from 38 individuals. We further assess the full detection capability of the ZnO NRs-based technique by quantifying TNF-α, whose levels in human urine are often below the detection limits of conventional methods. Using the ZnO NR platforms, we determine the TNF-α concentrations of all 46 patient samples tested, down to the fg per mL level. Subsequently, we screen for TNF-α levels in approximately 50 additional samples collected from different patient groups in order to demonstrate a potential use of the ZnO NRs-based assay in assessing cytokine levels useful for further clinical monitoring. Our research efforts demonstrate that ZnO NRs can be straightforwardly employed in the rapid, ultrasensitive, quantitative, and simultaneous detection of multiple AKI-related biomarkers directly in patient urine samples, providing an unparalleled detection capability beyond those of conventional analysis methods. Additional key advantages of the ZnO NRs-based approach include a fast detection speed, low-volume assay condition, multiplexing ability, and easy automation/integration capability to existing fluorescence instrumentation. Therefore, we anticipate that our ZnO NRs-based detection method will be highly beneficial for overcoming the frequent challenges in early biomarker development and treatment assessment, pertaining to the facile and ultrasensitive quantification of hard-to-trace biomolecules. Electronic supplementary information (ESI) available: Typical SEM images of the ZnO NRs used in the biomarker assays are provided in Fig. S1. See DOI: 10.1039/c5nr08706f
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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.
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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.
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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.
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In situ Probe Microphone Measurement for Testing the Direct Acoustical Cochlear Stimulator.
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.
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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.
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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
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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.
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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…
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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.
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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.
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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.
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14 CFR 27.1457 - Cockpit voice recorders.
Code of Federal Regulations, 2010 CFR
2010-01-01
... stations and voice communications of other crewmembers on the flight deck when directed to those stations... pilot stations. The microphone specified in this paragraph must be so located and, if necessary, the... are intelligible when recorded under flight cockpit noise conditions and played back. The level of...
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14 CFR 25.1457 - Cockpit voice recorders.
Code of Federal Regulations, 2010 CFR
2010-01-01
... stations and voice communications of other crewmembers on the flight deck when directed to those stations... as practicable when recorded under flight cockpit noise conditions and played back. Repeated aural or... pilot station. (2) For the second channel from each boom, mask, or hand-held microphone, headset, or...
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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.
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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
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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.
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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.
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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.
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Acoustic Characterization of a Multi-Rotor Unmanned Aircraft
NASA Astrophysics Data System (ADS)
Feight, Jordan; Gaeta, Richard; Jacob, Jamey
2017-11-01
In this study, the noise produced by a small multi-rotor rotary wing aircraft, or drone, is measured and characterized. The aircraft is tested in different configurations and environments to investigate specific parameters and how they affect the acoustic signature of the system. The parameters include rotor RPM, the number of rotors, distance and angle of microphone array from the noise source, and the ambient environment. The testing environments include an anechoic chamber for an idealized setting and both indoor and outdoor settings to represent real world conditions. PIV measurements are conducted to link the downwash and vortical flow structures from the rotors with the noise generation. The significant factors that arise from this study are the operational state of the aircraft and the microphone location (or the directivity of the noise source). The directivity in the rotor plane was shown to be omni-directional, regardless of the varying parameters. The tonal noise dominates the low to mid frequencies while the broadband noise dominates the higher frequencies. The fundamental characteristics of the acoustic signature appear to be invariant to the number of rotors. Flight maneuvers of the aircraft also significantly impact the tonal content in the acoustic signature.
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Visualizing Sound Directivity via Smartphone Sensors
ERIC Educational Resources Information Center
Hawley, Scott H.; McClain, Robert E., Jr.
2018-01-01
When Yang-Hann Kim received the Rossing Prize in Acoustics Education at the 2015 meeting of the Acoustical Society of America, he stressed the importance of offering visual depictions of sound fields when teaching acoustics. Often visualization methods require specialized equipment such as microphone arrays or scanning apparatus. We present a…
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14 CFR 23.1457 - Cockpit voice recorders.
Code of Federal Regulations, 2010 CFR
2010-01-01
... originating at the first and second pilot stations and voice communications of other crewmembers on the flight deck when directed to those stations. The microphone must be so located and, if necessary, the... conditions and played back. Repeated aural or visual playback of the record may be used in evaluating...
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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.
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Directionality of dog vocalizations
NASA Astrophysics Data System (ADS)
Frommolt, Karl-Heinz; Gebler, Alban
2004-07-01
The directionality patterns of sound emission in domestic dogs were measured in an anechoic environment using a microphone array. Mainly long-distance signals from four dogs were investigated. The radiation pattern of the signals differed clearly from an omnidirectional one with average differences in sound-pressure level between the frontal and rear position of 3-7 dB depending from the individual. Frequency dependence of directionality was shown for the range from 250 to 3200 Hz. The results indicate that when studying acoustic communication in mammals, more attention should be paid to the directionality pattern of sound emission.
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Directivity and noise reduction in hearing aids: speech perception and benefit.
Quintino, Camila Angélica; Mondelli, Maria Fernanda Capoani Garcia; Ferrari, Déborah Viviane
2010-01-01
Hearing aid. To compare the performance, benefit and satisfaction of users of ITE, CIC and BTE digital hearing aid with noise reduction and omnidirectional and directional microphones. 34 users of hearing aid were evaluated by means of speech perception in noise tests and APHAB and IOI self assessment questionnaires. Prospective study. Better results were obtained by users of ITE, CIC and directional hearing aids, however, no statistical significance was found between the groups. Directivity improved speech perception in noise and benefit in daily life situations.
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Multi-criteria anomaly detection in urban noise sensor networks.
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.
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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.
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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.
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Background noise in piezoresistive, electret condenser, and ceramic microphones.
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.
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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.
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[Value of the study of cochlear microphonic recordings in deep and severe deafness].
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.
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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.
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Huang, Liaojing; Zhang, Li; Yang, Liu; Yuan, Ruo; Yuan, Yali
2018-05-01
In this work, the manganese porphyrin (MnPP) decorated on DNA networks could serve as quencher and mimicking enzyme to efficiently reduce the photocurrent of photoactive material 3,4,9,10-perylene tetracarboxylic acid (PTCA), which was elaborately used to construct a novel label-free aptasensor for ultrasensitive detection of thrombin (TB) in a signal-off manner. The Au-doped PTCA (PTCA-PEI-Au) with outstanding membrane-forming and photoelectric property was modified on electrode to acquire a strong initial photoelectrochemistry (PEC) signal. Afterward, target binding aptamer Ι (TBAΙ) was modified on electrode to specially recognize target TB, which could further combine with TBAII and single-stranded DNA P1-modified platinum nanoparticles (TBAII-PtNPs-P1) for immobilizing DNA networks with abundant MnPP. Ingeniously, the MnPP could not only directly quench the photocurrent of PTCA, but also acted as hydrogen peroxide (HRP) mimicking enzyme to remarkably stimulate the deposition of benzo-4-chlorhexidine (4-CD) on electrode for further decreasing the photocurrent of PTCA, thereby obtaining a definitely low photocurrent for detection of TB. As a result, the proposed PEC aptasensor illustrated excellent sensitivity with a low detection limit down to 3 fM, exploiting a new avenue about intergrating two functions in one substance for ultrasensitive biological monitoring. Copyright © 2017 Elsevier B.V. All rights reserved.
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Zhu, Ye; Wang, Huijuan; Wang, Lin; Zhu, Jing; Jiang, Wei
2016-02-03
An ultrasensitive and highly selective electrochemical assay was first attempted by combining the rolling circle amplification (RCA) reaction with poly(thymine)-templated copper nanoparticles (CuNPs) for cascade signal amplification. As proof of concept, prostate specific antigen (PSA) was selected as a model target. Using a gold nanoparticle (AuNP) as a carrier, we synthesized the primer-AuNP-aptamer bioconjugate for signal amplification by increasing the primer/aptamer ratio. The specific construction of primer-AuNP-aptamer/PSA/anti-PSA sandwich structure triggered the effective RCA reaction, in which thousands of tandem poly(thymine) repeats were generated and directly served as the specific templates for the subsequent CuNP formation. The signal readout was easily achieved by dissolving the RCA product-templated CuNPs and detecting the released copper ions with differential pulse stripping voltammetry. Because of the designed cascade signal amplification strategy, the newly developed method achieved a linear range of 0.05-500 fg/mL, with a remarkable detection limit of 0.020 ± 0.001 fg/mL PSA. Finally, the feasibility of the developed method for practical application was investigated by analyzing PSA in the real clinical human serum samples. The ultrasensitivity, specificity, convenience, and capability for analyzing the clinical samples demonstrate that this method has great potential for practical disease diagnosis applications.
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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…
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Noise Attenuation Performance Assessment of the Joint Helmet Mounted Cueing System (JHMCS)
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
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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.
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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.
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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.
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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.
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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.
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A three-microphone acoustic reflection technique using transmitted acoustic waves in the airway.
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.
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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.
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A High-Resolution Stopwatch for Cents
ERIC Educational Resources Information Center
Gingl, Z.; Kopasz, K.
2011-01-01
A very low-cost, easy-to-make stopwatch is presented to support various experiments in mechanics. The high-resolution stopwatch is based on two photodetectors connected directly to the microphone input of a sound card. Dedicated free open-source software has been developed and made available to download. The efficiency is demonstrated by a free…
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Measuring Effects Of Lightning On Power And Telephone Lines
NASA Technical Reports Server (NTRS)
Jafferis, William; Thompson, E. M.; Medelius, P.; Rubinstein, M.; Tzeng, A.
1992-01-01
Spherical antenna senses both horizontal and vertical fields simultaneously. Measures "fast" components of electric field used in conjunction with other equipment, including antenna measuring "slow" vertical component of electric field; microphone that senses thunder; cameras making visual records, which locate lightning; magnetic-field sensor; optical sensors; and instruments measuring speed and direction of wind.
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Smartphones Offer New Opportunities in Clinical Voice Research.
Manfredi, C; Lebacq, J; Cantarella, G; Schoentgen, J; Orlandi, S; Bandini, A; DeJonckere, P H
2017-01-01
Smartphone technology provides new opportunities for recording standardized voice samples of patients and sending the files by e-mail to the voice laboratory. This drastically improves the collection of baseline data, as used in research on efficiency of voice treatments. However, the basic requirement is the suitability of smartphones for recording and digitizing pathologic voices (mainly characterized by period perturbations and noise) without significant distortion. In this experiment, two smartphones (a very inexpensive one and a high-level one) were tested and compared with direct microphone recordings in a soundproof room. The voice stimuli consisted in synthesized deviant voice samples (median of fundamental frequency: 120 and 200 Hz) with three levels of jitter and three levels of added noise. All voice samples were analyzed using PRAAT software. The results show high correlations between jitter, shimmer, and noise-to-harmonics ratio measured on the recordings via both smartphones, the microphone, and measured directly on the sound files from the synthesizer. Smartphones thus appear adequate for reliable recording and digitizing of pathologic voices. Copyright © 2017 The Voice Foundation. Published by Elsevier Inc. All rights reserved.
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Wu, Yongmei; Xu, Wenju; Bai, Lijuan; Yuan, Yali; Yi, Huayu; Chai, Yaqin; Yuan, Ruo
2013-12-15
For the first time, a sandwich-type electrochemical method was proposed for ultrasensitive thrombin (TB) detection based on direct electrochemistry of highly loaded hemoglobin spheres-encapsulated platinum nanoparticles (PtNPs@Hb) as labels and electrocatalysts. The prepared PtNPs@Hb not only exhibited good biocompatibility, excellent electrocatalytic activity, but also presented redox activity of Hb. Thus, it was employed for the fabrication of aptasensor without any extraneous redox mediators, leading to a simple preparation process for the aptasensor. The high loading of Hb spheres as redox mediators could enhance the electrochemical signal. Importantly, the synergetic electrocatalytic behavior of Hb and PtNPs toward H2O2 reduction greatly amplified the electrochemical signal, resulting in the high sensitivity of aptasensor. Consequently, under optimal conditions, the designed aptasensor exhibited a lower detection limit of 0.05 pM and wide dynamic linear range from 0.15 pM to 40 nM for TB detection. Additionally, the proposed mediator-free and signal-amplified electrochemical aptasensor showed great potential in portable and cost-effective TB sensing devices. Copyright © 2013 Elsevier B.V. All rights reserved.
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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.
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The Benefit of Remote Microphones Using Four Wireless Protocols.
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.
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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.
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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.
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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.
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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...
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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.
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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.
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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.
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Ion implantation system and process for ultrasensitive determination of target isotopes
DOE Office of Scientific and Technical Information (OSTI.GOV)
Farmer, III, Orville T.; Liezers, Martin
2016-09-13
A system and process are disclosed for ultrasensitive determination of target isotopes of analytical interest in a sample. Target isotopes may be implanted in an implant area on a high-purity substrate to pre-concentrate the target isotopes free of contaminants. A known quantity of a tracer isotope may also be implanted. Target isotopes and tracer isotopes may be determined in a mass spectrometer. The present invention provides ultrasensitive determination of target isotopes in the sample.
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Detection of HIV-1 p24 at Attomole Level by Ultrasensitive ELISA with Thio-NAD Cycling
Nakatsuma, Akira; Kaneda, Mugiho; Kodama, Hiromi; Morikawa, Mika; Watabe, Satoshi; Nakaishi, Kazunari; Yamashita, Masakane; Yoshimura, Teruki; Miura, Toshiaki; Ninomiya, Masaki; Ito, Etsuro
2015-01-01
To reduce the window period between HIV-1 infection and the ability to diagnose it, a fourth-generation immunoassay including the detection of HIV-1 p24 antigen has been developed. However, because the commercially available systems for this assay use special, high-cost instruments to measure, for example, chemiluminescence, it is performed only by diagnostics companies and hub hospitals. To overcome this limitation, we applied an ultrasensitive ELISA coupled with a thio-NAD cycling, which is based on a usual enzyme immunoassay without special instruments, to detect HIV-1 p24. The p24 detection limit by our ultrasensitive ELISA was 0.0065 IU/assay (i.e., ca. 10-18 moles/assay). Because HIV-1 p24 antigen is thought to be present in the virion in much greater numbers than viral RNA copies, the value of 10-18 moles of the p24/assay corresponds to ca. 103 copies of the HIV-1 RNA/assay. That is, our ultrasensitive ELISA is chasing the detection limit (102 copies/assay) obtained by PCR-based nucleic acid testing (NAT) with a margin of only one different order. Further, the detection limit by our ultrasensitive ELISA is less than that mandated for a CE-marked HIV antigen/antibody assay. An additional recovery test using blood supported the reliability of our ultrasensitive ELISA. PMID:26098695
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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.
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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
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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.
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NASA Technical Reports Server (NTRS)
Miles, Jeffrey Hilton
2011-01-01
A previous investigation on the presence of direct and indirect combustion noise for a full-scale turbofan engine using a far-field microphone at 130 is extended by also examining signals obtained at two additional downstream directions using far-field microphones at 110 deg and 160 deg. A generalized cross-correlation function technique is used to study the change in propagation time to the far field of the combined direct and indirect combustion noise signal as a sequence of low-pass filters are applied. The filtering procedure used produces no phase distortion. As the low-pass filter frequency is decreased, the travel time increases because the relative amount of direct combustion noise is reduced. The indirect combustion noise signal travels more slowly because in the combustor entropy fluctuations move with the flow velocity, which is slow compared to the local speed of sound. The indirect combustion noise signal travels at acoustic velocities after reaching the turbine and being converted into an acoustic signal. The direct combustion noise is always propagating at acoustic velocities. The results show that the estimated indirect combustion noise time delay values (post-combustion residence times) measured at each angle are fairly consistent with one another for a relevant range of operating conditions and demonstrate source separation of a mixture of direct and indirect combustion noise. The results may lead to a better idea about the acoustics in the combustor and may help develop and validate improved reduced-order physics-based methods for predicting turbofan engine core noise.
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Yan, Hong; Zhong, Mengjuan; Lv, Ze; Wan, Pengbo
2017-11-01
A stretchable, transparent, and body-attachable chemical sensor is assembled from the stretchable nanocomposite network film for ultrasensitive chemical vapor sensing. The stretchable nanocomposite network film is fabricated by in situ preparation of polyaniline/MoS 2 (PANI/MoS 2 ) nanocomposite in MoS 2 suspension and simultaneously nanocomposite deposition onto prestrain elastomeric polydimethylsiloxane substrate. The assembled stretchable electronic sensor demonstrates ultrasensitive sensing performance as low as 50 ppb, robust sensing stability, and reliable stretchability for high-performance chemical vapor sensing. The ultrasensitive sensing performance of the stretchable electronic sensors could be ascribed to the synergistic sensing advantages of MoS 2 and PANI, higher specific surface area, the reliable sensing channels of interconnected network, and the effectively exposed sensing materials. It is expected to hold great promise for assembling various flexible stretchable chemical vapor sensors with ultrasensitive sensing performance, superior sensing stability, reliable stretchability, and robust portability to be potentially integrated into wearable electronics for real-time monitoring of environment safety and human healthcare. © 2017 WILEY-VCH Verlag GmbH & Co. KGaA, Weinheim.
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Improving the accuracy of smart devices to measure noise exposure.
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.
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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.
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NASA Astrophysics Data System (ADS)
Park, Nam In; Kim, Seon Man; Kim, Hong Kook; Kim, Ji Woon; Kim, Myeong Bo; Yun, Su Won
In this paper, we propose a video-zoom driven audio-zoom algorithm in order to provide audio zooming effects in accordance with the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone system, in conjunction with a soft masking process that considers the phase differences between microphones. Thus, the audio-zoom processed signal is obtained by multiplying an audio gain derived from a video-zoom level by the masked signal. After all, a real-time audio-zoom system is implemented on an ARM-CORETEX-A8 having a clock speed of 600 MHz after different levels of optimization are performed such as algorithmic level, C-code, and memory optimizations. To evaluate the complexity of the proposed real-time audio-zoom system, test data whose length is 21.3 seconds long is sampled at 48 kHz. As a result, it is shown from the experiments that the processing time for the proposed audio-zoom system occupies 14.6% or less of the ARM clock cycles. It is also shown from the experimental results performed in a semi-anechoic chamber that the signal with the front direction can be amplified by approximately 10 dB compared to the other directions.
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Acoustic investigation of wall jet over a backward-facing step using a microphone phased array
NASA Astrophysics Data System (ADS)
Perschke, Raimund F.; Ramachandran, Rakesh C.; Raman, Ganesh
2015-02-01
The acoustic properties of a wall jet over a hard-walled backward-facing step of aspect ratios 6, 3, 2, and 1.5 are studied using a 24-channel microphone phased array at Mach numbers up to M=0.6. The Reynolds number based on inflow velocity and step height assumes values from Reh = 3.0 ×104 to 7.2 ×105. Flow without and with side walls is considered. The experimental setup is open in the wall-normal direction and the expansion ratio is effectively 1. In case of flow through a duct, symmetry of the flow in the spanwise direction is lost downstream of separation at all but the largest aspect ratio as revealed by oil paint flow visualization. Hydrodynamic scattering of turbulence from the trailing edge of the step contributes significantly to the radiated sound. Reflection of acoustic waves from the bottom plate results in a modulation of power spectral densities. Acoustic source localization has been conducted using a 24-channel microphone phased array. Convective mean-flow effects on the apparent source origin have been assessed by placing a loudspeaker underneath a perforated flat plate and evaluating the displacement of the beamforming peak with inflow Mach number. Two source mechanisms are found near the step. One is due to interaction of the turbulent wall jet with the convex edge of the step. Free-stream turbulence sound is found to be peaked downstream of the step. Presence of the side walls increases free-stream sound. Results of the flow visualization are correlated with acoustic source maps. Trailing-edge sound and free-stream turbulence sound can be discriminated using source localization.
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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...
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Novel Methods for Sensing Acoustical Emissions From the Knee for Wearable Joint Health Assessment.
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.
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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.
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NASA Technical Reports Server (NTRS)
Johnston, G. D.; Coleman, A. D.; Portwood, J. N.; Saunders, J. M.; Porter, A. J.
1985-01-01
Load-cell and acoustic responses indicate bonding condition nondestructively. Signal recorded by load cell direct and instantaneous measure of local stiffness of material at point of impact. Separate and distinctly different measurement that sensed by microphone. Spectrum analysis of pulse obtained from debonded point will only show frequencies below 425 Hz because insulation alone does not have stiffness to support energy at higher frequencies.
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Design, Fabrication, and Characterization of a Microelectromechanical Directional Microphone
2011-06-01
7. PERFORMING ORGANIZATION NAME(S) AND ADDRESS(ES) 8. PERFORMING ORGANIZATION REPORT NUMBER 9. SPONSORING/MONITORING AGENCY NAME(S) AND ADDRESS(ES...Figure 5.2 SOIC packaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38 Figure 5.3 Laboratory setup...Mean Squared SOC System-On-Chip SOIC Small Outline Integrated Circuit SOIMUMPS Silicon-On-Insulator Multi-User MEMS Process SPL Sound Pressure Level
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Modeling the Performance of MEMS Based Directional Microphones
2008-12-01
5 B. KARUNASIRI’S BIOMIMICRY WORK ................................................ 8... biomimicry efforts involving the fly’s ear. To show the motivation behind the design of an acoustics MEMS device, it includes a brief description of the...system (From: Miles et al., 1995) B. KARUNASIRI’S BIOMIMICRY WORK Two NPS thesis students working under the mentorship of Professor Gamani Karunasiri
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47 CFR 15.717 - TVBDs that rely on spectrum sensing.
Code of Federal Regulations, 2013 CFR
2013-10-01
... over a 100 kHz bandwidth; (C) Low power auxiliary, including wireless microphone, signals: -107 dBm, averaged over a 200 kHz bandwidth. (ii) The detection thresholds are referenced to an omnidirectional receive antenna with a gain of 0 dBi. If a receive antenna with a minimum directional gain of less than 0...
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47 CFR 15.717 - TVBDs that rely on spectrum sensing.
Code of Federal Regulations, 2014 CFR
2014-10-01
... over a 100 kHz bandwidth; (C) Low power auxiliary, including wireless microphone, signals: -107 dBm, averaged over a 200 kHz bandwidth. (ii) The detection thresholds are referenced to an omnidirectional receive antenna with a gain of 0 dBi. If a receive antenna with a minimum directional gain of less than 0...
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47 CFR 15.717 - TVBDs that rely on spectrum sensing.
Code of Federal Regulations, 2011 CFR
2011-10-01
... over a 100 kHz bandwidth; (C) Low power auxiliary, including wireless microphone, signals: -107 dBm, averaged over a 200 kHz bandwidth. (ii) The detection thresholds are referenced to an omnidirectional receive antenna with a gain of 0 dBi. If a receive antenna with a minimum directional gain of less than 0...
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47 CFR 15.717 - TVBDs that rely on spectrum sensing.
Code of Federal Regulations, 2012 CFR
2012-10-01
... over a 100 kHz bandwidth; (C) Low power auxiliary, including wireless microphone, signals: -107 dBm, averaged over a 200 kHz bandwidth. (ii) The detection thresholds are referenced to an omnidirectional receive antenna with a gain of 0 dBi. If a receive antenna with a minimum directional gain of less than 0...
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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.
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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.
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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.
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In vivo evaluation of mastication noise reduction for dual channel implantable microphone.
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.
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Gopalan, Anantha Iyengar; Lee, Kwang Pill; Komathi, Shanmugasundaram
2011-02-15
The present work demonstrates the utility of the functionalized carbon nanotubes, poly(4-aminobenzene sulfonic acid) (PABS) grafted multiwalled carbon nanotubes, MWNT-g-PABS, as an electrode modifier towards achieving ultrasensitive detection of a model drug, sildenafil citrate (SC). PABS units in MWNT-g-PABS interact with SC, pre-concentrate and accumulate at the surface. The electron transduction from SC to electrode is augmented via MWNT-g-PABS. As a result, the MWNT-g-PABS modified electrode exhibited ultrasensitive (57.7 μA/nM) and selective detection of SC with a detection limit of 4.7 pM. The present work provides scope towards targeting ultrasensitivity for the detection of biomolecules/drug through rational design and incorporation of appropriate chemical components to carbon nanotubes. Copyright © 2010 Elsevier B.V. All rights reserved.
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Lapitan Jr., Lorico D. S.; Guo, Yuan
2015-01-01
Single nucleotide polymorphisms (SNPs) constitute the most common types of genetic variations in the human genome. A number of SNPs have been linked to the development of life threatening diseases including cancer, cardiovascular diseases and neurodegenerative diseases. The ability for ultrasensitive and accurate detection of low abundant disease-related SNPs in bodily fluids (e.g. blood, serum, etc.) holds a significant value in the development of non-invasive future biodiagnostic tools. Over the past two decades, nanomaterials have been utilized in a myriad of biosensing applications due to their ability of detecting extremely low quantities of biologically important biomarkers with high sensitivity and accuracy. Of particular interest is the application of such technologies in the detection of SNPs. The use of various nanomaterials, coupled with different powerful signal amplification strategies, has paved the way for a new generation of ultrasensitive SNP biodiagnostic assays. Over the past few years, several ultrasensitive SNP biosensors capable of detecting specific targets down to the ultra-low regimes (ca. aM and below) and therefore holding great promises for early clinical diagnosis of diseases have been developed. This mini review will highlight some of the most recent, significant advances in nanomaterial-based ultrasensitive SNP sensing technologies capable of detecting specific targets on the attomolar (10–18 M) regime or below. In particular, the design of novel, powerful signal amplification strategies that hold the key to the ultrasensitivity is highlighted. PMID:25785914
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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...
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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.
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Sound-field measurement with moving microphones
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
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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.
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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.
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NASA Technical Reports Server (NTRS)
Rentz, P. E.
1976-01-01
Acoustical characteristics and source directionality measurement capabilities of the wind tunnel in the softwall configuration were evaluated, using aerodynamically clean microphone supports. The radius of measurement was limited by the size of the test section, instead of the 3.0 foot (1 m) limitation of the hardwall test section. The wind-on noise level in the test section was reduced 10 dB. Reflections from the microphone support boom, after absorptive covering, induced measurement errors in the lower frequency bands. Reflections from the diffuser back wall were shown to be significant. Tunnel noise coming up the diffuser was postulated as being responsible, at least partially, for the wind-on noise in the test section and settling chamber. The near field characteristics of finite-sized sources and the theoretical response of a porous strip sensor in the presence of wind are presented.
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Comparison Test of the Heavy Expanded Mobility Tactical Truck (HEMTT), M978 Tanker
1989-02-01
transmission shall include the following (MIL-T-PD-977, para 3.4.5.1): (1) A downshift inhibitor system that prevents driver shift control action from...DIRECTION HI 8 Peg. WIND SPEED: 9 KNOTS TAPE RECORDER: B&K 7006 IOCTAVE ANALYZER I BtK ?ni SOUND LEVEL METER: MICROPHONE: BUK ItISS...ISKY COVER: I CLEAR TAPE RECORDER: B&K 7006 IOCTAVE ANALYZER: I BtK 2131 WIND DIRECTION I ?18 Dee . SOUND LEVEL METER STATIONARY
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Sharma, Mukesh Kumar; Narayanan, J; Pardasani, Deepak; Srivastava, Divesh N; Upadhyay, Sanjay; Goel, Ajay Kumar
2016-06-15
Bacillus anthracis, the causative agent of anthrax, is a well known bioterrorism agent. The determination of surface array protein (Sap), a unique biomarker for B. anthracis can offer an opportunity for specific detection of B. anthracis in culture broth. In this study, we designed a new catalytic bionanolabel and fabricated a novel electrochemical immunosensor for ultrasensitive detection of B. anthracis Sap antigen. Bimetallic gold-palladium nanoparticles were in-situ grown on poly (diallyldimethylammonium chloride) functionalized boron nitride nanosheets (Au-Pd NPs@BNNSs) and conjugated with the mouse anti-B. anthracis Sap antibodies (Ab2); named Au-Pd NPs@BNNSs/Ab2. The resulting Au-Pd NPs@BNNSs/Ab2 bionanolabel demonstrated high catalytic activity towards reduction of 4-nitrophenol. The sensitivity of the electrochemical immunosensor along with redox cycling of 4-aminophenol to 4-quinoneimine was improved to a great extent. Under optimal conditions, the proposed immunosensor exhibited a wide working range from 5 pg/mL to 100 ng/mL with a minimum detection limit of 1 pg/mL B. anthracis Sap antigen. The practical applicability of the immunosensor was demonstrated by specific detection of Sap secreted by the B. anthracis in culture broth just after 1h of growth. These labels open a new direction for the ultrasensitive detection of different biological warfare agents and their markers in different matrices. Copyright © 2016 Elsevier B.V. All rights reserved.
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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].
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Aptamer-phage reporters for ultrasensitive lateral flow assays
Adhikari, Meena; Strych, Ulrich; Kim, Jinsu; Goux, Heather; Dhamane, Sagar; Poongavanam, Mohan-Vivekanandan; Hagström, Anna E. V.; Kourentzi, Katerina; Conrad, Jacinta C.; Willson, Richard C.
2015-01-01
We introduce the modification of bacteriophage particles with aptamers for the use as bioanalytical reporters, and demonstrate the use of these particles in ultrasensitive lateral flow assays. M13 phage displaying an in vivo biotinylatable peptide (AviTag) genetically fused to the phage tail protein pIII were used as reporter particle scaffolds, with biotinylated aptamers attached via avidin-biotin linkages, and horseradish peroxidase (HRP) reporter enzymes covalently attached to the pVIII coat protein. These modified viral nanoparticles were used in immunochromatographic sandwich assays for the direct detection of IgE and of the penicillin-binding protein from Staphylococcus aureus (PBP2a). We also developed an additional lateral flow assay for IgE, in which the analyte is sandwiched between immobilized anti-IgE antibodies and aptamer-bearing reporter phage modified with HRP. The limit of detection of this LFA was 0.13 ng/mL IgE, ~100 times lower than those of previously reported IgE assays. PMID:26456715
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Aptamer-Phage Reporters for Ultrasensitive Lateral Flow Assays.
Adhikari, Meena; Strych, Ulrich; Kim, Jinsu; Goux, Heather; Dhamane, Sagar; Poongavanam, Mohan-Vivekanandan; Hagström, Anna E V; Kourentzi, Katerina; Conrad, Jacinta C; Willson, Richard C
2015-12-01
We introduce the modification of bacteriophage particles with aptamers for use as bioanalytical reporters, and demonstrate the use of these particles in ultrasensitive lateral flow assays. M13 phage displaying an in vivo biotinylatable peptide (AviTag) genetically fused to the phage tail protein pIII were used as reporter particle scaffolds, with biotinylated aptamers attached via avidin-biotin linkages, and horseradish peroxidase (HRP) reporter enzymes covalently attached to the pVIII coat protein. These modified viral nanoparticles were used in immunochromatographic sandwich assays for the direct detection of IgE and of the penicillin-binding protein from Staphylococcus aureus (PBP2a). We also developed an additional lateral flow assay for IgE, in which the analyte is sandwiched between immobilized anti-IgE antibodies and aptamer-bearing reporter phage modified with HRP. The limit of detection of this LFA was 0.13 ng/mL IgE, ∼100 times lower than those of previously reported IgE assays.
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NASA Astrophysics Data System (ADS)
Wen, Yanli; Pei, Hao; Shen, Ye; Xi, Junjie; Lin, Meihua; Lu, Na; Shen, Xizhong; Li, Jiong; Fan, Chunhai
2012-11-01
MicroRNAs (miRNAs) have been identified as promising cancer biomarkers due to their stable presence in serum. As an alternative to PCR-based homogenous assays, surface-based electrochemical biosensors offer great opportunities for low-cost, point-of-care tests (POCTs) of disease-associated miRNAs. Nevertheless, the sensitivity of miRNA sensors is often limited by mass transport and crowding effects at the water-electrode interface. To address such challenges, we herein report a DNA nanostructure-based interfacial engineering approach to enhance binding recognition at the gold electrode surface and drastically improve the detection sensitivity. By employing this novel strategy, we can directly detect as few as attomolar (<1, 000 copies) miRNAs with high single-base discrimination ability. Given that this ultrasensitive electrochemical miRNA sensor (EMRS) is highly reproducible and essentially free of prior target labeling and PCR amplification, we also demonstrate its application by analyzing miRNA expression levels in clinical samples from esophageal squamous cell carcinoma (ESCC) patients.
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NASA Astrophysics Data System (ADS)
Csizmok, Veronika; Orlicky, Stephen; Cheng, Jing; Song, Jianhui; Bah, Alaji; Delgoshaie, Neda; Lin, Hong; Mittag, Tanja; Sicheri, Frank; Chan, Hue Sun; Tyers, Mike; Forman-Kay, Julie D.
2017-01-01
The ubiquitin ligase SCFCdc4 mediates phosphorylation-dependent elimination of numerous substrates by binding one or more Cdc4 phosphodegrons (CPDs). Methyl-based NMR analysis of the Cdc4 WD40 domain demonstrates that Cyclin E, Sic1 and Ash1 degrons have variable effects on the primary Cdc4WD40 binding pocket. Unexpectedly, a Sic1-derived multi-CPD substrate (pSic1) perturbs methyls around a previously documented allosteric binding site for the chemical inhibitor SCF-I2. NMR cross-saturation experiments confirm direct contact between pSic1 and the allosteric pocket. Phosphopeptide affinity measurements reveal negative allosteric communication between the primary CPD and allosteric pockets. Mathematical modelling indicates that the allosteric pocket may enhance ultrasensitivity by tethering pSic1 to Cdc4. These results suggest negative allosteric interaction between two distinct binding pockets on the Cdc4WD40 domain may facilitate dynamic exchange of multiple CPD sites to confer ultrasensitive dependence on substrate phosphorylation.
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Ultrasensitive detection enabled by nonlinear magnetization of nanomagnetic labels
Nikitin, M. P.; Orlov, A. V.; Sokolov, I. L.; ...
2018-01-01
The magnetically soft, disk-shaped particles reveal a strong nonlinearity of the magnetization process due to irreversible transitions from the spin vortex to single-domain configuration, enabling their ultrasensitive detection in high-background environments.
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Ultra-sensitive transducer advances micro-measurement range
NASA Technical Reports Server (NTRS)
Rogallo, V. L.
1964-01-01
An ultrasensitive piezoelectric transducer, that converts minute mechanical forces into electrical impulses, measures the impact of micrometeoroids against space vehicles. It has uniform sensitivity over the entire target area and a high degree of stability.
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Seibert, Anna-Maria; Koblitz, Jens C.; Denzinger, Annette; Schnitzler, Hans-Ulrich
2015-01-01
The Barbastelle bat (Barbastella barbastellus) preys almost exclusively on tympanate moths. While foraging, this species alternates between two different signal types. We investigated whether these signals differ in emission direction or source level (SL) as assumed from earlier single microphone recordings. We used two different settings of a 16-microphone array to determine SL and sonar beam direction at various locations in the field. Both types of search signals had low SLs (81 and 82 dB SPL rms re 1 m) as compared to other aerial-hawking bats. These two signal types were emitted in different directions; type 1 signals were directed downward and type 2 signals upward. The angle between beam directions was approximately 70°. Barbastelle bats are able to emit signals through both the mouth and the nostrils. As mouth and nostrils are roughly perpendicular to each other, we conclude that type 1 signals are emitted through the mouth while type 2 signals and approach signals are emitted through the nose. We hypothesize that the “stealth” echolocation system of B. barbastellus is bifunctional. The more upward directed nose signals may be mainly used for search and localization of prey. Their low SL prevents an early detection by eared moths but comes at the expense of a strongly reduced detection range for the environment below the bat. The more downward directed mouth signals may have evolved to compensate for this disadvantage and may be mainly used for spatial orientation. We suggest that the possibly bifunctional echolocation system of B. barbastellus has been adapted to the selective foraging of eared moths and is an excellent example of a sophisticated sensory arms race between predator and prey. PMID:26352271
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Seibert, Anna-Maria; Koblitz, Jens C; Denzinger, Annette; Schnitzler, Hans-Ulrich
2015-01-01
The Barbastelle bat (Barbastella barbastellus) preys almost exclusively on tympanate moths. While foraging, this species alternates between two different signal types. We investigated whether these signals differ in emission direction or source level (SL) as assumed from earlier single microphone recordings. We used two different settings of a 16-microphone array to determine SL and sonar beam direction at various locations in the field. Both types of search signals had low SLs (81 and 82 dB SPL rms re 1 m) as compared to other aerial-hawking bats. These two signal types were emitted in different directions; type 1 signals were directed downward and type 2 signals upward. The angle between beam directions was approximately 70°. Barbastelle bats are able to emit signals through both the mouth and the nostrils. As mouth and nostrils are roughly perpendicular to each other, we conclude that type 1 signals are emitted through the mouth while type 2 signals and approach signals are emitted through the nose. We hypothesize that the "stealth" echolocation system of B. barbastellus is bifunctional. The more upward directed nose signals may be mainly used for search and localization of prey. Their low SL prevents an early detection by eared moths but comes at the expense of a strongly reduced detection range for the environment below the bat. The more downward directed mouth signals may have evolved to compensate for this disadvantage and may be mainly used for spatial orientation. We suggest that the possibly bifunctional echolocation system of B. barbastellus has been adapted to the selective foraging of eared moths and is an excellent example of a sophisticated sensory arms race between predator and prey.
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Ultra-low-noise preamplifier for condenser microphones.
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.
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Cost-Effective, Ultra-Sensitive Groundwater Monitoring for Site Remediation and Management
2015-08-01
Micrometer ml Milliliter MS Mass Spectrometry MW Molecular Weight MΩ Mega-ohm NAS Naval Air Station 6 NASNI Naval Air Station North Island...feasibility studies. ..........42 Table 5-2 Compounds screened in the laboratory for IS2 sampling ......................................44 Table 5-3 Mass ...concentration data is derived directly from the mass of analyte recovered from the sorbent cartridge and the known volume of water processed. This
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Pisanic, Thomas R.; Athamanolap, Pornpat; Poh, Weijie; Chen, Chen; Hulbert, Alicia; Brock, Malcolm V.; Herman, James G.; Wang, Tza-Huei
2015-01-01
Many cancers comprise heterogeneous populations of cells at primary and metastatic sites throughout the body. The presence or emergence of distinct subclones with drug-resistant genetic and epigenetic phenotypes within these populations can greatly complicate therapeutic intervention. Liquid biopsies of peripheral blood from cancer patients have been suggested as an ideal means of sampling intratumor genetic and epigenetic heterogeneity for diagnostics, monitoring and therapeutic guidance. However, current molecular diagnostic and sequencing methods are not well suited to the routine assessment of epigenetic heterogeneity in difficult samples such as liquid biopsies that contain intrinsically low fractional concentrations of circulating tumor DNA (ctDNA) and rare epigenetic subclonal populations. Here we report an alternative approach, deemed DREAMing (Discrimination of Rare EpiAlleles by Melt), which uses semi-limiting dilution and precise melt curve analysis to distinguish and enumerate individual copies of epiallelic species at single-CpG-site resolution in fractions as low as 0.005%, providing facile and inexpensive ultrasensitive assessment of locus-specific epigenetic heterogeneity directly from liquid biopsies. The technique is demonstrated here for the evaluation of epigenetic heterogeneity at p14ARF and BRCA1 gene-promoter loci in liquid biopsies obtained from patients in association with non-small cell lung cancer (NSCLC) and myelodysplastic/myeloproliferative neoplasms (MDS/MPN), respectively. PMID:26304549
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Method for Determination of the Wind Velocity and Direction
NASA Technical Reports Server (NTRS)
Dahlin, Goesta Johan
1988-01-01
Accurate determination of the position of an artillery piece, for example, using sound measurement systems through measurement of the muzzle noise requires access to wind data that is representative of the portion of the air from where the sound wave is propagated up the microphone base of the system. The invention provides a system for determining such representative wind data.
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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.
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NASA Astrophysics Data System (ADS)
Nishiura, Takanobu; Nakamura, Satoshi
2002-11-01
It is very important to capture distant-talking speech for a hands-free speech interface with high quality. A microphone array is an ideal candidate for this purpose. However, this approach requires localizing the target talker. Conventional talker localization algorithms in multiple sound source environments not only have difficulty localizing the multiple sound sources accurately, but also have difficulty localizing the target talker among known multiple sound source positions. To cope with these problems, we propose a new talker localization algorithm consisting of two algorithms. One is DOA (direction of arrival) estimation algorithm for multiple sound source localization based on CSP (cross-power spectrum phase) coefficient addition method. The other is statistical sound source identification algorithm based on GMM (Gaussian mixture model) for localizing the target talker position among localized multiple sound sources. In this paper, we particularly focus on the talker localization performance based on the combination of these two algorithms with a microphone array. We conducted evaluation experiments in real noisy reverberant environments. As a result, we confirmed that multiple sound signals can be identified accurately between ''speech'' or ''non-speech'' by the proposed algorithm. [Work supported by ATR, and MEXT of Japan.
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Two-dimensional grid-free compressive beamforming.
Yang, Yang; Chu, Zhigang; Xu, Zhongming; Ping, Guoli
2017-08-01
Compressive beamforming realizes the direction-of-arrival (DOA) estimation and strength quantification of acoustic sources by solving an underdetermined system of equations relating microphone pressures to a source distribution via compressive sensing. The conventional method assumes DOAs of sources to lie on a grid. Its performance degrades due to basis mismatch when the assumption is not satisfied. To overcome this limitation for the measurement with plane microphone arrays, a two-dimensional grid-free compressive beamforming is developed. First, a continuum based atomic norm minimization is defined to denoise the measured pressure and thus obtain the pressure from sources. Next, a positive semidefinite programming is formulated to approximate the atomic norm minimization. Subsequently, a reasonably fast algorithm based on alternating direction method of multipliers is presented to solve the positive semidefinite programming. Finally, the matrix enhancement and matrix pencil method is introduced to process the obtained pressure and reconstruct the source distribution. Both simulations and experiments demonstrate that under certain conditions, the grid-free compressive beamforming can provide high-resolution and low-contamination imaging, allowing accurate and fast estimation of two-dimensional DOAs and quantification of source strengths, even with non-uniform arrays and noisy measurements.
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NASA Technical Reports Server (NTRS)
Horne, William C.
2011-01-01
Measurements of background noise were recently obtained with a 24-element phased microphone array in the test section of the Arnold Engineering Development Center 80- by120-Foot Wind Tunnel at speeds of 50 to 100 knots (27.5 to 51.4 m/s). The array was mounted in an aerodynamic fairing positioned with array center 1.2m from the floor and 16 m from the tunnel centerline, The array plate was mounted flush with the fairing surface as well as recessed in. (1.27 cm) behind a porous Kevlar screen. Wind-off speaker measurements were also acquired every 15 on a 10 m semicircular arc to assess directional resolution of the array with various processing algorithms, and to estimate minimum detectable source strengths for future wind tunnel aeroacoustic studies. The dominant background noise of the facility is from the six drive fans downstream of the test section and first set of turning vanes. Directional array response and processing methods such as background-noise cross-spectral-matrix subtraction suggest that sources 10-15 dB weaker than the background can be detected.
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Time Delay Analysis of Turbofan Engine Direct and Indirect Combustion Noise Sources
NASA Technical Reports Server (NTRS)
Miles, Jeffrey Hilton
2008-01-01
The core noise components of a dual spool turbofan engine were separated by the use of a coherence function. A source location technique based on adjusting the time delay between the combustor pressure sensor signal and the far-field microphone signal to maximize the coherence and remove as much variation of the phase angle with frequency as possible was used. The discovery was made that for the 130o microphone a 90.027 ms time shift worked best for the frequency band from 0 to 200 Hz while a 86.975 ms time shift worked best for the frequency band from 200 to 400 Hz. Hence, the 0 to 200 Hz band signal took more time than the 200 to 400 Hz band signal to travel the same distance. This suggests the 0 to 200 Hz coherent cross spectral density band is partly due to indirect combustion noise attributed to entropy fluctuations, which travel at the flow velocity, interacting with the turbine. The signal in the 200 to 400 Hz frequency band is attributed mostly to direct combustion noise. Results are presented herein for engine power settings of 48, 54, and 60 percent of the maximum power setting
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Experimental study of a smart foam sound absorber.
Leroy, Pierre; Berry, Alain; Herzog, Philippe; Atalla, Noureddine
2011-01-01
This article presents the experimental implementation and results of a hybrid passive/active absorber (smart foam) made up from the combination of a passive absorbent (foam) and a curved polyvinylidene fluoride (PVDF) film actuator bonded to the rear surface of the foam. Various smart foam prototypes were built and tested in active absorption experiments conducted in an impedance tube under plane wave propagation condition at frequencies between 100 and 1500 Hz. Three control cases were tested. The first case used a fixed controller derived in the frequency domain from estimations of the primary disturbance at a directive microphone position in the tube and the transfer function between the control PVDF and the directive microphone. The two other cases used an adaptive time-domain feedforward controller to absorb either a single-frequency incident wave or a broadband incident wave. The non-linearity of the smart foams and the causality constraint were identified to be important factors influencing active control performance. The effectiveness of the various smart foam prototypes is discussed in terms of the active and passive absorption coefficients as well as the control voltage of the PVDF actuator normalized by the incident sound pressure.
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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.
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Contribution of crosstalk to the uncertainty of electrostatic actuator calibrations.
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.
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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.
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Radiation impedance of condenser microphones and their diffuse-field responses.
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.
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An analytical-numerical method for determining the mechanical response of a condenser microphone
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
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An analytical-numerical method for determining the mechanical response of a condenser microphone.
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
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Adaptive Wiener filtering for improved acquisition of distortion product otoacoustic emissions.
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.
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A transmission-line model of back-cavity dynamics for in-plane pressure-differential microphones.
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.
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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.
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Recording high quality speech during tagged cine-MRI studies using a fiber optic microphone.
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.
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Infrasound array observation at Sakurajima volcano
NASA Astrophysics Data System (ADS)
Yokoo, A.; Suzuki, Y. J.; Iguchi, M.
2012-12-01
Showa crater at the southeastern flank of the Sakurajima volcano has erupted since 2006, accompanying intermittent Vulcanian eruptions with small scale ash emissions. We conducted an array observation in the last half of 2011 in order to locate infrasound source generated by the eruptions. The array located 3.5 km apart from the crater was composed of 5 microphones (1kHz sampling) aligned in the radial direction from the crater with 100-m-intervals, and additional 4 microphones (200Hz sampling) in tangential direction to the first line in December 2011. Two peaks, around 2Hz and 0.5Hz, in power spectrum of the infrasound were identified; the former peak would be related to the eigen frequency of the vent of Showa crater, but the latter would be related to ejection of eruption clouds. They should be checked by experimental studies. The first 10 s infrasound signal was made by explosion directly and the following small amplitude infrasound tremors for about 2 min were mostly composed of diffraction and reflection waves from the topography around the volcano, mainly the wall of the Aira Caldera. It shows propagation direction of infrasound tremor after the explosion signals should be carefully examined. Clear change in the height of the infrasound source was not identified while volcanic cloud grew up. Strong eddies of the growing volcanic cloud would not be main sources of such weak infrasound signals, thus, infrasound waves are emitted mainly from (or through) the vent itself.
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Chai, Ying; Tian, Dayong; Wang, Wei; Cui, Hua
2010-10-28
Luminol functionalized gold nanoparticles were used as labels for electrochemiluminescence signal amplification and an ultrasensitive, highly selective, convenient, low cost DNA detection strategy was developed.
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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.
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Dynamically Reconfigurable Microphone Arrays
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
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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.
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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
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Song, Weiling; Zhang, Qiao; Sun, Wenbo
2015-02-11
An ultrasensitive protocol for fluorescent detection of DNA is designed by combining the template enhanced hybridization process (TEHP) with Rolling Circle Amplification (RCA) and Catalytic Hairpin Assembly (CHA), showing a remarkable amplification efficiency.
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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.
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Learning to Decode Nonverbal Cues in Cross-Cultural Interactions
2009-06-01
iPhones support Mac OS X v10.4.10 or later operating system, as well as Windows Vista and XP, and iTunes 7.5 or later. Apple has designed the iPhones to be...Processor; 1G RAM, 1G HD, Direct X9/ATI Radeon 9800 card with dedicated memory; Noise-canceling headset w/ microphone. Apple video iPod (can be
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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.
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Uncovering Spatial Variation in Acoustic Environments Using Sound Mapping.
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.
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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.
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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.
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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
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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.
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Zheng, Tingting; Zhang, Rui; Zhang, Qingfeng; Tan, Tingting; Zhang, Kui; Zhu, Jun-Jie; Wang, Hui
2013-09-18
We have developed a robust enzymatic peptide cleavage-based assay for the ultrasensitive dual-channel detection of matrix metalloproteinase-2 (MMP-2) in human serum using gold-quantum dot (Au-QD) core-satellite nanoprobes.
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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.
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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.
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Lumped-parameters equivalent circuit for condenser microphones modeling.
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.
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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.
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Ultrasensitive Electrochemical Detection of mRNA Using Branched DNA Amplifiers
DOE Office of Scientific and Technical Information (OSTI.GOV)
Mao, Xun; Liu, Guodong; Wang, Shengfu
2008-11-01
We describe here an ultrasensitive electrochemical detection of m RNA protocol without RNA purification and PCR amplification. The new m RNA electrical detection capability is coupled to the amplification feature of branched DNA (bDNA) technology and with the nagnetic beads based electrochemical bioassay.
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FPGA implementation of adaptive beamforming in hearing aids.
Samtani, Kartik; Thomas, Jobin; Varma, G Abhinav; Sumam, David S; Deepu, S P
2017-07-01
Beamforming is a spatial filtering technique used in hearing aids to improve target sound reception by reducing interference from other directions. In this paper we propose improvements in an existing architecture present for two omnidirectional microphone array based adaptive beamforming for hearing aid applications and implement the same on Xilinx Artix 7 FPGA using VHDL coding and Xilinx Vivado ® 2015.2. The nulls are introduced in particular directions by combination of two fixed polar patterns. This combination can be adaptively controlled to steer the null in the direction of noise. The beamform patterns and improvements in SNR values obtained from experiments in a conference room environment are analyzed.
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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.
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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.
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NASA Technical Reports Server (NTRS)
Hall, David G.; Bridges, James
1992-01-01
A sophisticated, multi-channel computerized data acquisition and processing system was developed at the NASA LeRC for use in noise experiments. This technology, which is available for transfer to industry, provides a convenient, cost-effective alternative to analog tape recording for high frequency acoustic measurements. This system provides 32-channel acquisition of microphone signals with an analysis bandwidth up to 100 kHz per channel. Cost was minimized through the use of off-the-shelf components. Requirements to allow for future expansion were met by choosing equipment which adheres to established industry standards for hardware and software. Data processing capabilities include narrow band and 1/3 octave spectral analysis, compensation for microphone frequency response/directivity, and correction of acoustic data to standard day conditions. The system was used successfully in a major wind tunnel test program at NASA LeRC to acquire and analyze jet noise data in support of the High Speed Civil Transport (HSCT) program.
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Acoustic source localization in mixed field using spherical microphone arrays
NASA Astrophysics Data System (ADS)
Huang, Qinghua; Wang, Tong
2014-12-01
Spherical microphone arrays have been used for source localization in three-dimensional space recently. In this paper, a two-stage algorithm is developed to localize mixed far-field and near-field acoustic sources in free-field environment. In the first stage, an array signal model is constructed in the spherical harmonics domain. The recurrent relation of spherical harmonics is independent of far-field and near-field mode strengths. Therefore, it is used to develop spherical estimating signal parameter via rotational invariance technique (ESPRIT)-like approach to estimate directions of arrival (DOAs) for both far-field and near-field sources. In the second stage, based on the estimated DOAs, simple one-dimensional MUSIC spectrum is exploited to distinguish far-field and near-field sources and estimate the ranges of near-field sources. The proposed algorithm can avoid multidimensional search and parameter pairing. Simulation results demonstrate the good performance for localizing far-field sources, or near-field ones, or mixed field sources.
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Torres, Ana M; Lopez, Jose J; Pueo, Basilio; Cobos, Maximo
2013-04-01
Plane-wave decomposition (PWD) methods using microphone arrays have been shown to be a very useful tool within the applied acoustics community for their multiple applications in room acoustics analysis and synthesis. While many theoretical aspects of PWD have been previously addressed in the literature, the practical advantages of the PWD method to assess the acoustic behavior of real rooms have been barely explored so far. In this paper, the PWD method is employed to analyze the sound field inside a selected set of real rooms having a well-defined purpose. To this end, a circular microphone array is used to capture and process a number of impulse responses at different spatial positions, providing angle-dependent data for both direct and reflected wavefronts. The detection of reflected plane waves is performed by means of image processing techniques applied over the raw array response data and over the PWD data, showing the usefulness of image-processing-based methods for room acoustics analysis.
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Polymers in solar energy utilization
NASA Technical Reports Server (NTRS)
Liang, R. H.; Coulter, D. R.; Dao, C.; Gupta, A.
1983-01-01
A laser photoacoustic technique (LPAT) has been verified for performing accelerated life testing of outdoor photooxidation of polymeric materials used in solar energy applications. Samples of the material under test are placed in a chamber with a sensitive microphone, then exposed to chopped laser radiation. The sample absorbs the light and converts it to heat by a nonradiative deexcitation process, thereby reducing pressure fluctuations within the cell. The acoustic signal detected by the microphone is directly proportional to the amount of light absorbed by the specimen. Tests were performed with samples of ethylene/methylacrylate copolymer (EMA) reprecipitated from hot cyclohexane, compressed, and molded into thin (25-50 microns) films. The films were exposed outdoors and sampled by LPAT weekly. The linearity of the light absorbed with respect to the acoustic signal was verified.Correlations were established between the photoacoustic behavior of the materials aged outdoors and the same kinds of samples cooled and heated in a controlled environment reactor. The reactor tests were validated for predicting outdoor exosures up to 55 days.
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Measurement of hearing aid internal noise1
Lewis, James D.; Goodman, Shawn S.; Bentler, Ruth A.
2010-01-01
Hearing aid equivalent input noise (EIN) measures assume the primary source of internal noise to be located prior to amplification and to be constant regardless of input level. EIN will underestimate internal noise in the case that noise is generated following amplification. The present study investigated the internal noise levels of six hearing aids (HAs). Concurrent with HA processing of a speech-like stimulus with both adaptive features (acoustic feedback cancellation, digital noise reduction, microphone directionality) enabled and disabled, internal noise was quantified for various stimulus levels as the variance across repeated trials. Changes in noise level as a function of stimulus level demonstrated that (1) generation of internal noise is not isolated to the microphone, (2) noise may be dependent on input level, and (3) certain adaptive features may contribute to internal noise. Quantifying internal noise as the variance of the output measures allows for noise to be measured under real-world processing conditions, accounts for all sources of noise, and is predictive of internal noise audibility. PMID:20370034
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The BetaCage, an ultra-sensitive screener for surface contamination
NASA Astrophysics Data System (ADS)
Bunker, R.; Ahmed, Z.; Bowles, M. A.; Golwala, S. R.; Grant, D. R.; Kos, M.; Nelson, R. H.; Schnee, R. W.; Rider, A.; Wang, B.; Zahn, A.
2013-08-01
Material screening for identifying low-energy electron emitters and alpha-decaying isotopes is now a prerequisite for rare-event searches (e.g., dark-matter direct detection and neutrinoless double-beta decay) for which surface radiocon-tamination has become an increasingly important background. The BetaCage, a gaseous neon time-projection chamber, is a proposed ultra-sensitive (and nondestructive) screener for alpha-and beta-emitting surface contaminants to which existing screening facilities are insufficiently sensitive. Sensitivity goals are 0.1 betas keV-1 m-2 day-1 and 0.1 alphas m-2 day-1, with the former limited by Compton scattering of photons in the screening samples and (thanks to tracking) the latter expected to be signal-limited; radioassays and simulations indicate backgrounds from detector materials and radon daughters should be subdominant. We report on details of the background simulations and detector design that provide the discrimination, shielding, and radiopurity necessary to reach our sensitivity goals for a chamber with a 95 × 95 cm2 sample area positioned below a 40 cm drift region and monitored by crisscrossed anode and cathode planes consisting of 151 wires each.
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New Fluorescent Nanoparticles for Ultrasensitive Detection of Nucleic Acids by Optical Methods.
Westergaard Mulberg, Mads; Taskova, Maria; Thomsen, Rasmus P; Okholm, Anders H; Kjems, Jørgen; Astakhova, Kira
2017-08-17
For decades the detection of nucleic acids and their interactions at low abundances has been a challenging task that has thus far been solved by enzymatic target amplification. In this work we aimed at developing efficient tools for amplification-free nucleic acid detection, which resulted in the synthesis of new fluorescent nanoparticles. Here, the fluorescent nanoparticles were made by simple and inexpensive radical emulsion polymerization of butyl acrylate in the presence of fluorescent dyes and additional functionalization reagents. This provided ultra-bright macrofluorophores of 9-84 nm mean diameter, modified with additional alkyne and amino groups for bioconjugation. By using click and NHS chemistries, the new nanoparticles were attached to target-specific DNA probes that were used in fluorimetry and fluorescence microscopy. Overall, these fluorescent nanoparticles and their oligonucleotide derivatives have higher photostability, brighter fluorescence and hence dramatically lower limits of target detection than the individual organic dyes. These properties make them useful in approaches directed towards ultrasensitive detection of nucleic acids, in particular for imaging and in vitro diagnostics of DNA. © 2017 Wiley-VCH Verlag GmbH & Co. KGaA, Weinheim.
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Gao, Anran; Lu, Na; Dai, Pengfei; Fan, Chunhai; Wang, Yuelin; Li, Tie
2014-11-07
Sensitive and quantitative analysis of proteins is central to disease diagnosis, drug screening, and proteomic studies. Here, a label-free, real-time, simultaneous and ultrasensitive prostate-specific antigen (PSA) sensor was developed using CMOS-compatible silicon nanowire field effect transistors (SiNW FET). Highly responsive n- and p-type SiNW arrays were fabricated and integrated on a single chip with a complementary metal oxide semiconductor (CMOS) compatible anisotropic self-stop etching technique which eliminated the need for a hybrid method. The incorporated n- and p-type nanowires revealed complementary electrical response upon PSA binding, providing a unique means of internal control for sensing signal verification. The highly selective, simultaneous and multiplexed detection of PSA marker at attomolar concentrations, a level useful for clinical diagnosis of prostate cancer, was demonstrated. The detection ability was corroborated to be effective by comparing the detection results at different pH values. Furthermore, the real-time measurement was also carried out in a clinically relevant sample of blood serum, indicating the practicable development of rapid, robust, high-performance, and low-cost diagnostic systems.
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Xiong, Xu-Jie; Rao, Wan-Bing; Guo, Xiao-Feng; Wang, Hong; Zhang, Hua-Shan
2012-05-23
An ultrasensitive and selective high-performance liquid chromatographic method for the volatile signaling hormone, jasmonic acid, has been developed based on precolumn derivatization with 1,3,5,7-tetramethyl-8-aminozide-difluoroboradiaza-s-indacene (BODIPY-aminozide). The derivatization reaction was carried out at 60 °C for 30 min in the presence of phosphoric acid. The formed jasmonic acid derivative was eluted using a mobile phase of methanol/pH 6.50 ammonium formate buffer/tetrahydrofuran (67:30:3, v/v/v) in 10 min on a C(18) column and detected with fluorescence detection at excitation and emission wavelengths of 495 and 505 nm, respectively. The detection limit (signal-to-noise ratio = 4) reached 1.14 × 10(-10) M or 2.29 fmol per injection (20 μL), which is the lowest of the existing methods. The proposed method has been successfully applied to the direct determination of trace jasmonic acid in the crude extracts of soybean leaves from soybean mosaic virus-infected and normal plants with recoveries of 95-104%.
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Guo, Zhiyong; Sha, Yuhong; Hu, Yufang; Yu, Zhongqing; Tao, Yingying; Wu, Yanjie; Zeng, Min; Wang, Sui; Li, Xing; Zhou, Jun; Su, Xiurong
2016-10-01
A novel Faraday cage-type electrochemiluminescence (ECL) immunosensor devoted to the detection of Vibrio vulnificus (VV) was fabricated. The sensing strategy was presented by a unique Faraday cage-type immunocomplex based on immunomagnetic beads (IMBs) and multi-functionalized graphene oxide (GO) labeled with (2,2'-bipyridine)(5-aminophenanthroline)ruthenium (Ru-NH2). The multi-functionalized GO could sit on the electrode surface directly due to the large surface area, abundant functional groups, and good electronic transport property. It ensures that more Ru-NH2 is entirely caged and become "effective," thus improving sensitivity significantly, which resembles extending the outer Helmholtz plane (OHP) of the electrode. Under optimal conditions, the developed immunosensor achieves a limit of detection as low as 1 CFU/mL. Additionally, the proposed immunosensor with high sensitivity and selectivity can be used for the detection of real samples. The novel Faraday cage-type method has shown potential application for the diagnosis of VV and opens up a new avenue in ECL immunoassay. Graphical abstract Faraday cage-type immunoassay mode for ultrasensitive detection by extending OHP.
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DNA-engineered chiroplasmonic heteropyramids for ultrasensitive detection of mercuryion
USDA-ARS?s Scientific Manuscript database
In this study, plasmonic heteropyramids (HPs) made from two different sized gold nanoparticles (Au NPs) and five ssDNA sequences and their application for ultrasensitive detection of mercury ion (Hg2+) were demonstrated. Four ssDNA sequences were used as building blocks to form apyramidal DNA frame,...
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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.
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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.
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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.
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The Measurement of Unsteady Surface Pressure Using a Remote Microphone Probe.
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.
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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.
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Initial Results from the Variable Intensity Sonic Boom Propagation Database
NASA Technical Reports Server (NTRS)
Haering, Edward A., Jr.; Cliatt, Larry J., II; Bunce, Thomas J.; Gabrielson, Thomas B.; Sparrow, Victor W.; Locey, Lance L.
2008-01-01
An extensive sonic boom propagation database with low- to normal-intensity booms (overpressures of 0.08 lbf/sq ft to 2.20 lbf/sq ft) was collected for propagation code validation, and initial results and flight research techniques are presented. Several arrays of microphones were used, including a 10 m tall tower to measure shock wave directionality and the effect of height above ground on acoustic level. A sailplane was employed to measure sonic booms above and within the atmospheric turbulent boundary layer, and the sailplane was positioned to intercept the shock waves between the supersonic airplane and the ground sensors. Sailplane and ground-level sonic boom recordings were used to generate atmospheric turbulence filter functions showing excellent agreement with ground measurements. The sonic boom prediction software PCBoom4 was employed as a preflight planning tool using preflight weather data. The measured data of shock wave directionality, arrival time, and overpressure gave excellent agreement with the PCBoom4-calculated results using the measured aircraft and atmospheric data as inputs. C-weighted acoustic levels generally decreased with increasing height above the ground. A-weighted and perceived levels usually were at a minimum for a height where the elevated microphone pressure rise time history was the straightest, which is a result of incident and ground-reflected shock waves interacting.
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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.
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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.
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Plasma Enhanced Growth of Carbon Nanotubes For Ultrasensitive Biosensors
NASA Technical Reports Server (NTRS)
Cassell, Alan M.; Meyyappan, M.
2004-01-01
The multitude of considerations facing nanostructure growth and integration lends itself to combinatorial optimization approaches. Rapid optimization becomes even more important with wafer-scale growth and integration processes. Here we discuss methodology for developing plasma enhanced CVD growth techniques for achieving individual, vertically aligned carbon nanostructures that show excellent properties as ultrasensitive electrodes for nucleic acid detection. We utilize high throughput strategies for optimizing the upstream and downstream processing and integration of carbon nanotube electrodes as functional elements in various device types. An overview of ultrasensitive carbon nanotube based sensor arrays for electrochemical bio-sensing applications and the high throughput methodology utilized to combine novel electrode technology with conventional MEMS processing will be presented.
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Plasma Enhanced Growth of Carbon Nanotubes For Ultrasensitive Biosensors
NASA Technical Reports Server (NTRS)
Cassell, Alan M.; Li, J.; Ye, Q.; Koehne, J.; Chen, H.; Meyyappan, M.
2004-01-01
The multitude of considerations facing nanostructure growth and integration lends itself to combinatorial optimization approaches. Rapid optimization becomes even more important with wafer-scale growth and integration processes. Here we discuss methodology for developing plasma enhanced CVD growth techniques for achieving individual, vertically aligned carbon nanostructures that show excellent properties as ultrasensitive electrodes for nucleic acid detection. We utilize high throughput strategies for optimizing the upstream and downstream processing and integration of carbon nanotube electrodes as functional elements in various device types. An overview of ultrasensitive carbon nanotube based sensor arrays for electrochemical biosensing applications and the high throughput methodology utilized to combine novel electrode technology with conventional MEMS processing will be presented.
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The Benefit of a Visually Guided Beamformer in a Dynamic Speech Task
Roverud, Elin; Streeter, Timothy; Mason, Christine R.; Kidd, Gerald
2017-01-01
The aim of this study was to evaluate the performance of a visually guided hearing aid (VGHA) under conditions designed to capture some aspects of “real-world” communication settings. The VGHA uses eye gaze to steer the acoustic look direction of a highly directional beamforming microphone array. Although the VGHA has been shown to enhance speech intelligibility for fixed-location, frontal targets, it is currently not known whether these benefits persist in the face of frequent changes in location of the target talker that are typical of conversational turn-taking. Participants were 14 young adults, 7 with normal hearing and 7 with bilateral sensorineural hearing impairment. Target stimuli were sequences of 12 question–answer pairs that were embedded in a mixture of competing conversations. The participant’s task was to respond via a key press after each answer indicating whether it was correct or not. Spatialization of the stimuli and microphone array processing were done offline using recorded impulse responses, before presentation over headphones. The look direction of the array was steered according to the eye movements of the participant as they followed a visual cue presented on a widescreen monitor. Performance was compared for a “dynamic” condition in which the target stimulus moved between three locations, and a “fixed” condition with a single target location. The benefits of the VGHA over natural binaural listening observed in the fixed condition were reduced in the dynamic condition, largely because visual fixation was less accurate. PMID:28758567
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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.
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González-Guerrero, Ana Belén; Maldonado, Jesús; Dante, Stefania; Grajales, Daniel; Lechuga, Laura M
2017-01-01
A label-free interferometric transducer showing a theoretical detection limit for homogeneous sensing of 5 × 10 -8 RIU, being equivalent to a protein mass coverage resolution of 2.8 fg mm -2 , is used to develop a high sensitive biosensor for protein detection. The extreme sensitivity of this transducer combined with a selective bioreceptor layer enables the direct evaluation of the human growth hormone (hGH) in undiluted urine matrix in the 10 pg mL -1 range. © 2016 WILEY-VCH Verlag GmbH & Co. KGaA, Weinheim.
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Theoretical Investigation of Device Aspects of Semiconductor Superlattices.
1983-09-01
n-i-p-i devices include bulk field effect transistors, ultrasensitive or ultrafast IR photodetectors , tunable light-emitting devices, and ultrafast...transistor4 ultrasensitive or ultrafast IR photodetectors , tunable light-emitt tg devices, and ultrafast optical modulators. Particularlylppealing...differential conductivity ( NDC ) ......................... 19 3.2.2. Spontaneous and stimulated FIR emission from interlayer transitions
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Yan, Qiang; Yuan, Jinying; Kang, Yan; Cai, Zhinan; Zhou, Lilin; Yin, Yingwu
2010-04-28
A porphyrin-containing copolymer has dual-sensing in response to metal ions and temperature as a novel nanosensor. Triggered by ions, the sensor exhibits full-color tunable behavior as a cationic detector and colorimeter. Responding to temperature, the sensor displays an "isothermal" thermochromic point as an ultra-sensitive thermometer.
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Evaluation of a novel ultra-sensitive nanoparticle probe-based assay for ricin detection.
Yin, Hui-qiong; Jia, Min-xian; Shi, Li-jun; Liu, Jun; Wang, Rui; Lv, Mao-min; Ma, Yu-yuan; Zhao, Xiong; Zhang, Jin-gang
2014-01-01
A gold nanoparticle (GNP) probe-based assay (GNPA) modified from the bio-barcode assay (BCA) was developed for ultrasensitive detection of ricin, a potential biothreat agent. In the GNPA, a chain of ricin was captured by a GNP probe coated with polyclonal antibodies and single-stranded signal DNA. A magnetic microparticle (MMP) probe coated with ricin A chain monoclonal antibody was then added to form an immuno-complex. After being magnetically separated, the immuno-complex containing the single-stranded signal DNA was characterized by PCR and real-time PCR. A detection limit of 10(-2) fg/ml was determined for the ricin A chain; this is eight orders of magnitude more sensitive than that achieved with an ELISA and two orders more sensitive than that obtained with the BCA. The coefficients of variation (CV) of the intra- and inter-assay values ranged from 3.82-6.46%. The results here show that this novel assay is an ultrasensitive method for detection of ricin proteins and may be suitable for the ultrasensitive detection of other proteins.
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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...
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Characterizing acoustic shocks in high-performance jet aircraft flyover noise.
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.
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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.
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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.
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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
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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.
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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.
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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.
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Nanobody medicated immunoassay for ultrasensitive detection of cancer biomarker alpha-fetoprotein.
Chen, Jing; He, Qing-hua; Xu, Yang; Fu, Jin-heng; Li, Yan-ping; Tu, Zhui; Wang, Dan; Shu, Mei; Qiu, Yu-lou; Yang, Hong-wei; Liu, Yuan-yuan
2016-01-15
Immunoassay for cancer biomarkers plays an important role in cancer prevention and early diagnosis. To the development of immunoassay, the quality and stability of applied antibody is one of the key points to obtain reliability and high sensitivity for immunoassay. The main purpose of this study was to develop a novel immunoassay for ultrasensitive detection of cancer biomarker alpha-fetoprotein (AFP) based on nanobody against AFP. Two nanobodies which bind to AFP were selected from a phage display nanobody library by biopanning strategy. The prepared nanobodies are clonable, thermally stable and applied in both sandwich enzyme linked immunoassay (ELISA) and immuno-PCR assay for ultrasensitive detection of AFP. The limit detection of sandwich ELISA setup with optimized nanobodies was 0.48ng mL(-1), and the half of saturation concentration (SC50) value was 6.68±0.56ng mL(-1). These nanobodies were also used to develop an immuno-PCR assay for ultrasensitive detection of AFP, its limit detection values was 0.005ng mL(-1), and the linear range was 0.01-10,000ng mL(-1). These established immunoassays based on nanobodies were highly specific to AFP and with negligible cross reactivity with other tested caner biomarkers. Furthermore, this novel concept of nanobodies mediated immunoassay may provide potential applications in a general method for the ultrasensitive detection of various cancer biomarkers. Copyright © 2015 Elsevier B.V. All rights reserved.
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Immuno Nanosensor for the Ultrasensitive Naked Eye Detection of Tuberculosis.
Mohd Bakhori, Noremylia; Yusof, Nor Azah; Abdullah, Jaafar; Wasoh, Helmi; Md Noor, Siti Suraiya; Ahmad Raston, Nurul Hanun; Mohammad, Faruq
2018-06-14
In the present study, a beneficial approach for the ultrasensitive and affordable naked eye detection and diagnosis of tuberculosis (TB) by utilizing plasmonic enzyme-linked immunosorbent assay (ELISA) via antibody-antigen interaction was studied. Here, the biocatalytic cycle of the intracellular enzymes links to the formation and successive growth of the gold nanoparticles (GNPs) for ultrasensitive detection. The formation of different colored solutions by the plasmonic nanoparticles in the presence of enzyme labels links directly to the existence or non-existence of the TB analytes in the sample solutions. For disease detection, the adapted protocol is based mainly on the conventional ELISA procedure that involves catalase-labeled antibodies, i.e., the enzymes consume hydrogen peroxide and further produce GNPs with the addition of gold (III) chloride. The amount of hydrogen peroxide remaining in the solution determines whether the GNPs solution is to be formed in the color blue or the color red, as it serves as a confirmation for the naked eye detection of TB analytes. However, the conventional ELISA method only shows tonal colors that need a high concentration of analyte to achieve high confidence levels for naked eye detection. Also, in this research, we proposed the incorporation of protein biomarker, Mycobacterium tuberculosis ESAT-6-like protein esxB (CFP-10), as a means of TB detection using plasmonic ELISA. With the use of this technique, the CFP-10 detection limit can be lowered to 0.01 µg/mL by the naked eye. Further, our developed technique was successfully tested and confirmed with sputum samples from patients diagnosed with positive TB, thereby providing enough evidence for the utilization of our technique in the early diagnosis of TB disease.
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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...
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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.
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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.
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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.
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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.
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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.
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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
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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).
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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.
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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
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Advantages of binaural amplification to acceptable noise level of directional hearing aid users.
Kim, Ja-Hee; Lee, Jae Hee; Lee, Ho-Ki
2014-06-01
The goal of the present study was to examine whether Acceptable Noise Levels (ANLs) would be lower (greater acceptance of noise) in binaural listening than in monaural listening condition and also whether meaningfulness of background speech noise would affect ANLs for directional microphone hearing aid users. In addition, any relationships between the individual binaural benefits on ANLs and the individuals' demographic information were investigated. Fourteen hearing aid users (mean age, 64 years) participated for experimental testing. For the ANL calculation, listeners' most comfortable listening levels and background noise level were measured. Using Korean ANL material, ANLs of all participants were evaluated under monaural and binaural amplification with a counterbalanced order. The ANLs were also compared across five types of competing speech noises, consisting of 1- through 8-talker background speech maskers. Seven young normal-hearing listeners (mean age, 27 years) participated for the same measurements as a pilot testing. The results demonstrated that directional hearing aid users accepted more noise (lower ANLs) with binaural amplification than with monaural amplification, regardless of the type of competing speech. When the background speech noise became more meaningful, hearing-impaired listeners accepted less amount of noise (higher ANLs), revealing that ANL is dependent on the intelligibility of the competing speech. The individuals' binaural advantages in ANLs were significantly greater for the listeners with longer experience of hearing aids, yet not related to their age or hearing thresholds. Binaural directional microphone processing allowed hearing aid users to accept a greater amount of background noise, which may in turn improve listeners' hearing aid success. Informational masking substantially influenced background noise acceptance. Given a significant association between ANLs and duration of hearing aid usage, ANL measurement can be useful for clinical counseling of binaural hearing aid candidates or unsuccessful users.
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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
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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
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Feasible pickup from intact ossicular chain with floating piezoelectric microphone.
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.
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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.
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Rectified directional sensing in long-range cell migration
Nakajima, Akihiko; Ishihara, Shuji; Imoto, Daisuke; Sawai, Satoshi
2014-01-01
How spatial and temporal information are integrated to determine the direction of cell migration remains poorly understood. Here, by precise microfluidics emulation of dynamic chemoattractant waves, we demonstrate that, in Dictyostelium, directional movement as well as activation of small guanosine triphosphatase Ras at the leading edge is suppressed when the chemoattractant concentration is decreasing over time. This ‘rectification’ of directional sensing occurs only at an intermediate range of wave speed and does not require phosphoinositide-3-kinase or F-actin. From modelling analysis, we show that rectification arises naturally in a single-layered incoherent feedforward circuit with zero-order ultrasensitivity. The required stimulus time-window predicts ~5 s transient for directional sensing response close to Ras activation and inhibitor diffusion typical for protein in the cytosol. We suggest that the ability of Dictyostelium cells to move only in the wavefront is closely associated with rectification of adaptive response combined with local activation and global inhibition. PMID:25373620
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Kwon, Seok Joon; Lee, Kyung Bok; Solakyildirim, Kemal; Masuko, Sayaka; Ly, Mellisa; Zhang, Fuming; Li, Lingyun; Dordick, Jonathan S.; Linhardt, Robert J.
2012-01-01
Tiny amounts of carbohydrates (ca. 1 zmol) can be detected quantitatively by a real-time method based on the conjugation of carbohydrates with DNA markers (see picture). The proposed method (glyco-qPCR) provides uniform, ultrasensitive detection of carbohydrates, which can be applied to glycobiology, as well as carbohydrate-based drug discovery. PMID:23073897
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USDA-ARS?s Scientific Manuscript database
A rapid lateral flow fluorescent microspheres immunochromatography test strip (FMs-ICTS) has been developed for the detection of aflatoxin M1 (AFM1) residues in milk. For this purpose, an ultra-sensitive anti-AFM1 monoclonal antibody (MAb) 1D3 was prepared and identified. The IC50 value of the MA...
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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...
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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.
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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.
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Direct Measurements of Interplanetary Dust Particles in the Vicinity of Earth
NASA Technical Reports Server (NTRS)
McCracken, C. W.; Alexander, W. M.; Dubin, M.
1961-01-01
The direct measurements made by the Explorer VIII satellite provide the first sound basis for analyzing all available direct measurements of the distribution of interplanetary dust particles. The model average distribution curve established by such an analysis departs significantly from that predicted by the (uncertain) extrapolation of results from meteor observations. A consequence of this difference is that the daily accretion of interplanetary particulate matter by the earth is now considered to be mainly dust particles of the direct measurements range of particle size. Almost all the available direct measurements obtained with microphone systems on rockets, satellites, and spacecraft fit directly on the distribution curve defined by Explorer VIII data. The lack of reliable datum points departing significantly from the model average distribution curve means that available direct measurements show no discernible evidence of an appreciable geocentric concentration of interplanetary dust particles.
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Visualizing Sound Directivity via Smartphone Sensors
NASA Astrophysics Data System (ADS)
Hawley, Scott H.; McClain, Robert E.
2018-02-01
When Yang-Hann Kim received the Rossing Prize in Acoustics Education at the 2015 meeting of the Acoustical Society of America, he stressed the importance of offering visual depictions of sound fields when teaching acoustics. Often visualization methods require specialized equipment such as microphone arrays or scanning apparatus. We present a simple method for visualizing angular dependence in sound fields, made possible via the confluence of sensors available via a new smartphone app that the authors have developed.
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Active control of noise on the source side of a partition to increase its sound isolation
NASA Astrophysics Data System (ADS)
Tarabini, Marco; Roure, Alain; Pinhede, Cedric
2009-03-01
This paper describes a local active noise control system that virtually increases the sound isolation of a dividing wall by means of a secondary source array. With the proposed method, sound pressure on the source side of the partition is reduced using an array of loudspeakers that generates destructive interference on the wall surface, where an array of error microphones is placed. The reduction of sound pressure on the incident side of the wall is expected to decrease the sound radiated into the contiguous room. The method efficiency was experimentally verified by checking the insertion loss of the active noise control system; in order to investigate the possibility of using a large number of actuators, a decentralized FXLMS control algorithm was used. Active control performances and stability were tested with different array configurations, loudspeaker directivities and enclosure characteristics (sound source position and absorption coefficient). The influence of all these parameters was investigated with the factorial design of experiments. The main outcome of the experimental campaign was that the insertion loss produced by the secondary source array, in the 50-300 Hz frequency range, was close to 10 dB. In addition, the analysis of variance showed that the active noise control performance can be optimized with a proper choice of the directional characteristics of the secondary source and the distance between loudspeakers and error microphones.
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Echolocating bats use future-target information for optimal foraging.
Fujioka, Emyo; Aihara, Ikkyu; Sumiya, Miwa; Aihara, Kazuyuki; Hiryu, Shizuko
2016-04-26
When seeing or listening to an object, we aim our attention toward it. While capturing prey, many animal species focus their visual or acoustic attention toward the prey. However, for multiple prey items, the direction and timing of attention for effective foraging remain unknown. In this study, we adopted both experimental and mathematical methodology with microphone-array measurements and mathematical modeling analysis to quantify the attention of echolocating bats that were repeatedly capturing airborne insects in the field. Here we show that bats select rational flight paths to consecutively capture multiple prey items. Microphone-array measurements showed that bats direct their sonar attention not only to the immediate prey but also to the next prey. In addition, we found that a bat's attention in terms of its flight also aims toward the next prey even when approaching the immediate prey. Numerical simulations revealed a possibility that bats shift their flight attention to control suitable flight paths for consecutive capture. When a bat only aims its flight attention toward its immediate prey, it rarely succeeds in capturing the next prey. These findings indicate that bats gain increased benefit by distributing their attention among multiple targets and planning the future flight path based on additional information of the next prey. These experimental and mathematical studies allowed us to observe the process of decision making by bats during their natural flight dynamics.
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Directional hearing aid using hybrid adaptive beamformer (HAB) and binaural ITE array
NASA Astrophysics Data System (ADS)
Shaw, Scott T.; Larow, Andy J.; Gibian, Gary L.; Sherlock, Laguinn P.; Schulein, Robert
2002-05-01
A directional hearing aid algorithm called the Hybrid Adaptive Beamformer (HAB), developed for NIH/NIA, can be applied to many different microphone array configurations. In this project the HAB algorithm was applied to a new array employing in-the-ear microphones at each ear (HAB-ITE), to see if previous HAB performance could be achieved with a more cosmetically acceptable package. With diotic output, the average benefit in threshold SNR was 10.9 dB for three HoH and 11.7 dB for five normal-hearing subjects. These results are slightly better than previous results of equivalent tests with a 3-in. array. With an innovative binaural fitting, a small benefit beyond that provided by diotic adaptive beamforming was observed: 12.5 dB for HoH and 13.3 dB for normal-hearing subjects, a 1.6 dB improvement over the diotic presentation. Subjectively, the binaural fitting preserved binaural hearing abilities, giving the user a sense of space, and providing left-right localization. Thus the goal of creating an adaptive beamformer that simultaneously provides excellent noise reduction and binaural hearing was achieved. Further work remains before the HAB-ITE can be incorporated into a real product, optimizing binaural adaptive beamforming, and integrating the concept with other technologies to produce a viable product prototype. [Work supported by NIH/NIDCD.
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Wind noise within and across behind-the-ear and miniature behind-the-ear hearing aids.
Zakis, Justin A; Hawkins, Daniel J
2015-10-01
Previous studies investigated wind noise with Behind-The-Ear (BTE) hearing aids, but not the more common mini-BTE style of device, which typically has a smaller shell and microphones located more deeply behind the pinna. The current study investigated wind-noise levels across one BTE and two mini-BTE devices, and between the front and rear omni-directional microphones within devices. Levels were measured at two wind speeds (3 and 6 m/s) and 36 wind azimuths (10° increments). The pattern of wind-noise level versus azimuth was similar across mini-BTE devices, and differed for the BTE device. However, mean levels were markedly different across mini-BTE devices, and could be higher, lower, or similar to those of the BTE device. For within-device level differences, the pattern and mean across azimuth were similar across mini-BTE devices, and differed for the BTE device. Wind noise had the potential to slightly or severely reduce speech intelligibility at 3 or 6 m/s, respectively, across all devices.
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NASA Technical Reports Server (NTRS)
Haviland, J. K.; Schroeder, J. C.
1978-01-01
As part of an overall study of the scaling laws for the fluctuating pressures induced on the wings and flaps of STOL aircraft by jet engine impingement, an experimental investigation was made of the near field fluctuating pressures behind a cold circular jet, both when it was free and when it was impinging on a flat plate. Miniature static pressure probes were developed for measurements in the free jet and on the flat plate which were connected by plastic tubing to 1/8 inch microphones and acted as pressure transducers. Using a digital correlator together with an FFT program on the CDC 6400 computer, spectral densities, relative amplitudes, phase lags, and coherences were also obtained for the signals from pairs of these probes, and were used to calibrate these probes directly against microphones. This system of instrumentation was employed to obtain single point rms and third octave surveys of the static pressures in the free jet and on the surface of the plate.
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NASA Astrophysics Data System (ADS)
Hiryu, Shizuko; Katsura, Koji; Lin, Liang-Kong; Riquimaroux, Hiroshi; Watanabe, Yoshiaki
2005-12-01
Biosonar behavior was examined in Taiwanese leaf-nosed bats (Hipposideros terasensis; CF-FM bats) during flight. Echolocation sounds were recorded using a telemetry microphone mounted on the bat's head. Flight speed and three-dimensional trajectory of the bat were reconstructed from images taken with a dual high-speed video camera system. Bats were observed to change the intensity and emission rate of pulses depending on the distance from the landing site. Frequencies of the dominant second harmonic constant frequency component (CF2) of calls estimated from the bats' flight speed agreed strongly with observed values. Taiwanese leaf-nosed bats changed CF2 frequencies depending on flight speed, which caused the CF2 frequencies of the Doppler-shifted echoes to remain constant. Pulse frequencies were also estimated using echoes returning directly ahead of the bat and from its sides for two different flight conditions: landing and U-turn. Bats in flight may periodically alter their attended angles from the front to the side when emitting echolocation pulses.
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Analyzing acoustic phenomena with a smartphone microphone
NASA Astrophysics Data System (ADS)
Kuhn, Jochen; Vogt, Patrik
2013-02-01
This paper describes how different sound types can be explored using the microphone of a smartphone and a suitable app. Vibrating bodies, such as strings, membranes, or bars, generate air pressure fluctuations in their immediate vicinity, which propagate through the room in the form of sound waves. Depending on the triggering mechanism, it is possible to differentiate between four types of sound waves: tone, sound, noise, and bang. In everyday language, non-experts use the terms "tone" and "sound" synonymously; however, from a physics perspective there are very clear differences between the two terms. This paper presents experiments that enable learners to explore and understand these differences. Tuning forks and musical instruments (e.g., recorders and guitars) can be used as equipment for the experiments. The data are captured using a smartphone equipped with the appropriate app (in this paper we describe the app Audio Kit for iOS systems ). The values captured by the smartphone are displayed in a screen shot and then viewed directly on the smartphone or exported to a computer graphics program for printing.
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Investigation of helicopter rotor blade/wake interactive impulsive noise
NASA Technical Reports Server (NTRS)
Miley, S. J.; Hall, G. F.; Vonlavante, E.
1987-01-01
An analysis of the Tip Aerodynamic/Aeroacoustic Test (TAAT) data was performed to identify possible aerodynamic sources of blade/vortex interaction (BVI) impulsive noise. The identification is based on correlation of measured blade pressure time histories with predicted blade/vortex intersections for the flight condition(s) where impulsive noise was detected. Due to the location of the recording microphones, only noise signatures associated with the advancing blade were available, and the analysis was accordingly restricted to the first and second azimuthal quadrants. The results show that the blade tip region is operating transonically in the azimuthal range where previous BVI experiments indicated the impulsive noise to be. No individual blade/vortex encounter is identifiable in the pressure data; however, there is indication of multiple intersections in the roll-up region which could be the origin of the noise. Discrete blade/vortex encounters are indicated in the second quadrant; however, if impulsive noise were produced here, the directivity pattern would be such that it was not recorded by the microphones. It is demonstrated that the TAAT data base is a valuable resource in the investigation of rotor aerodynamic/aeroacoustic behavior.
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Hands-free human-machine interaction with voice
NASA Astrophysics Data System (ADS)
Juang, B. H.
2004-05-01
Voice is natural communication interface between a human and a machine. The machine, when placed in today's communication networks, may be configured to provide automation to save substantial operating cost, as demonstrated in AT&T's VRCP (Voice Recognition Call Processing), or to facilitate intelligent services, such as virtual personal assistants, to enhance individual productivity. These intelligent services often need to be accessible anytime, anywhere (e.g., in cars when the user is in a hands-busy-eyes-busy situation or during meetings where constantly talking to a microphone is either undersirable or impossible), and thus call for advanced signal processing and automatic speech recognition techniques which support what we call ``hands-free'' human-machine communication. These techniques entail a broad spectrum of technical ideas, ranging from use of directional microphones and acoustic echo cancellatiion to robust speech recognition. In this talk, we highlight a number of key techniques that were developed for hands-free human-machine communication in the mid-1990s after Bell Labs became a unit of Lucent Technologies. A video clip will be played to demonstrate the accomplishement.
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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.
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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.
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NASA Astrophysics Data System (ADS)
Wang, Guoqing; Bu, Tong; Zako, Tamotsu; Watanabe-Tamaki, Ryoko; Tanaka, Takuo; Maeda, Mizuo
2017-09-01
Due to the potential of gold nanoparticle (AuNP)-based trace analysis, the discrimination of small AuNP clusters with different assembling stoichiometry is a subject of fundamental and technological importance. Here we prepare oligomerized AuNPs with controlled stoichiometry through DNA-directed assembly, and demonstrate that AuNP monomers, dimers and trimers can be clearly distinguished using dark field microscopy (DFM). The scattering intensity for of AuNP structures with stoichiometry ranging from 1 to 3 agrees well with our theoretical calculations. This study demonstrates the potential of utilizing the DFM approach in ultra-sensitive detection as well as the use of DNA-directed assembly for plasmonic nano-architectures.
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The BetaCage, an ultra-sensitive screener for surface contamination
DOE Office of Scientific and Technical Information (OSTI.GOV)
Bunker, R.; Bowles, M. A.; Schnee, R. W.
Material screening for identifying low-energy electron emitters and alpha-decaying isotopes is now a prerequisite for rare-event searches (e.g., dark-matter direct detection and neutrinoless double-beta decay) for which surface radiocon-tamination has become an increasingly important background. The BetaCage, a gaseous neon time-projection chamber, is a proposed ultra-sensitive (and nondestructive) screener for alpha-and beta-emitting surface contaminants to which existing screening facilities are insufficiently sensitive. Sensitivity goals are 0.1 betas keV{sup −1} m{sup −2} day{sup −1} and 0.1 alphas m{sup −2} day{sup −1}, with the former limited by Compton scattering of photons in the screening samples and (thanks to tracking) the latter expectedmore » to be signal-limited; radioassays and simulations indicate backgrounds from detector materials and radon daughters should be subdominant. We report on details of the background simulations and detector design that provide the discrimination, shielding, and radiopurity necessary to reach our sensitivity goals for a chamber with a 95 × 95 cm{sup 2} sample area positioned below a 40 cm drift region and monitored by crisscrossed anode and cathode planes consisting of 151 wires each.« less
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Passaes, Caroline Pereira Bittencourt; Bruel, Timothée; Decalf, Jérémie; David, Annie; Angin, Mathieu; Monceaux, Valerie; Muller-Trutwin, Michaela; Noel, Nicolas; Bourdic, Katia; Lambotte, Olivier; Albert, Matthew L; Duffy, Darragh; Schwartz, Olivier; Sáez-Cirión, Asier
2017-03-15
The existence of HIV reservoirs in infected individuals under combined antiretroviral therapy (cART) represents a major obstacle toward cure. Viral reservoirs are assessed by quantification of HIV nucleic acids, a method which does not discriminate between infectious and defective viruses, or by viral outgrowth assays, which require large numbers of cells and long-term cultures. Here, we used an ultrasensitive p24 digital assay, which we report to be 1,000-fold more sensitive than classical enzyme-linked immunosorbent assays (ELISAs) in the quantification of HIV-1 Gag p24 production in samples from HIV-infected individuals. Results from ultrasensitive p24 assays were compared to those from conventional viral RNA reverse transcription-quantitative PCR (RT-qPCR)-based assays and from outgrowth assay readout by flow cytometry. Using serial dilutions and flow-based single-cell sorting, we show that viral proteins produced by a single infected cell can be detected by the ultrasensitive p24 assay. This unique sensitivity allowed the early (as soon as day 1 in 43% of cases) and more efficient detection and quantification of p24 in phytohemagglutinin-L (PHA)-stimulated CD4 + T cells from individuals under effective cART. When seven different classes of latency reversal agents (LRA) in resting CD4 + T cells from HIV-infected individuals were tested, the ultrasensitive p24 assay revealed differences in the extent of HIV reactivation. Of note, HIV RNA production was infrequently accompanied by p24 protein production (19%). Among the drugs tested, prostratin showed a superior capacity in inducing viral protein production. In summary, the ultrasensitive p24 assay allows the detection and quantification of p24 produced by single infected CD4 + T cells and provides a unique tool to assess early reactivation of infectious virus from reservoirs in HIV-infected individuals. IMPORTANCE The persistence of HIV reservoirs in infected individuals under effective antiretroviral treatment represents a major obstacle toward cure. Different methods to estimate HIV reservoirs exist, but there is currently no optimal assay to measure HIV reservoirs in HIV eradication interventions. In the present study, we report an ultrasensitive digital ELISA platform for quantification of the HIV-1 protein p24. This method was employed to assess the early reactivation of infectious virus from reservoirs in HIV-1-infected individuals. We found that viral proteins produced by a single infected cell can be detected by an ultrasensitive p24 assay. This unprecedented resolution showed major advantages in comparison to other techniques currently used to assess viral replication in reactivation studies. In addition, such a highly sensitive assay allows discrimination of drug-induced reactivation of productive HIV based on protein expression. The present study heralds new opportunities to evaluate the HIV reservoir and the efficacy of drugs used to target it. Copyright © 2017 American Society for Microbiology.
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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…
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Launcher Dynamic Data Acquisition
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
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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.
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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,
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Wang, Hongzhi; Wang, Yu; Liu, Su; Yu, Jinghua; Xu, Wei; Guo, Yuna; Huang, Jiadong
2015-05-14
A novel electrochemical aptasensor for ultrasensitive detection of antibiotics by combining polymerase-assisted target recycling amplification with strand displacement amplification with the help of polymerase and nicking endonuclease has been reported. This work is the first time that target-aptamer binding triggered quadratic recycling amplification has been utilized for electrochemical detection of antibiotics.
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Yin, Honghong; Kuang, Hua; Liu, Liqiang; Xu, Liguang; Ma, Wei; Wang, Libing; Xu, Chuanlai
2014-04-09
A novel biosensor for ultrasensitive detection of copper (Cu(2+)) was established based on the assembly of magnetic nanoparticles induced by the Cu(2+)-dependent ligation DNAzyme. With a low limit of detection of 2.8 nM and high specificity, this method has the potential to serve as a general platform for the detection of heavy metal ions.
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Jayakumar, Kumarasamy; Camarada, María Belén; Dharuman, Venkataraman; Ju, Huangxian; Dey, Ramendra Sundar; Wen, Yangping
2018-02-01
Correction for 'One-step coelectrodeposition-assisted layer-by-layer assembly of gold nanoparticles and reduced graphene oxide and its self-healing three-dimensional nanohybrid for an ultrasensitive DNA sensor' by Jayakumar Kumarasamy, et al., Nanoscale, 2018, DOI: 10.1039/c7nr06952a.
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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…
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Active noise canceling system for mechanically cooled germanium radiation detectors
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.
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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…
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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...
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Amendola, Alessandra; Bloisi, Maria; Marsella, Patrizia; Sabatini, Rosella; Bibbò, Angela; Angeletti, Claudio; Capobianchi, Maria Rosaria
2011-09-01
Numerous studies investigating clinical significance of HIV-1 minimal residual viremia (MRV) suggest potential utility of assays more sensitive than those routinely used to monitor viral suppression. However currently available methods, based on different technologies, show great variation in detection limit and input plasma volume, and generally suffer from lack of standardization. In order to establish new tools suitable for routine quantification of minimal residual viremia in patients under virological suppression, some modifications were introduced into standard procedure of the Abbott RealTime HIV-1 assay leading to a "modified" and an "ultrasensitive" protocols. The following modifications were introduced: calibration curve extended towards low HIV-1 RNA concentration; 4 fold increased sample volume by concentrating starting material; reduced volume of internal control; adoption of "open-mode" software for quantification. Analytical performances were evaluated using the HIV-1 RNA Working Reagent 1 for NAT assays (NIBSC). Both tests were applied to clinical samples from virologically suppressed patients. The "modified" and the "ultrasensitive" configurations of the assay reached a limit of detection of 18.8 (95% CI: 11.1-51.0 cp/mL) and 4.8 cp/mL (95% CI: 2.6-9.1 cp/mL), respectively, with high precision and accuracy. In clinical samples from virologically suppressed patients, "modified" and "ultrasensitive" protocols allowed to detect and quantify HIV RNA in 12.7% and 46.6%, respectively, of samples resulted "not-detectable", and in 70.0% and 69.5%, respectively, of samples "detected <40 cp/mL" in the standard assay. The "modified" and "ultrasensitive" assays are precise and accurate, and easily adoptable in routine diagnostic laboratories for measuring MRV. Copyright © 2011 Elsevier B.V. All rights reserved.
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Dyer, Gregory Conrad; Shaner, Eric A.; Reno, John L.; Aizin, Gregory
2015-08-11
A tunable plasmonic crystal comprises several periods in a two-dimensional electron or hole gas plasmonic medium that is both extremely subwavelength (.about..lamda./100) and tunable through the application of voltages to metal electrodes. Tuning of the plasmonic crystal band edges can be realized in materials such as semiconductors and graphene to actively control the plasmonic crystal dispersion in the terahertz and infrared spectral regions. The tunable plasmonic crystal provides a useful degree of freedom for applications in slow light devices, voltage-tunable waveguides, filters, ultra-sensitive direct and heterodyne THz detectors, and THz oscillators.
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A Comparative Study of a 1/4-Scale Gulfstream G550 Aircraft Nose Gear Model
NASA Technical Reports Server (NTRS)
Khorrami, Mehdi R.; Neuhart, Dan H.; Zawodny, Nikolas S.; Liu, Fei; Yardibi, Tarik; Cattafesta, Louis; Van de Ven, Thomas
2009-01-01
A series of fluid dynamic and aeroacoustic wind tunnel experiments are performed at the University of Florida Aeroacoustic Flow Facility and the NASA-Langley Basic Aerodynamic Research Tunnel Facility on a high-fidelity -scale model of Gulfstream G550 aircraft nose gear. The primary objectives of this study are to obtain a comprehensive aeroacoustic dataset for a nose landing gear and to provide a clearer understanding of landing gear contributions to overall airframe noise of commercial aircraft during landing configurations. Data measurement and analysis consist of mean and fluctuating model surface pressure, noise source localization maps using a large-aperture microphone directional array, and the determination of far field noise level spectra using a linear array of free field microphones. A total of 24 test runs are performed, consisting of four model assembly configurations, each of which is subjected to three test section speeds, in two different test section orientations. The different model assembly configurations vary in complexity from a fully-dressed to a partially-dressed geometry. The two model orientations provide flyover and sideline views from the perspective of a phased acoustic array for noise source localization via beamforming. Results show that the torque arm section of the model exhibits the highest rms pressures for all model configurations, which is also evidenced in the sideline view noise source maps for the partially-dressed model geometries. Analysis of acoustic spectra data from the linear array microphones shows a slight decrease in sound pressure levels at mid to high frequencies for the partially-dressed cavity open model configuration. In addition, far field sound pressure level spectra scale approximately with the 6th power of velocity and do not exhibit traditional Strouhal number scaling behavior.
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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.
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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.
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Kim, Joungmok; Yoon, Moon-Young
2010-06-01
Here, we review the cumulative efforts to develop rapid and ultrasensitive diagnostic systems, especially for the infectious agent, Bacillus anthracis, as a model system. This Minireview focuses on demonstrating the features of various probes for target molecule detection and recent methods of signal generation within the biosensors. Also, we discuss the possibility of using peptides as next-generation probe molecules.
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Guo, Zhiyong; Sha, Yuhong; Hu, Yufang; Wang, Sui
2016-03-28
A new-concept of an "in-electrode" Faraday cage-type electrochemiluminescence immunoassay (ECLIA) method for the ultrasensitive detection of neurotensin (NT) was reported with capture antibody (Ab1)-nanoFe3O4@graphene (GO) and detector antibody (Ab2)&N-(4-aminobutyl)-N-ethylisoluminol (ABEI)@GO, which led to about 1000-fold improvement in sensitivity by extending the Helmholtz plane (OHP) of the proposed electrode assembly effectively.
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Senthivel, Vivek Raj; Sturrock, Marc; Piedrafita, Gabriel; Isalan, Mark
2016-12-16
Nonlinear responses to signals are widespread natural phenomena that affect various cellular processes. Nonlinearity can be a desirable characteristic for engineering living organisms because it can lead to more switch-like responses, similar to those underlying the wiring in electronics. Steeper functions are described as ultrasensitive, and can be applied in synthetic biology by using various techniques including receptor decoys, multiple co-operative binding sites, and sequential positive feedbacks. Here, we explore the inherent non-linearity of a biological signaling system to identify functions that can potentially be exploited using cell genome engineering. For this, we performed genome-wide transcription profiling to identify genes with ultrasensitive response functions to Hepatocyte Growth Factor (HGF). We identified 3,527 genes that react to increasing concentrations of HGF, in Madin-Darby canine kidney (MDCK) cells, grown as cysts in 3D collagen cell culture. By fitting a generic Hill function to the dose-responses of these genes we obtained a measure of the ultrasensitivity of HGF-responsive genes, identifying a subset with higher apparent Hill coefficients (e.g. MMP1, TIMP1, SNORD75, SNORD86 and ERRFI1). The regulatory regions of these genes are potential candidates for future engineering of synthetic mammalian gene circuits requiring nonlinear responses to HGF signalling.
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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…
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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)
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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...
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Distance Learning Plan for the Defense Finance and Accounting Service (DFAS): A Study for the DBMU
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
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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...
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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.
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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...
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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...
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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).
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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).
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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.
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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.
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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.
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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.
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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.
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Plasmon-Based Colorimetric Nanosensors for Ultrasensitive Molecular Diagnostics.
Tang, Longhua; Li, Jinghong
2017-07-28
Colorimetric detection of target analytes with high specificity and sensitivity is of fundamental importance to clinical and personalized point-of-care diagnostics. Because of their extraordinary optical properties, plasmonic nanomaterials have been introduced into colorimetric sensing systems, which provide significantly improved sensitivity in various biosensing applications. Here we review the recent progress on these plasmonic nanoparticles-based colorimetric nanosensors for ultrasensitive molecular diagnostics. According to their different colorimetric signal generation mechanisms, these plasmonic nanosensors are classified into two categories: (1) interparticle distance-dependent colorimetric assay based on target-induced forming cross-linking assembly/aggregate of plasmonic nanoparticles; and (2) size/morphology-dependent colorimetric assay by target-controlled growth/etching of the plasmonic nanoparticles. The sensing fundamentals and cutting-edge applications will be provided for each of them, particularly focusing on signal generation and/or amplification mechanisms that realize ultrasensitive molecular detection. Finally, we also discuss the challenge and give our future perspective in this emerging field.
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Ultrasensitive ROS-Responsive Coassemblies of Tellurium-Containing Molecules and Phospholipids.
Wang, Lu; Fan, Fuqiang; Cao, Wei; Xu, Huaping
2015-07-29
Reactive oxygen species (ROS) play crucial roles in cell signaling and redox homeostasis and are strongly related to metabolic activities. The increase of the ROS concentration in organisms can result in several diseases, such as cardiovascular diseases and cancer. The concentration of ROS in biologically relevant conditions is typically as low as around tens of micromolars to 100 μM H2O2, which makes it necessary to develop ultrasensitive ROS-responsive systems. A general approach is reported here to fabricate an ultrasensitive ROS-responsive system via coassembly between tellurium-containing molecules and phospholipids, combining the ROS-responsiveness of tellurium and the biocompatibility of phospholipids. By using dynamic light scattering, transmission electron microscopy, scanning electron microscopy, and NMR spectra, coassembly behaviors and the responsiveness of the coassemblies have been investigated. These coassemblies can respond to 100 μM H2O2, which is a biologically relevant ROS concentration, and demonstrate reversible redox properties.
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Ultrasensitive sensing with three-dimensional terahertz metamaterial absorber
NASA Astrophysics Data System (ADS)
Tan, Siyu; Yan, Fengping; Wang, Wei; Zhou, Hong; Hou, Yafei
2018-05-01
Planar metasurfaces and metamaterial absorbers have shown great promise for label-free sensing applications at microwaves, optical and terahertz frequencies. The realization of high-quality-factor resonance in these structures is of significant interest to enhance the sensing sensitivities to detect minute frequency shifts. We propose and demonstrate in this manuscript an ultrasensitive terahertz metamaterial absorber sensor based on a three-dimensional split ring resonator absorber with a high quality factor of 60.09. The sensing performance of the proposed absorber sensor was systematically investigated through detailed numerical calculations and a maximum refractive index sensitivity of 34.40% RIU‑1 was obtained. Furthermore, the absorber sensor can maintain a high sensitivity for a wide range of incidence angles up to 60° under TM polarization incidence. These findings would improve the design flexibility of the absorber sensors and further open up new avenues to achieve ultrasensitive sensing in the terahertz regime.
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Evaluation of a new ultrasensitive assay for cardiac troponin I.
Casals, Gregori; Filella, Xavier; Bedini, Josep Lluis
2007-12-01
We evaluated the analytical and clinical performance of a new ultrasensitive cardiac troponin I assay (cTnI) on the ADVIA Centaur system (TnI-Ultra). The evaluation included the determination of detection limit, within-assay and between-assay variation and comparison with two other non-ultrasensitive methods. Moreover, cTnI was determined in 120 patients with acute chest pain with three methods. To evaluate the ability of the new method to detect MI earlier, it was assayed in 8 MI patients who first tested negative then positive by the other methods. The detection limit was 0.009 microg/L and imprecision was <10% at all concentrations evaluated. In comparison with two other methods, 10% of the anginas diagnosed were recategorized to MI. The ADVIA Centaur TnI-Ultra assay presented high reproducibility and high sensitivity. The use of the recommended lower cutpoint (0.044 microg/L) implied an increased and earlier identification of MI.
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Modified graphene oxide sensors for ultra-sensitive detection of nitrate ions in water.
Ren, Wen; Mura, Stefania; Irudayaraj, Joseph M K
2015-10-01
Nitrate ions is a very common contaminant in drinking water and has a significant impact on the environment, necessitating routine monitoring. Due to its chemical and physical properties, it is hard to directly detect nitrate ions with high sensitivity in a simple and inexpensive manner. Herein with amino group modified graphene oxide (GO) as a sensing element, we show a direct and ultra-sensitive method to detect nitrate ions, at a lowest detected concentration of 5 nM in river water samples, much lower than the reported methods based on absorption spectroscopy. Furthermore, unlike the reported strategies based on absorption spectroscopy wherein the nitrate concentration is determined by monitoring an increase in aggregation of gold nanoparticles (GNPs), our method evaluates the concentration of nitrate ions based on reduction in aggregation of GNPs for monitoring in real samples. To improve sensitivity, several optimizations were performed, including the assessment of the amount of modified GO required, concentration of GNPs and incubation time. The detection methodology was characterized by zeta potential, TEM and SEM. Our results indicate that an enrichment of modified GO with nitrate ions contributed to excellent sensitivity and the entire detection procedure could be completed within 75 min with only 20 μl of sample. This simple and rapid methodology was applied to monitor nitrate ions in real samples with excellent sensitivity and minimum pretreatment. The proposed approach paves the way for a novel means to detect anions in real samples and highlights the potential of GO based detection strategy for water quality monitoring. Copyright © 2015 Elsevier B.V. All rights reserved.
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Zhang, Peng; Liu, Hui; Ma, Suzhen; Men, Shuai; Li, Qingzhou; Yang, Xin; Wang, Hongning; Zhang, Anyun
2016-06-15
The harm of Salmonella enteritidis (S. enteritidis ) to public health mainly by contaminating fresh food and water emphasizes the urgent need for rapid detection techniques to help control the spread of the pathogen. In this assay, an newly designed capture probe complex that contained specific S. enteritidis-aptamer and hybridized signal target sequence was used for viable S. enteritidis recognition directly. In the presence of the target S. enteritidis, single-stranded target sequences were liberated and initiated the replication-cleavage reaction, producing numerous G-quadruplex structures with a linker on the 3'-end. And then, the sensing system took innovative advantage of quadratic linker-induced strand-displacement for the first time to release target sequence in succession, leading to the cyclic reuse of the target sequences and cascade signal amplification, thereby achieving the successive production of G-quadruplex structures. The fluorescent dye, N-Methyl mesoporphyrin IX, binded to these G-quadruplex structures and generated significantly enhanced fluorescent signals to achieve highly sensitive detection of S. enteritidis down to 60 CFU/mL with a linear range from 10(2) to 10(7)CFU/mL. By coupling the cascade two-stage target sequences-recyclable toehold strand-displacement with aptamer-based target recognition successfully, it is the first report on a novel non-label, modification-free and DNA extraction-free ultrasensitive fluorescence biosensor for detecting viable S. enteritidis directly, which can discriminate from dead S. enteritidis. Copyright © 2016 Elsevier B.V. All rights reserved.
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Effects of Orientation and Weatherproofing on the Detection of Bat Echolocation Calls
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 (...
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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…
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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…
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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 ...
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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 ...
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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 ...
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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…
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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.
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Measuring the Seismic and Acoustic Time of Flight - Lessons in Earthquakes and Thunder
NASA Astrophysics Data System (ADS)
Leeman, J.; Ammon, C. J.
2016-12-01
When teaching the fundamentals of waves and wave propagation, students must appreciate and understand that different waves travel through different materials at different speeds. We describe a simple experiment to explore acoustic wave propagation through the ground and the air and how to use those observations to locate the source of the waves. The experiment that can be performed with a geophone, a microphone, and an oscilloscope. For this activity, students will strike a metal plate, equipped with a vibration trigger, with a hammer. The blow triggers an oscilloscope to begin recording data in a "single-shot" mode. The two channels of the oscilloscope record the output of the microphone (measuring the energy of sound waves in the air) and the geophone (measuring the seismic wave energy in the ground). Seismic waves reach the geophone earlier than the sound waves since they travel at approximately ten times the speed. Students can measure the travel time on the oscilloscope, or using data downloaded to a computer. With measurements of the travel time and distance to the hammer, students can calculate the velocity of each wave. Then, the hammer can be used at other distances from the sensors and by measuring the difference in arrival time of the two waves, students can estimate the distance to the source which they check by direct measurement. This exercise can be directly connected to common observations such as seeing lighting before hearing thunder. The activity also connects directly to concepts related to earthquake location. We describe pedagogical materials, including experiment instructions, videos and data for those who do not have access to the equipment, and simple exercise suggestions for classroom activities.
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Noise generated by flow through large butterfly valves
NASA Technical Reports Server (NTRS)
Huff, Ronald G.
1987-01-01
A large butterfly valve (1.37 m diam) was acoustically tested to measure the noise generated and propagating in both the upstream and downstream directions. The experimental investigation used wall mounted pressure transducers to measure the fluctuating component of the pipe static pressure upstream and downstream of the valve. Microphones upstream of the pipe inlet and located in a plenum were used to measure the noise radiated from the valve in the upstream direction. Comparison of the wall pressure downstream of the valve to a prediction were made. Reasonable agreement was obtained with the valve operating at a choked condition. The noise upstream of the valve is 30 dB less than that measured downstream.
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Zainabadi, Kayvan; Adams, Matthew; Han, Zay Yar; Lwin, Hnin Wai; Han, Kay Thwe; Ouattara, Amed; Thura, Si; Plowe, Christopher V; Nyunt, Myaing M
2017-09-18
Greater Mekong Subregion countries are committed to eliminating Plasmodium falciparum malaria by 2025. Current elimination interventions target infections at parasite densities that can be detected by standard microscopy or rapid diagnostic tests (RDTs). More sensitive detection methods have been developed to detect lower density "asymptomatic" infections that may represent an important transmission reservoir. These ultrasensitive polymerase chain reaction (usPCR) tests have been used to identify target populations for mass drug administration (MDA). To date, malaria usPCR tests have used either venous or capillary blood sampling, which entails complex sample collection, processing and shipping requirements. An ultrasensitive method performed on standard dried blood spots (DBS) would greatly facilitate the molecular surveillance studies needed for targeting elimination interventions. A highly sensitive method for detecting Plasmodium falciparum and P. vivax 18S ribosomal RNA from DBS was developed by empirically optimizing nucleic acid extraction conditions. The limit of detection (LoD) was determined using spiked DBS samples that were dried and stored under simulated field conditions. Further, to assess its utility for routine molecular surveillance, two cross-sectional surveys were performed in Myanmar during the wet and dry seasons. The lower LoD of the DBS-based ultrasensitive assay was 20 parasites/mL for DBS collected on Whatman 3MM filter paper and 23 parasites/mL for Whatman 903 Protein Saver cards-equivalent to 1 parasite per 50 µL DBS. This is about 5000-fold more sensitive than standard RDTs and similar to the LoD of ≤16-22 parasites/mL reported for other ultrasensitive methods based on whole blood. In two cross-sectional surveys in Myanmar, nearly identical prevalence estimates were obtained from contemporaneous DBS samples and capillary blood samples collected during the wet and dry season. The DBS-based ultrasensitive method described in this study shows equal sensitivity as previously described methods based on whole blood, both in its limit of detection and prevalence estimates in two field surveys. The reduced cost and complexity of this method will allow for the scale-up of surveillance studies to target MDA and other malaria elimination interventions, and help lead to a better understanding of the epidemiology of low-density malaria infections.
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The Ultrasensitivity of Living Polymers
NASA Astrophysics Data System (ADS)
O'Shaughnessy, Ben; Vavylonis, Dimitrios
2003-03-01
Synthetic and biological living polymers are self-assembling chains whose chain length distributions (CLDs) are dynamic. We show these dynamics are ultrasensitive: Even a small perturbation (e.g., temperature jump) nonlinearly distorts the CLD, eliminating or massively augmenting short chains. The origin is fast relaxation of mass variables (mean chain length, monomer concentration) which perturbs CLD shape variables before these can relax via slow chain growth rate fluctuations. Viscosity relaxation predictions agree with experiments on the best-studied synthetic system, α-methylstyrene.
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Ultrasensitive, Biocompatible, Self-Calibrating, Multiparametric Temperature Sensors.
Zhao, Haiguang; Vomiero, Alberto; Rosei, Federico
2015-11-18
Core-shell quantum dots serve as self-calibrating, ultrasensitive, multiparametric, near-infrared, and biocompatible temperature sensors. They allow temperature measurement with nanometer accuracy in the range 150-373 K, the broadest ever recorded for a nanothermometer, with sensitivities among the highest ever reported, which makes them essentially unique in the panorama of biocompatible nanothermometers with potential for in vivo biological thermal imaging and/or thermoablative therapy. © 2015 WILEY-VCH Verlag GmbH & Co. KGaA, Weinheim.
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Zhao, Hui; Wang, Yong-Sheng; Tang, Xian; Zhou, Bin; Xue, Jin-Hua; Liu, Hui; Liu, Shan-Du; Cao, Jin-Xiu; Li, Ming-Hui; Chen, Si-Han
2015-08-05
We report on an enzyme-free and label-free strategy for the ultrasensitive determination of adenosine. A novel multipurpose adenosine aptamer (MAAP) is designed, which serves as an effective target recognition probe and a capture probe for malachite green. In the presence of adenosine, the conformation of the MAAP is converted from a hairpin structure to a G-quadruplex. Upon addition of malachite green into this solution, a noticeable enhancement of resonance light scattering was observed. The signal response is directly proportional to the concentration of adenosine ranging from 75 pM to 2.2 nM with a detection limit of 23 pM, which was 100-10,000 folds lower than those obtained by previous reported methods. Moreover, this strategy has been applied successfully for detecting adenosine in human urine and blood samples, further proving its reliability. The mechanism of adenosine inducing MAAP to form a G-quadruplex was demonstrated by a series of control experiments. Such a MAAP probe can also be used to other strategies such as fluorescence or spectrophotometric ones. We suppose that this strategy can be expanded to develop a universal analytical platform for various target molecules in the biomedical field and clinical diagnosis. Copyright © 2015 Elsevier B.V. All rights reserved.
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Lee, I M; Bartoszyk, I M; Gundersen, D E; Mogen, B; Davis, R E
1997-07-01
Oligonucleotide primers derived from sequences of the 16S rRNA gene (CMR16F1, CMR16R1, CMR16F2, and CMR16R2) and insertion element IS1121 of Clavibacter michiganensis subsp. sepedonicus (CMSIF1, CMSIR1, CMSIF2, and CMISR2) were used in nested PCR to detect the potato ring rot bacterium C. michiganensis subsp. sepedonicus. Nested PCR with primer pair CMSIF1-CMSIR1 followed by primer pair CMSIF2-CMSIR2 specifically detected C. michiganensis subsp. sepedonicus, while nested PCR with CMR16F1-CMR16R1 followed by CMR16F2-CMR16R2 detected C. michiganensis subsp. sepedonicus and the other C. michiganensis subspecies. In the latter case, C. michiganensis subsp. sepedonicus can be differentiated from the other subspecies by restriction fragment length polymorphism (RFLP) analyses of the nested PCR products (16S rDNA sequences). The nested PCR assays developed in this work allow ultrasensitive detection of very low titers of C. michiganensis subsp. sepedonicus which may be present in symptomiess potato plants or tubers and which cannot be readily detected by direct PCR (single PCR amplification). RFLP analysis of PCR products provides for an unambiguous confirmation of the identify of C. michiganensis subsp. sepedonicus.
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Comparability of AMH levels among commercially available immunoassays
Su, H. Irene; Sammel, Mary D.; Homer, Michael V.; Bui, Kim; Haunschild, Carolyn; Stanczyk, Frank Z.
2015-01-01
Objective To compare AMH levels among three commercially available AMH immunoassays (AMH Gen II, Beckman Coulter; Ultrasensitive AMH, AnshLab; picoAMH, AnshLab) Design Cross-sectional Setting Academic reproductive endocrinology program Patients 90 newly diagnosed breast cancer patients prior to cancer treatment Interventions None Outcome 1) proportion of detectable AMH levels by immunoassay, 2) comparability among assays Results At a mean age of 38.1, the median (interquartile range) for AMH levels for the cohort were 0.92 [1.35] ng/mL for the Gen II assay, 1.68 [2.30] ng/mL for the Ultrasensitive and 1.5 [2.41] ng/mL for the picoAMH assays. Significantly higher proportions of detectable AMH levels were observed with the picoAMH kit (97%) compared to both Gen II (84%) and Ultrasensitive (92%) assays. Although AMH results were highly correlated among assays (r=0.92–0.99), Gen II AMH levels were consistently lower than both Ultrasensitive and picoAMH levels. Moreover, as AMH levels increased, the magnitude of difference grew larger between Gen II and each of the other two assays. Conclusions Measurement of AMH levels with the picoAMH kit maximized detection at very low levels, particularly in contrast to the Gen II kit. Conversion of AMH levels from different immunoassays using regression equations is potentially highly inaccurate. PMID:24726216
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Zou, Bin; Guo, Yunlong; Shen, Nannan; Xiao, Anshan; Li, Mingjun; Zhu, Liang; Wan, Pengbo; Sun, Xiaoming
2017-12-19
Ultrasensitive room temperature real-time NO₂ sensors are highly desirable due to potential threats on environmental security and personal respiratory. Traditional NO₂ gas sensors with highly operated temperatures (200-600 °C) and limited reversibility are mainly constructed from semiconducting oxide-deposited ceramic tubes or inter-finger probes. Herein, we report the functionalized graphene network film sensors assembled on an electrospun three-dimensional (3D) nanonetwork skeleton for ultrasensitive NO₂ sensing. The functional 3D scaffold was prepared by electrospinning interconnected polyacrylonitrile (PAN) nanofibers onto a nylon window screen to provide a 3D nanonetwork skeleton. Then, the sulfophenyl-functionalized reduced graphene oxide (SFRGO) was assembled on the electrospun 3D nanonetwork skeleton to form SFRGO network films. The assembled functionalized graphene network film sensors exhibit excellent NO₂ sensing performance (10 ppb to 20 ppm) at room temperature, reliable reversibility, good selectivity, and better sensing cycle stability. These improvements can be ascribed to the functionalization of graphene with electron-withdrawing sulfophenyl groups, the high surface-to-volume ratio, and the effective sensing channels from SFRGO wrapping onto the interconnected 3D scaffold. The SFRGO network-sensing film has the advantages of simple preparation, low cost, good processability, and ultrasensitive NO₂ sensing, all advantages that can be utilized for potential integration into smart windows and wearable electronic devices for real-time household gas sensors.
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Garain, Samiran; Jana, Santanu; Sinha, Tridib Kumar; Mandal, Dipankar
2016-02-01
We report an efficient, low-cost in situ poled fabrication strategy to construct a large area, highly sensitive, flexible pressure sensor by electrospun Ce(3+) doped PVDF/graphene composite nanofibers. The entire device fabrication process is scalable and enabling to large-area integration. It can able to detect imparting pressure as low as 2 Pa with high level of sensitivity. Furthermore, Ce(3+)-doped PVDF/graphene nanofiber based ultrasensitive pressure sensors can also be used as an effective nanogenerator as it generating an output voltage of 11 V with a current density ∼6 nA/cm(2) upon repetitive application of mechanical stress that could lit up 10 blue light emitting diodes (LEDs) instantaneously. Furthermore, to use it in environmental random vibrations (such as wind flow, water fall, transportation of vehicles, etc.), nanogenerator is integrated with musical vibration that exhibits to power up three blue LEDs instantly that promises as an ultrasensitive acoustic nanogenerator (ANG). The superior sensing properties in conjunction with mechanical flexibility, integrability, and robustness of nanofibers enabled real-time monitoring of sound waves as well as detection of different type of musical vibrations. Thus, ANG promises to use as an ultrasensitive pressure sensor, mechanical energy harvester, and effective power source for portable electronic and wearable devices.
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Noise Source Visualization Using a Digital Voice Recorder and Low-Cost Sensors
Cho, Yong Thung
2018-01-01
Accurate sound visualization of noise sources is required for optimal noise control. Typically, noise measurement systems require microphones, an analog-digital converter, cables, a data acquisition system, etc., which may not be affordable for potential users. Also, many such systems are not highly portable and may not be convenient for travel. Handheld personal electronic devices such as smartphones and digital voice recorders with relatively lower costs and higher performance have become widely available recently. Even though such devices are highly portable, directly implementing them for noise measurement may lead to erroneous results since such equipment was originally designed for voice recording. In this study, external microphones were connected to a digital voice recorder to conduct measurements and the input received was processed for noise visualization. In this way, a low cost, compact sound visualization system was designed and introduced to visualize two actual noise sources for verification with different characteristics: an enclosed loud speaker and a small air compressor. Reasonable accuracy of noise visualization for these two sources was shown over a relatively wide frequency range. This very affordable and compact sound visualization system can be used for many actual noise visualization applications in addition to educational purposes. PMID:29614038
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All-optical non-mechanical fiber-coupled sensor for liquid- and airborne sound detection.
NASA Astrophysics Data System (ADS)
Rohringer, Wolfgang; Preißer, Stefan; Fischer, Balthasar
2017-04-01
Most fiber-optic devices for pressure, strain or temperature measurements are based on measuring the mechanical deformation of the optical fiber by various techniques. While excellently suited for detecting strain, pressure or structure-borne sound, their sensitivity to liquid- and airborne sound is so far not comparable with conventional capacitive microphones or piezoelectric hydrophones. Here, we present an all-optical acoustic sensor which relies on the detection of pressure-induced changes of the optical refractive index inside a rigid, millimeter-sized, fiber-coupled Fabry-Pérot interferometer (FPI). No mechanically movable or deformable parts take part in the signal transduction chain. Therefore, due to the absence of mechanical resonances, this sensing principle allows for high sensitivity as well as a flat frequency response over an extraordinary measurement bandwidth. As a fiber-coupled device, it can be integrated easily into already available distributed fiber-optic networks for geophysical sensing. We present characterization measurements demonstrating the sensitivity, frequency response and directivity of the device for sound and ultrasound detection in air and water. We show that low-frequency temperature and pressure drifts can be recorded in addition to acoustic sensing. Finally, selected application tests of the laser-based hydrophone and microphone implementation are presented.
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Spectral Separation of the Turbofan Engine Coherent Combustion Noise Component
NASA Technical Reports Server (NTRS)
Miles, Jeffrey Hilton
2008-01-01
The core noise components of a dual spool turbofan engine (Honeywell TECH977) were separated by the use of a coherence function. A source location technique based on adjusting the time delay between the combustor pressure sensor signal and the far-field microphone signal to maximize the coherence and remove as much variation of the phase angle with frequency as possible was used. While adjusting the time delay to maximize the coherence and minimize the cross spectrum phase angle variation with frequency, the discovery was made that for the 130 microphone a 90.027 ms time shift worked best for the frequency band from 0 to 200 Hz while a 86.975 ms time shift worked best for the frequency band from 200 to 400 Hz. Since the 0 to 200 Hz band signal took more time to travel the same distance, it is slower than the 200 to 400 Hz band signal. This suggests the 0 to 200 Hz coherent cross spectral density band is partly due to indirect combustion noise attributed to hot spots interacting with the turbine. The signal in the 200 to 400 Hz frequency band is attributed mostly to direct combustion noise.
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Noise measurement flight test: Data-analyses Aerospatiale SA-365N Dauphin 2 helicopter
NASA Astrophysics Data System (ADS)
Newman, J. S.; Rickely, E. J.; Daboin, S. A.; Beattie, K. R.
1984-04-01
This report documents the results of a Federal Aviation Administration (FAA) noise measurement flight test program with the Dauphin twin-jet helicopter. The report contains documentary sections describing the acoustical characteristics of the subject helicopter and provides analyses and discussions addressing topics ranging from acoustical propagation to environmental impact of helicopter noise. This report is the second in a series of seven documenting the FAA helicopter noise measurement program conducted at Dulles International Airport during the summer of 1983. The Dauphin test program involved the acquisition of detailed acoustical, position and meteorological data. This test program was designed to address a series of objectives including: (1) acquisition of acoustical data for use in assessing heliport environment impact, (2) documentation of directivity characteristics for static operation of helicopters, (3) establishment of ground-to-ground and air-to-ground acoustical propagation relationships for helicopters, (4) determination of noise event duration influences on energy dose acoustical metrics, (5) examination of the differences between noise measured by a surface mounted microphone and a microphone mounted at a height of four feet (1.2 meters), and (6) documentation of noise levels acquired using international helicopter noise certification test procedures.
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Microphone Handling Noise: Measurements of Perceptual Threshold and Effects on Audio Quality
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
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Robust speaker's location detection in a vehicle environment using GMM models.
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.
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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.
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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.
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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...
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Adaptive Noise Reduction Techniques for Airborne Acoustic Sensors
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
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A New Kind of Laser Microphone for Photoacoustic Applications
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
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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.
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The frequency range of TMJ sounds.
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.
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Application of the remote microphone method to active noise control in a mobile phone.
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.
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Wind noise measured at the ground surface.
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.
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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.
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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.
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Implementation of a Virtual Microphone Array to Obtain High Resolution Acoustic Images
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
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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.
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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).
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Elbeik, Tarek; Charlebois, Edwin; Nassos, Patricia; Kahn, James; Hecht, Frederick M.; Yajko, David; Ng, Valerie; Hadley, Keith
2000-01-01
Quantification of human immunodeficiency virus type 1 (HIV-1) RNA as a measure of viral load has greatly improved the monitoring of therapies for infected individuals. With the significant reductions in viral load now observed in individuals treated with highly active anti-retroviral therapy (HAART), viral load assays have been adapted to achieve greater sensitivity. Two commercially available ultrasensitive assays, the Bayer Quantiplex HIV-1 bDNA version 3.0 (bDNA 3.0) assay and the Roche Amplicor HIV-1 Monitor Ultrasensitive version 1.5 (Amplicor 1.5) assay, are now being used to monitor HIV-1-infected individuals. Both of these ultrasensitive assays have a reported lower limit of 50 HIV-1 RNA copies/ml and were developed from corresponding older generation assays with lower limits of 400 to 500 copies/ml. However, the comparability of viral load data generated by these ultrasensitive assays and the relative costs of labor, disposables, and biohazardous wastes were not determined in most cases. In this study, we used matched clinical plasma samples to compare the quantification of the newer bDNA 3.0 assay with that of the older bDNA 2.0 assay and to compare the quantification and costs of the bDNA 3.0 assay and the Amplicor 1.5 assay. We found that quantification by the bDNA 3.0 assay was approximately twofold higher than that by the bDNA 2.0 assay and was highly correlated to that by the Amplicor 1.5 assay. Moreover, cost analysis based on labor, disposables, and biohazardous wastes showed significant savings with the bDNA 3.0 assay as compared to the costs of the Amplicor 1.5 assay. PMID:10699005
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Elbeik, T; Charlebois, E; Nassos, P; Kahn, J; Hecht, F M; Yajko, D; Ng, V; Hadley, K
2000-03-01
Quantification of human immunodeficiency virus type 1 (HIV-1) RNA as a measure of viral load has greatly improved the monitoring of therapies for infected individuals. With the significant reductions in viral load now observed in individuals treated with highly active anti-retroviral therapy (HAART), viral load assays have been adapted to achieve greater sensitivity. Two commercially available ultrasensitive assays, the Bayer Quantiplex HIV-1 bDNA version 3.0 (bDNA 3.0) assay and the Roche Amplicor HIV-1 Monitor Ultrasensitive version 1.5 (Amplicor 1.5) assay, are now being used to monitor HIV-1-infected individuals. Both of these ultrasensitive assays have a reported lower limit of 50 HIV-1 RNA copies/ml and were developed from corresponding older generation assays with lower limits of 400 to 500 copies/ml. However, the comparability of viral load data generated by these ultrasensitive assays and the relative costs of labor, disposables, and biohazardous wastes were not determined in most cases. In this study, we used matched clinical plasma samples to compare the quantification of the newer bDNA 3.0 assay with that of the older bDNA 2.0 assay and to compare the quantification and costs of the bDNA 3.0 assay and the Amplicor 1.5 assay. We found that quantification by the bDNA 3.0 assay was approximately twofold higher than that by the bDNA 2.0 assay and was highly correlated to that by the Amplicor 1.5 assay. Moreover, cost analysis based on labor, disposables, and biohazardous wastes showed significant savings with the bDNA 3.0 assay as compared to the costs of the Amplicor 1.5 assay.
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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.
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Effects of Angle of Attack and Velocity on Trailing Edge Noise
NASA Technical Reports Server (NTRS)
Hutcheson, Florence V.; Brooks, Thomas F.
2006-01-01
Trailing edge (TE) noise measurements for a NACA 63-215 airfoil model are presented, providing benchmark experimental data for a cambered airfoil. The effects of flow Mach number and angle of attack of the airfoil model with different TE bluntnesses are shown. Far-field noise spectra and directivity are obtained using a directional microphone array. Standard and diagonal removal beamforming techniques are evaluated employing tailored weighting functions for quantitatively accounting for the distributed line character of TE noise. Diagonal removal processing is used for the primary database as it successfully removes noise contaminates. Some TE noise predictions are reported to help interpret the data, with respect to flow speed, angle of attack, and TE bluntness on spectral shape and peak levels. Important findings include the validation of a TE noise directivity function for different airfoil angles of attack and the demonstration of the importance of the directivity function s convective amplification terms.
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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.
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MEMS PolyMUMPS-Based Miniature Microphone for Directional Sound Sensing
2007-09-01
of the translating mode Phir=-atan((2*wr*er*w)/(wr^2-w^2));% Phase constant rocking Phit =-atan((2*wt*et*w)/(wt^2-w^2));% Phase constant translating...2.5e-6)+1 Yl(count)=8e6*(At*sin(w.*t(count)+ Phit ) + Ar*cos(w.*t(count)+Phir)); %left membrane displacement as a function of time in micrometers...Xl(count)=-(((.5)^2-Yl(count).^2).^.5); Yr(count)=8e6*(At*sin(w.*t(count)+ Phit ) - Ar*cos(w.*t(count)+Phir)); %right membrane displacement
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Burki, Umar; Straub, Volker
2017-01-01
Determining the concentration of oligonucleotide in biological samples such as tissue lysate and serum is essential for determining the biodistribution and pharmacokinetic profile, respectively. ELISA-based assays have shown far greater sensitivities compared to other methods such as HPLC and LC/MS. Here, we describe a novel ultrasensitive hybridization-based ELISA method for quantitating morpholino oligonucleotides in mouse tissue lysate and serum samples. The assay has a linear detection range of 5-250 pM (R2 > 0.99).
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Virus templated plasmonic nanoclusters with icosahedral symmetry via directed assembly
NASA Astrophysics Data System (ADS)
Ratna, Banahalli; Fontana, Jake; Dressick, Walter; Phelps, Jamie; Johnson, John; Sampson, Travian; Rendell, Ronald; Soto, Carissa
2015-03-01
Controlling the spatial and orientational order of plasmonic nanoparticles may lead to structures with novel electromagnetic properties and applications such as sub-wavelength imaging and ultra-sensitive chemical sensors. Here we report the directed assembly of three-dimensional, icosahedral plasmonic nanoclusters with resonances at visible wavelengths. We show using transmission electron microcopy and in situ dynamic light scattering the nanoclusters consist of twelve gold nanospheres attached to thiol groups at predefined locations on the surface of a genetically engineered cowpea mosaic virus with icosahedral symmetry. We measured the bulk absorbance from aqueous suspensions of nanoclusters and reproduced the major features of the spectrum using finite-element simulations. Furthermore, because the viruses are easily produced in gram quantities the directed assembly approach is capable of high-throughput, providing a strategy to realize large quantities for applications. NRL summer intern under the HBCU/MI Summer Research Program.
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NASA Astrophysics Data System (ADS)
Jers, Harald
2005-09-01
Studies of acoustical balance between singers within a choir by means of room acoustical measurements have shown that the directional sound propagation of the source is important. For this reason the directivity of female and male singers for different vowels has been measured in this investigation. Measurements of a pilot study and some first measurements in 1998 have been supplemented with new measurements and an enhanced setup. A special measurement setup with reference and recording microphones was used to collect the directivity data. A resolution of 10 deg for azimuth and elevation angle was obtained. The results will be shown in 3D spherical plots with frequency adjustments in semitones from 80 to 8000 Hz. The measurements are compared to an artificial singer's directivity, and the influence of a sheet music binder in front of a singer will be shown. The results give information on the directivity of singers and are relevant for the prediction of self-to-other-ratios that result from placement and formation aspects within a choir.
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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.
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Detection of Humans and Light Vehicles Using Acoustic-to-Seismic Coupling
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
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Toward Active Control of Noise from Hot Supersonic Jets
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
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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...
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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...
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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...
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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.
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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.
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Pichlo, Magdalena; Bungert-Plümke, Stefanie; Weyand, Ingo; Seifert, Reinhard; Bönigk, Wolfgang; Strünker, Timo; Kashikar, Nachiket Dilip; Goodwin, Normann; Müller, Astrid; Körschen, Heinz G.; Collienne, Ursel; Pelzer, Patric; Van, Qui; Enderlein, Jörg; Klemm, Clementine; Krause, Eberhard; Trötschel, Christian; Poetsch, Ansgar; Kremmer, Elisabeth
2014-01-01
Guanylyl cyclases (GCs), which synthesize the messenger cyclic guanosine 3′,5′-monophosphate, control several sensory functions, such as phototransduction, chemosensation, and thermosensation, in many species from worms to mammals. The GC chemoreceptor in sea urchin sperm can decode chemoattractant concentrations with single-molecule sensitivity. The molecular and cellular underpinnings of such ultrasensitivity are not known for any eukaryotic chemoreceptor. In this paper, we show that an exquisitely high density of 3 × 105 GC chemoreceptors and subnanomolar ligand affinity provide a high ligand-capture efficacy and render sperm perfect absorbers. The GC activity is terminated within 150 ms by dephosphorylation steps of the receptor, which provides a means for precise control of the GC lifetime and which reduces “molecule noise.” Compared with other ultrasensitive sensory systems, the 10-fold signal amplification by the GC receptor is surprisingly low. The hallmarks of this signaling mechanism provide a blueprint for chemical sensing in small compartments, such as olfactory cilia, insect antennae, or even synaptic boutons. PMID:25135936
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Self-Biased 215MHz Magnetoelectric NEMS Resonator for Ultra-Sensitive DC Magnetic Field Detection
NASA Astrophysics Data System (ADS)
Nan, Tianxiang; Hui, Yu; Rinaldi, Matteo; Sun, Nian X.
2013-06-01
High sensitivity magnetoelectric sensors with their electromechanical resonance frequencies < 200 kHz have been recently demonstrated using magnetostrictive/piezoelectric magnetoelectric heterostructures. In this work, we demonstrate a novel magnetoelectric nano-electromechanical systems (NEMS) resonator with an electromechanical resonance frequency of 215 MHz based on an AlN/(FeGaB/Al2O3) × 10 magnetoelectric heterostructure for detecting DC magnetic fields. This magnetoelectric NEMS resonator showed a high quality factor of 735, and strong magnetoelectric coupling with a large voltage tunable sensitivity. The admittance of the magnetoelectric NEMS resonator was very sensitive to DC magnetic fields at its electromechanical resonance, which led to a new detection mechanism for ultra-sensitive self-biased RF NEMS magnetoelectric sensor with a low limit of detection of DC magnetic fields of ~300 picoTelsa. The magnetic/piezoelectric heterostructure based RF NEMS magnetoelectric sensor is compact, power efficient and readily integrated with CMOS technology, which represents a new class of ultra-sensitive magnetometers for DC and low frequency AC magnetic fields.
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Flexible suspended gate organic thin-film transistors for ultra-sensitive pressure detection
NASA Astrophysics Data System (ADS)
Zang, Yaping; Zhang, Fengjiao; Huang, Dazhen; Gao, Xike; di, Chong-An; Zhu, Daoben
2015-03-01
The utilization of organic devices as pressure-sensing elements in artificial intelligence and healthcare applications represents a fascinating opportunity for the next-generation electronic products. To satisfy the critical requirements of these promising applications, the low-cost construction of large-area ultra-sensitive organic pressure devices with outstanding flexibility is highly desired. Here we present flexible suspended gate organic thin-film transistors (SGOTFTs) as a model platform that enables ultra-sensitive pressure detection. More importantly, the unique device geometry of SGOTFTs allows the fine-tuning of their sensitivity by the suspended gate. An unprecedented sensitivity of 192 kPa-1, a low limit-of-detection pressure of <0.5 Pa and a short response time of 10 ms were successfully realized, allowing the real-time detection of acoustic waves. These excellent sensing properties of SGOTFTs, together with their advantages of facile large-area fabrication and versatility in detecting various pressure signals, make SGOTFTs a powerful strategy for spatial pressure mapping in practical applications.
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Cheng, Wei; Zhang, Wei; Yan, Yurong; Shen, Bo; Zhu, Dan; Lei, Pinhua; Ding, Shijia
2014-12-15
A novel electrochemical biosensing strategy was developed for ultrasensitive and specific detection of target DNA using a cascade signal amplification based on molecular beacon (MB) mediated circular strand displacement (CSD), rolling circle amplification (RCA), biotin-strepavidin system, and enzymatic amplification. The target DNA hybridized with the loop portion of MB probe immobilized on the gold electrode and triggered the CSD, leading to multiple biotin-tagged DNA duplex. Furthermore, via biotin-streptavidin interaction, the RCA was implemented, producing long massive tandem-repeat DNA sequences for binding numerous biotinylated detection probes. This enabled an ultrasensitive electrochemical readout by further employing the streptavidin-alkaline phosphatase. The proposed biosensor showed very high sensitivity and selectivity with a dynamic response range from 1 fM to 100 pM. The proposed strategy could have the potential for applying in clinical molecular diagnostics and environmental monitoring. Copyright © 2014 Elsevier B.V. All rights reserved.
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Advances in ultrasensitive mass spectrometry of organic molecules.
Kandiah, Mathivathani; Urban, Pawel L
2013-06-21
Ultrasensitive mass spectrometric analysis of organic molecules is important for various branches of chemistry, and other fields including physics, earth and environmental sciences, archaeology, biomedicine, and materials science. It finds applications--as an enabling tool--in systems biology, biological imaging, clinical analysis, and forensics. Although there are a number of technical obstacles associated with the analysis of samples by mass spectrometry at ultratrace level (for example analyte losses during sample preparation, insufficient sensitivity, ion suppression), several noteworthy developments have been made over the years. They include: sensitive ion sources, loss-free interfaces, ion optics components, efficient mass analyzers and detectors, as well as "smart" sample preparation strategies. Some of the mass spectrometric methods published to date can achieve sensitivity which is by several orders of magnitude higher than that of alternative approaches. Femto- and attomole level limits of detection are nowadays common, while zepto- and yoctomole level limits of detection have also been reported. We envision that the ultrasensitive mass spectrometric assays will soon contribute to new discoveries in bioscience and other areas.
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Luminescent Quantum Dots as Ultrasensitive Biological Labels
NASA Astrophysics Data System (ADS)
Nie, Shuming
2000-03-01
Highly luminescent semiconductor quantum dots have been covalently coupled to biological molecules for use in ultrasensitive biological detection. This new class of luminescent labels is considerably brighter and more resistant againt photobleaching in comparison with organic dyes. Quantum dots labeled with the protein transferrin undergo receptor-mediated endocytosis (RME) in cultured HeLa cells, and those dots that were conjugated to immunomolecules recognize specific antibodies or antigens. In addition, we show that DNA functionalized quantum dots can be used to target specific genes by hybridization. We expect that quantum dot bioconjugates will have a broad range of biological applications, such as ligand-receptor interactions, real-time monitoring of molecular trafficking inside living cells, multicolor fluorescence in-situ hybridization (FISH), high-sensitivity detection in miniaturized devices (e.g., DNA chips), and fluorescent tagging of combinatorial chemical libraries. A potential clinical application is the use of quantum dots for ultrasensitive viral RNA detection, in which as low as 100 copies of hepatitis C and HIV viruses per ml blood should be detected.
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Tunable signal processing in synthetic MAP kinase cascades.
O'Shaughnessy, Ellen C; Palani, Santhosh; Collins, James J; Sarkar, Casim A
2011-01-07
The flexibility of MAPK cascade responses enables regulation of a vast array of cell fate decisions, but elucidating the mechanisms underlying this plasticity is difficult in endogenous signaling networks. We constructed insulated mammalian MAPK cascades in yeast to explore how intrinsic and extrinsic perturbations affect the flexibility of these synthetic signaling modules. Contrary to biphasic dependence on scaffold concentration, we observe monotonic decreases in signal strength as scaffold concentration increases. We find that augmenting the concentration of sequential kinases can enhance ultrasensitivity and lower the activation threshold. Further, integrating negative regulation and concentration variation can decouple ultrasensitivity and threshold from the strength of the response. Computational analyses show that cascading can generate ultrasensitivity and that natural cascades with different kinase concentrations are innately biased toward their distinct activation profiles. This work demonstrates that tunable signal processing is inherent to minimal MAPK modules and elucidates principles for rational design of synthetic signaling systems. Copyright © 2011 Elsevier Inc. All rights reserved.
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Optical fiber LPG biosensor integrated microfluidic chip for ultrasensitive glucose detection
Yin, Ming-jie; Huang, Bobo; Gao, Shaorui; Zhang, A. Ping; Ye, Xuesong
2016-01-01
An optical fiber sensor integrated microfluidic chip is presented for ultrasensitive detection of glucose. A long-period grating (LPG) inscribed in a small-diameter single-mode fiber (SDSMF) is employed as an optical refractive-index (RI) sensor. With the layer-by-layer (LbL) self-assembly technique, poly (ethylenimine) (PEI) and poly (acrylic acid) (PAA) multilayer film is deposited on the SDSMF-LPG sensor for both supporting and signal enhancement, and then a glucose oxidase (GOD) layer is immobilized on the outer layer for glucose sensing. A microfluidic chip for glucose detection is fabricated after embedding the SDSMF-LPG biosensor into the microchannel of the chip. Experimental results reveal that the SDSMF-LPG biosensor based on such a hybrid sensing film can ultrasensitively detect glucose concentration as low as 1 nM. After integration into the microfluidic chip, the detection range of the sensor is extended from 2 µM to 10 µM, and the response time is remarkablely shortened from 6 minutes to 70 seconds. PMID:27231643
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Flexible suspended gate organic thin-film transistors for ultra-sensitive pressure detection
Zang, Yaping; Zhang, Fengjiao; Huang, Dazhen; Gao, Xike; Di, Chong-an; Zhu, Daoben
2015-01-01
The utilization of organic devices as pressure-sensing elements in artificial intelligence and healthcare applications represents a fascinating opportunity for the next-generation electronic products. To satisfy the critical requirements of these promising applications, the low-cost construction of large-area ultra-sensitive organic pressure devices with outstanding flexibility is highly desired. Here we present flexible suspended gate organic thin-film transistors (SGOTFTs) as a model platform that enables ultra-sensitive pressure detection. More importantly, the unique device geometry of SGOTFTs allows the fine-tuning of their sensitivity by the suspended gate. An unprecedented sensitivity of 192 kPa−1, a low limit-of-detection pressure of <0.5 Pa and a short response time of 10 ms were successfully realized, allowing the real-time detection of acoustic waves. These excellent sensing properties of SGOTFTs, together with their advantages of facile large-area fabrication and versatility in detecting various pressure signals, make SGOTFTs a powerful strategy for spatial pressure mapping in practical applications. PMID:25872157
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Factors affecting the performance of large-aperture microphone arrays.
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.
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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.
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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.
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Anderson, Melinda C; Arehart, Kathryn H; Souza, Pamela E
2018-02-01
Current guidelines for adult hearing aid fittings recommend the use of a prescriptive fitting rationale with real-ear verification that considers the audiogram for the determination of frequency-specific gain and ratios for wide dynamic range compression. However, the guidelines lack recommendations for how other common signal-processing features (e.g., noise reduction, frequency lowering, directional microphones) should be considered during the provision of hearing aid fittings and fine-tunings for adult patients. The purpose of this survey was to identify how audiologists make clinical decisions regarding common signal-processing features for hearing aid provision in adults. An online survey was sent to audiologists across the United States. The 22 survey questions addressed four primary topics including demographics of the responding audiologists, factors affecting selection of hearing aid devices, the approaches used in the fitting of signal-processing features, and the strategies used in the fine-tuning of these features. A total of 251 audiologists who provide hearing aid fittings to adults completed the electronically distributed survey. The respondents worked in a variety of settings including private practice, physician offices, university clinics, and hospitals/medical centers. Data analysis was based on a qualitative analysis of the question responses. The survey results for each of the four topic areas (demographics, device selection, hearing aid fitting, and hearing aid fine-tuning) are summarized descriptively. Survey responses indicate that audiologists vary in the procedures they use in fitting and fine-tuning based on the specific feature, such that the approaches used for the fitting of frequency-specific gain differ from other types of features (i.e., compression time constants, frequency lowering parameters, noise reduction strength, directional microphones, feedback management). Audiologists commonly rely on prescriptive fitting formulas and probe microphone measures for the fitting of frequency-specific gain and rely on manufacturers' default settings and recommendations for both the initial fitting and the fine-tuning of signal-processing features other than frequency-specific gain. The survey results are consistent with a lack of published protocols and guidelines for fitting and adjusting signal-processing features beyond frequency-specific gain. To streamline current practice, a transparent evidence-based tool that enables clinicians to prescribe the setting of other features from individual patient characteristics would be desirable. American Academy of Audiology
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Comparability of antimüllerian hormone levels among commercially available immunoassays.
Su, H Irene; Sammel, Mary D; Homer, Michael V; Bui, Kim; Haunschild, Carolyn; Stanczyk, Frank Z
2014-06-01
To compare antimüllerian hormone (AMH) levels among three commercially available AMH immunoassays: AMH Gen II (Beckman Coulter), Ultrasensitive AMH (Ansh Labs), and picoAMH (Ansh Labs). Cross-sectional. Academic reproductive endocrinology program. 90 newly diagnosed breast cancer patients before cancer treatment. None. Proportion of detectable AMH levels by immunoassay, and comparability among assays. At a mean age of 38.1 years, the median (interquartile range) AMH level for the cohort was 0.92 [1.35] ng/mL for the Gen II assay, 1.68 [2.30] ng/mL for the Ultrasensitive assay, and 1.52 [2.41] ng/mL for the picoAMH assay. Significantly higher proportions of detectable AMH levels were observed with the picoAMH kit (97%) compared with both the Gen II (84%) and Ultrasensitive (92%) assays. Although the AMH results were highly correlated among the assays (r = 0.92-0.99), the Gen II AMH levels were consistently lower than both Ultrasensitive and picoAMH levels. Moreover, as AMH levels increased, the magnitude of difference grew larger between Gen II and each of the other two assays. Measurement of AMH levels with the picoAMH kit maximized detection at very low levels, particularly in contrast with the Gen II kit. Conversion of AMH levels from different immunoassays using regression equations is potentially highly inaccurate. Copyright © 2014 American Society for Reproductive Medicine. Published by Elsevier Inc. All rights reserved.
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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.
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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.
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Joint Eglin Acoustics Week 2013 Data Report
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
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Analysis of Factors Affecting System Performance in the ASpIRE Challenge
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
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Locating arbitrarily time-dependent sound sources in three dimensional space in real time.
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.
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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.
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Active Self-Testing Noise Measurement Sensors for Large-Scale Environmental Sensor Networks
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
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Evaluation of hearing protection used by police officers in the shooting range.
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.