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.

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.

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

PubMed

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

2014-01-01

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

A combined microphone and camera calibration technique with application to acoustic imaging.

PubMed

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.

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

NASA Technical Reports Server (NTRS)

Greenwood, Eric II; Schmitz, Fredric H.

2009-01-01

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

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.

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

PubMed

Chen, Yung-Yue

2018-05-08

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

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

PubMed Central

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

2017-01-01

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

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

PubMed

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

2017-11-03

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

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.

Spatial acoustic signal processing for immersive communication

NASA Astrophysics Data System (ADS)

Atkins, Joshua

Computing is rapidly becoming ubiquitous as users expect devices that can augment and interact naturally with the world around them. In these systems it is necessary to have an acoustic front-end that is able to capture and reproduce natural human communication. Whether the end point is a speech recognizer or another human listener, the reduction of noise, reverberation, and acoustic echoes are all necessary and complex challenges. The focus of this dissertation is to provide a general method for approaching these problems using spherical microphone and loudspeaker arrays.. In this work, a theory of capturing and reproducing three-dimensional acoustic fields is introduced from a signal processing perspective. In particular, the decomposition of the spatial part of the acoustic field into an orthogonal basis of spherical harmonics provides not only a general framework for analysis, but also many processing advantages. The spatial sampling error limits the upper frequency range with which a sound field can be accurately captured or reproduced. In broadband arrays, the cost and complexity of using multiple transducers is an issue. This work provides a flexible optimization method for determining the location of array elements to minimize the spatial aliasing error. The low frequency array processing ability is also limited by the SNR, mismatch, and placement error of transducers. To address this, a robust processing method is introduced and used to design a reproduction system for rendering over arbitrary loudspeaker arrays or binaurally over headphones. In addition to the beamforming problem, the multichannel acoustic echo cancellation (MCAEC) issue is also addressed. A MCAEC must adaptively estimate and track the constantly changing loudspeaker-room-microphone response to remove the sound field presented over the loudspeakers from that captured by the microphones. In the multichannel case, the system is overdetermined and many adaptive schemes fail to converge to the true impulse response. This forces the need to track both the near and far end room responses. A transform domain method that mitigates this problem is derived and implemented. Results with a real system using a 16-channel loudspeaker array and 32-channel microphone array are presented.

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.

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

PubMed

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

1993-08-01

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

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.

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.

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

PubMed Central

Chung, King

2004-01-01

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

Implementation issues of the nearfield equivalent source imaging microphone array

NASA Astrophysics Data System (ADS)

Bai, Mingsian R.; Lin, Jia-Hong; Tseng, Chih-Wen

2011-01-01

This paper revisits a nearfield microphone array technique termed nearfield equivalent source imaging (NESI) proposed previously. In particular, various issues concerning the implementation of the NESI algorithm are examined. The NESI can be implemented in both the time domain and the frequency domain. Acoustical variables including sound pressure, particle velocity, active intensity and sound power are calculated by using multichannel inverse filters. Issues concerning sensor deployment are also investigated for the nearfield array. The uniform array outperformed a random array previously optimized for far-field imaging, which contradicts the conventional wisdom in far-field arrays. For applications in which only a patch array with scarce sensors is available, a virtual microphone approach is employed to ameliorate edge effects using extrapolation and to improve imaging resolution using interpolation. To enhance the processing efficiency of the time-domain NESI, an eigensystem realization algorithm (ERA) is developed. Several filtering methods are compared in terms of computational complexity. Significant saving on computations can be achieved using ERA and the frequency-domain NESI, as compared to the traditional method. The NESI technique was also experimentally validated using practical sources including a 125 cc scooter and a wooden box model with a loudspeaker fitted inside. The NESI technique proved effective in identifying broadband and non-stationary sources produced by the sources.

Room acoustics analysis using circular arrays: an experimental study based on sound field plane-wave decomposition.

PubMed

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.

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.

Atmospheric effects on microphone array analysis of aircraft vortex sound

DOT National Transportation Integrated Search

2006-05-08

This paper provides the basis of a comprehensive analysis of vortex sound propagation : through the atmosphere in order to assess real atmospheric effects on acoustic array : processing. Such effects may impact vortex localization accuracy and detect...

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

PubMed

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

2018-03-01

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

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

NASA Technical Reports Server (NTRS)

Horne, Clifton; Burnside, Nathan J.

2013-01-01

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

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.

Passive Wake Acoustics Measurements at Denver International Airport

NASA Technical Reports Server (NTRS)

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

2004-01-01

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

Llamas: Large-area microphone arrays and sensing systems

NASA Astrophysics Data System (ADS)

Sanz-Robinson, Josue

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

Acoustic Beam Forming Array Using Feedback-Controlled Microphones for Tuning and Self-Matching of Frequency Response

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.

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.

Initial Assessment of Acoustic Source Visibility with a 24-Element Microphone Array in the Arnold Engineering Development Center 80- by 120-Foot Wind Tunnel at NASA Ames Research Center

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.

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.

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.

Development and Calibration of a Field-Deployable Microphone Phased Array for Propulsion and Airframe Noise Flyover Measurements

NASA Technical Reports Server (NTRS)

Humphreys, William M., Jr.; Lockard, David P.; Khorrami, Mehdi R.; Culliton, William G.; McSwain, Robert G.; Ravetta, Patricio A.; Johns, Zachary

2016-01-01

A new aeroacoustic measurement capability has been developed consisting of a large channelcount, field-deployable microphone phased array suitable for airframe noise flyover measurements for a range of aircraft types and scales. The array incorporates up to 185 hardened, weather-resistant sensors suitable for outdoor use. A custom 4-mA current loop receiver circuit with temperature compensation was developed to power the sensors over extended cable lengths with minimal degradation of the signal to noise ratio and frequency response. Extensive laboratory calibrations and environmental testing of the sensors were conducted to verify the design's performance specifications. A compact data system combining sensor power, signal conditioning, and digitization was assembled for use with the array. Complementing the data system is a robust analysis system capable of near real-time presentation of beamformed and deconvolved contour plots and integrated spectra obtained from array data acquired during flyover passes. Additional instrumentation systems needed to process the array data were also assembled. These include a commercial weather station and a video monitoring / recording system. A detailed mock-up of the instrumentation suite (phased array, weather station, and data processor) was performed in the NASA Langley Acoustic Development Laboratory to vet the system performance. The first deployment of the system occurred at Finnegan Airfield at Fort A.P. Hill where the array was utilized to measure the vehicle noise from a number of sUAS (small Unmanned Aerial System) aircraft. A unique in-situ calibration method for the array microphones using a hovering aerial sound source was attempted for the first time during the deployment.

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.

Dynamically Reconfigurable Microphone Arrays

DTIC Science & Technology

2011-05-01

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

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.

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

NASA Astrophysics Data System (ADS)

Sarradj, Ennes

2010-04-01

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

A Sparsity-Based Approach to 3D Binaural Sound Synthesis Using Time-Frequency Array Processing

NASA Astrophysics Data System (ADS)

Cobos, Maximo; Lopez, JoseJ; Spors, Sascha

2010-12-01

Localization of sounds in physical space plays a very important role in multiple audio-related disciplines, such as music, telecommunications, and audiovisual productions. Binaural recording is the most commonly used method to provide an immersive sound experience by means of headphone reproduction. However, it requires a very specific recording setup using high-fidelity microphones mounted in a dummy head. In this paper, we present a novel processing framework for binaural sound recording and reproduction that avoids the use of dummy heads, which is specially suitable for immersive teleconferencing applications. The method is based on a time-frequency analysis of the spatial properties of the sound picked up by a simple tetrahedral microphone array, assuming source sparseness. The experiments carried out using simulations and a real-time prototype confirm the validity of the proposed approach.

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.

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.

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

PubMed

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

2006-10-01

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

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.

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.

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

PubMed

Mens, Lucas H M

2011-01-01

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

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

NASA Astrophysics Data System (ADS)

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

2017-10-01

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

Adaptive Noise Reduction Techniques for Airborne Acoustic Sensors

DTIC Science & Technology

2012-01-01

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

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.

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.

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.

Uncovering Spatial Variation in Acoustic Environments Using Sound Mapping.

PubMed

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

2016-01-01

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

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

NASA Technical Reports Server (NTRS)

Stephens, David

2011-01-01

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

Multi-microphone adaptive array augmented with visual cueing.

PubMed

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

2012-01-01

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

Localization and separation of acoustic sources by using a 2.5-dimensional circular microphone array.

PubMed

Bai, Mingsian R; Lai, Chang-Sheng; Wu, Po-Chen

2017-07-01

Circular microphone arrays (CMAs) are sufficient in many immersive audio applications because azimuthal angles of sources are considered more important than the elevation angles in those occasions. However, the fact that CMAs do not resolve the elevation angle well can be a limitation for some applications which involves three-dimensional sound images. This paper proposes a 2.5-dimensional (2.5-D) CMA comprised of a CMA and a vertical logarithmic-spacing linear array (LLA) on the top. In the localization stage, two delay-and-sum beamformers are applied to the CMA and the LLA, respectively. The direction of arrival (DOA) is estimated from the product of two array output signals. In the separation stage, Tikhonov regularization and convex optimization are employed to extract the source amplitudes on the basis of the estimated DOA. The extracted signals from two arrays are further processed by the normalized least-mean-square algorithm with the internal iteration to yield the source signal with improved quality. To validate the 2.5-D CMA experimentally, a three-dimensionally printed circular array comprised of a 24-element CMA and an eight-element LLA is constructed. Objective perceptual evaluation of speech quality test and a subjective listening test are also undertaken.

Aeroacoustic Characterization of the NASA Ames Experimental Aero-Physics Branch 32- by 48-Inch Subsonic Wind Tunnel with a 24-Element Phased Microphone Array

NASA Technical Reports Server (NTRS)

Costanza, Bryan T.; Horne, William C.; Schery, S. D.; Babb, Alex T.

2011-01-01

The Aero-Physics Branch at NASA Ames Research Center utilizes a 32- by 48-inch subsonic wind tunnel for aerodynamics research. The feasibility of acquiring acoustic measurements with a phased microphone array was recently explored. Acoustic characterization of the wind tunnel was carried out with a floor-mounted 24-element array and two ceiling-mounted speakers. The minimum speaker level for accurate level measurement was evaluated for various tunnel speeds up to a Mach number of 0.15 and streamwise speaker locations. A variety of post-processing procedures, including conventional beamforming and deconvolutional processing such as TIDY, were used. The speaker measurements, with and without flow, were used to compare actual versus simulated in-flow speaker calibrations. Data for wind-off speaker sound and wind-on tunnel background noise were found valuable for predicting sound levels for which the speakers were detectable when the wind was on. Speaker sources were detectable 2 - 10 dB below the peak background noise level with conventional data processing. The effectiveness of background noise cross-spectral matrix subtraction was assessed and found to improve the detectability of test sound sources by approximately 10 dB over a wide frequency range.

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.

Real-time algorithm for acoustic imaging with a microphone array.

PubMed

Huang, Xun

2009-05-01

Acoustic phased array has become an important testing tool in aeroacoustic research, where the conventional beamforming algorithm has been adopted as a classical processing technique. The computation however has to be performed off-line due to the expensive cost. An innovative algorithm with real-time capability is proposed in this work. The algorithm is similar to a classical observer in the time domain while extended for the array processing to the frequency domain. The observer-based algorithm is beneficial mainly for its capability of operating over sampling blocks recursively. The expensive experimental time can therefore be reduced extensively since any defect in a testing can be corrected instantaneously.

Theory and investigation of acoustic multiple-input multiple-output systems based on spherical arrays in a room.

PubMed

Morgenstern, Hai; Rafaely, Boaz; Zotter, Franz

2015-11-01

Spatial attributes of room acoustics have been widely studied using microphone and loudspeaker arrays. However, systems that combine both arrays, referred to as multiple-input multiple-output (MIMO) systems, have only been studied to a limited degree in this context. These systems can potentially provide a powerful tool for room acoustics analysis due to the ability to simultaneously control both arrays. This paper offers a theoretical framework for the spatial analysis of enclosed sound fields using a MIMO system comprising spherical loudspeaker and microphone arrays. A system transfer function is formulated in matrix form for free-field conditions, and its properties are studied using tools from linear algebra. The system is shown to have unit-rank, regardless of the array types, and its singular vectors are related to the directions of arrival and radiation at the microphone and loudspeaker arrays, respectively. The formulation is then generalized to apply to rooms, using an image source method. In this case, the rank of the system is related to the number of significant reflections. The paper ends with simulation studies, which support the developed theory, and with an extensive reflection analysis of a room impulse response, using the platform of a MIMO system.

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.

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

NASA Astrophysics Data System (ADS)

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

2011-08-01

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

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

PubMed

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

2011-08-01

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

Plane-wave decomposition by spherical-convolution microphone array

NASA Astrophysics Data System (ADS)

Rafaely, Boaz; Park, Munhum

2004-05-01

Reverberant sound fields are widely studied, as they have a significant influence on the acoustic performance of enclosures in a variety of applications. For example, the intelligibility of speech in lecture rooms, the quality of music in auditoria, the noise level in offices, and the production of 3D sound in living rooms are all affected by the enclosed sound field. These sound fields are typically studied through frequency response measurements or statistical measures such as reverberation time, which do not provide detailed spatial information. The aim of the work presented in this seminar is the detailed analysis of reverberant sound fields. A measurement and analysis system based on acoustic theory and signal processing, designed around a spherical microphone array, is presented. Detailed analysis is achieved by decomposition of the sound field into waves, using spherical Fourier transform and spherical convolution. The presentation will include theoretical review, simulation studies, and initial experimental results.

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.

Noise Reduction with Microphone Arrays for Speaker Identification

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

Cohen, Z

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

Broadband implementation of coprime linear microphone arrays for direction of arrival estimation.

PubMed

Bush, Dane; Xiang, Ning

2015-07-01

Coprime arrays represent a form of sparse sensing which can achieve narrow beams using relatively few elements, exceeding the spatial Nyquist sampling limit. The purpose of this paper is to expand on and experimentally validate coprime array theory in an acoustic implementation. Two nested sparse uniform linear subarrays with coprime number of elements ( M and N) each produce grating lobes that overlap with one another completely in just one direction. When the subarray outputs are combined it is possible to retain the shared beam while mostly canceling the other superfluous grating lobes. In this way a small number of microphones ( N+M-1) creates a narrow beam at higher frequencies, comparable to a densely populated uniform linear array of MN microphones. In this work beampatterns are simulated for a range of single frequencies, as well as bands of frequencies. Narrowband experimental beampatterns are shown to correspond with simulated results even at frequencies other than the arrays design frequency. Narrowband side lobe locations are shown to correspond to the theoretical values. Side lobes in the directional pattern are mitigated by increasing bandwidth of analyzed signals. Direction of arrival estimation is also implemented for two simultaneous noise sources in a free field condition.

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

PubMed Central

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

2014-01-01

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

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

PubMed

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

2014-03-05

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

Modal smoothing for analysis of room reflections measured with spherical microphone and loudspeaker arrays.

PubMed

Morgenstern, Hai; Rafaely, Boaz

2018-02-01

Spatial analysis of room acoustics is an ongoing research topic. Microphone arrays have been employed for spatial analyses with an important objective being the estimation of the direction-of-arrival (DOA) of direct sound and early room reflections using room impulse responses (RIRs). An optimal method for DOA estimation is the multiple signal classification algorithm. When RIRs are considered, this method typically fails due to the correlation of room reflections, which leads to rank deficiency of the cross-spectrum matrix. Preprocessing methods for rank restoration, which may involve averaging over frequency, for example, have been proposed exclusively for spherical arrays. However, these methods fail in the case of reflections with equal time delays, which may arise in practice and could be of interest. In this paper, a method is proposed for systems that combine a spherical microphone array and a spherical loudspeaker array, referred to as multiple-input multiple-output systems. This method, referred to as modal smoothing, exploits the additional spatial diversity for rank restoration and succeeds where previous methods fail, as demonstrated in a simulation study. Finally, combining modal smoothing with a preprocessing method is proposed in order to increase the number of DOAs that can be estimated using low-order spherical loudspeaker arrays.

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.

Acoustic centering of sources measured by surrounding spherical microphone arrays.

PubMed

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

2011-10-01

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

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.

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

NASA Image and Video Library

2004-01-13

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

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

PubMed

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

2006-06-01

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

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.

Lightning Location Using Acoustic Signals

NASA Astrophysics Data System (ADS)

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

2013-05-01

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

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.

Improved methods for fan sound field determination

NASA Technical Reports Server (NTRS)

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

1981-01-01

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

Sound source localization on an axial fan at different operating points

NASA Astrophysics Data System (ADS)

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

2016-08-01

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

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

NASA Technical Reports Server (NTRS)

Babkin, Alexander S.

1988-01-01

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

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.

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

NASA Astrophysics Data System (ADS)

Gover, Bradford Noel

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

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.

Measurements of Infrared and Acoustic Source Distributions in Jet Plumes

NASA Technical Reports Server (NTRS)

Agboola, Femi A.; Bridges, James; Saiyed, Naseem

2004-01-01

The aim of this investigation was to use the linear phased array (LPA) microphones and infrared (IR) imaging to study the effects of advanced nozzle-mixing techniques on jet noise reduction. Several full-scale engine nozzles were tested at varying power cycles with the linear phased array setup parallel to the jet axis. The array consisted of 16 sparsely distributed microphones. The phased array microphone measurements were taken at a distance of 51.0 ft (15.5 m) from the jet axis, and the results were used to obtain relative overall sound pressure levels from one nozzle design to the other. The IR imaging system was used to acquire real-time dynamic thermal patterns of the exhaust jet from the nozzles tested. The IR camera measured the IR radiation from the nozzle exit to a distance of six fan diameters (X/D(sub FAN) = 6), along the jet plume axis. The images confirmed the expected jet plume mixing intensity, and the phased array results showed the differences in sound pressure level with respect to nozzle configurations. The results show the effects of changes in configurations to the exit nozzles on both the flows mixing patterns and radiant energy dissipation patterns. By comparing the results from these two measurements, a relationship between noise reduction and core/bypass flow mixing is demonstrated.

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.

Moving microphone arrays to reduce spatial aliasing in the beamforming technique: theoretical background and numerical investigation.

PubMed

Cigada, Alfredo; Lurati, Massimiliano; Ripamonti, Francesco; Vanali, Marcello

2008-12-01

This paper introduces a measurement technique aimed at reducing or possibly eliminating the spatial aliasing problem in the beamforming technique. Beamforming main disadvantages are a poor spatial resolution, at low frequency, and the spatial aliasing problem, at higher frequency, leading to the identification of false sources. The idea is to move the microphone array during the measurement operation. In this paper, the proposed approach is theoretically and numerically investigated by means of simple sound propagation models, proving its efficiency in reducing the spatial aliasing. A number of different array configurations are numerically investigated together with the most important parameters governing this measurement technique. A set of numerical results concerning the case of a planar rotating array is shown, together with a first experimental validation of the method.

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

PubMed

Oreinos, Chris; Buchholz, Jörg M

2015-06-01

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

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

PubMed

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

2015-04-01

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

Three-Dimensional Application of DAMAS Methodology for Aeroacoustic Noise Source Definition

NASA Technical Reports Server (NTRS)

Brooks, Thomas F.; Humphreys, William M., Jr.

2005-01-01

At the 2004 AIAA/CEAS Aeroacoustic Conference, a breakthrough in acoustic microphone array technology was reported by the authors. A Deconvolution Approach for the Mapping of Acoustic Sources (DAMAS) was developed which decouples the array design and processing influence from the noise being measured, using a simple and robust algorithm. For several prior airframe noise studies, it was shown to permit an unambiguous and accurate determination of acoustic source position and strength. As a follow-on effort, this paper examines the technique for three-dimensional (3D) applications. First, the beamforming ability for arrays, of different size and design, to focus longitudinally and laterally is examined for a range of source positions and frequency. Advantage is found for larger array designs with higher density microphone distributions towards the center. After defining a 3D grid generalized with respect to the array s beamforming characteristics, DAMAS is employed in simulated and experimental noise test cases. It is found that spatial resolution is much less sharp in the longitudinal direction in front of the array compared to side-to-side lateral resolution. 3D DAMAS becomes useful for sufficiently large arrays at sufficiently high frequency. But, such can be a challenge to computational capabilities, with regard to the required expanse and number of grid points. Also, larger arrays can strain basic physical modeling assumptions that DAMAS and all traditional array methodologies use. An important experimental result is that turbulent shear layers can negatively impact attainable beamforming resolution. Still, the usefulness of 3D DAMAS is demonstrated by the measurement of landing gear noise source distributions in a difficult hard-wall wind tunnel environment.

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

NASA Astrophysics Data System (ADS)

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

2016-01-01

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

Impedance Eduction in Ducts with Higher-Order Modes and Flow

NASA Technical Reports Server (NTRS)

Watson, Willie R.; Jones, Michael G.

2009-01-01

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

Ground effects on aircraft noise. [near grazing incidence

NASA Technical Reports Server (NTRS)

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

1979-01-01

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

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.

Speaker Localisation Using Time Difference of Arrival

DTIC Science & Technology

2008-04-01

School of Electrical and Electronic Engineering of the University of Adelaide. His area of expertise and interest is in Signal Processing including audio ...support of Theatre intelligence capabilities. His recent research interests include: information visualisation , text and data mining, and speech and...by: steering microphone arrays to improve the quality of audio pickup for recording, communication and transcription; enhancing the separation – and

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.

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.

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

DTIC Science & Technology

1955-05-01

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

Acoustic imaging of aircraft wake vortex dynamics

DOT National Transportation Integrated Search

2005-06-01

The experience in utilizing a phased microphone array to passively image aircraft wake : vortices is highlighted. It is demonstrated that the array can provide visualization of wake : dynamics similar to smoke release or natural condensation of vorti...

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.

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

NASA Astrophysics Data System (ADS)

Fleury, Vincent; Davy, Renaud

2016-03-01

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

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.

Locating and Quantifying Broadband Fan Sources Using In-Duct Microphones

NASA Technical Reports Server (NTRS)

Dougherty, Robert P.; Walker, Bruce E.; Sutliff, Daniel L.

2010-01-01

In-duct beamforming techniques have been developed for locating broadband noise sources on a low-speed fan and quantifying the acoustic power in the inlet and aft fan ducts. The NASA Glenn Research Center's Advanced Noise Control Fan was used as a test bed. Several of the blades were modified to provide a broadband source to evaluate the efficacy of the in-duct beamforming technique. Phased arrays consisting of rings and line arrays of microphones were employed. For the imaging, the data were mathematically resampled in the frame of reference of the rotating fan. For both the imaging and power measurement steps, array steering vectors were computed using annular duct modal expansions, selected subsets of the cross spectral matrix elements were used, and the DAMAS and CLEAN-SC deconvolution algorithms were applied.

J-FLiC UAS Flights for Acoustic Testing Research

NASA Technical Reports Server (NTRS)

Motter, Mark A.; High, James W.

2016-01-01

The jet-powered flying testbed (J-FLiC) unmanned aircraft system (UAS) successfully completed twenty-six flights at Fort AP Hill, VA, from 27 August until September 3 2015, supporting tests of a microphone array system for aircraft noise measurement. The test vehicles, J-FLiC NAVY2 (N508NU), and J-FLiC 4 (N509NU), were flown under manual and autopiloted control in a variety of test conditions: clean at speeds ranging from 80 to 150 knots; and full landing configuration at speeds ranging from 50 to 95 knots. During the test campaign, autopilot capability was incrementally improved to ultimately provide a high degree of accuracy and repeatability of the critical test requirements for airspeed, altitude, runway alignment and position over the microphone array. Manual flights were performed for test conditions at the both ends of the speed envelope where autopiloted flight would have required flight beyond visual range and more extensive developmental work. The research objectives of the campaign were fully achieved. The ARMD Integrated Systems Research Program (ISRP) Environmentally Responsible Aviation (ERA) Project aims to develop the enabling capabilities/technologies that will allow prediction/reduction of aircraft noise. A primary measurement tool for ascertaining and characterizing empirically the effectiveness of various noise reduction technologies is a microphone phased array system. Such array systems need to be vetted and certified for operational use via field deployments and overflights of the array with test aircraft, in this case with sUAS aircraft such as J-FLiC.

Sound field reconstruction within an entire cavity by plane wave expansions using a spherical microphone array.

PubMed

Wang, Yan; Chen, Kean

2017-10-01

A spherical microphone array has proved effective in reconstructing an enclosed sound field by a superposition of spherical wave functions in Fourier domain. It allows successful reconstructions surrounding the array, but the accuracy will be degraded at a distance. In order to extend the effective reconstruction to the entire cavity, a plane-wave basis in space domain is used owing to its non-decaying propagating characteristic and compared with the conventional spherical wave function method in a low frequency sound field within a cylindrical cavity. The sensitivity to measurement noise, the effects of the numbers of plane waves, and measurement positions are discussed. Simulations show that under the same measurement conditions, the plane wave function method is superior in terms of reconstruction accuracy and data processing efficiency, that is, the entire sound field imaging can be achieved by only one time calculation instead of translations of local sets of coefficients with respect to every measurement position into a global one. An experiment was conducted inside an aircraft cabin mock-up for validation. Additionally, this method provides an alternative possibility to recover the coefficients of high order spherical wave functions in a global coordinate system without coordinate translations with respect to local origins.

First Test of Fan Active Noise Control (ANC) Completed

NASA Technical Reports Server (NTRS)

2005-01-01

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

A Deconvolution Approach for the Mapping of Acoustic Sources (DAMAS) Determined from Phased Microphone Arrays

NASA Technical Reports Server (NTRS)

Brooks, Thomas F.; Humphreys, William M.

2006-01-01

Current processing of acoustic array data is burdened with considerable uncertainty. This study reports an original methodology that serves to demystify array results, reduce misinterpretation, and accurately quantify position and strength of acoustic sources. Traditional array results represent noise sources that are convolved with array beamform response functions, which depend on array geometry, size (with respect to source position and distributions), and frequency. The Deconvolution Approach for the Mapping of Acoustic Sources (DAMAS) method removes beamforming characteristics from output presentations. A unique linear system of equations accounts for reciprocal influence at different locations over the array survey region. It makes no assumption beyond the traditional processing assumption of statistically independent noise sources. The full rank equations are solved with a new robust iterative method. DAMAS is quantitatively validated using archival data from a variety of prior high-lift airframe component noise studies, including flap edge/cove, trailing edge, leading edge, slat, and calibration sources. Presentations are explicit and straightforward, as the noise radiated from a region of interest is determined by simply summing the mean-squared values over that region. DAMAS can fully replace existing array processing and presentations methodology in most applications. It appears to dramatically increase the value of arrays to the field of experimental acoustics.

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.

Helicopter noise experiments in an urban environment

DOT National Transportation Integrated Search

1974-08-01

In two series of helicopter noise experiments, soundpressurelevel recordings were made on the ground while a helicopter flew over (i) an array of microphones placed in an open field, and (ii) a similar array placed in the center of a city stree...

MEMS Based Acoustic Array

NASA Technical Reports Server (NTRS)

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

2006-01-01

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

Assessment of Microphone Phased Array for Measuring Launch Vehicle Lift-off Acoustics

NASA Technical Reports Server (NTRS)

Garcia, Roberto

2012-01-01

The specific purpose of the present work was to demonstrate the suitability of a microphone phased array for launch acoustics applications via participation in selected firings of the Ares I Scale Model Acoustics Test. The Ares I Scale Model Acoustics Test is a part of the discontinued Constellation Program Ares I Project, but the basic understanding gained from this test is expected to help development of the Space Launch System vehicles. Correct identification of sources not only improves the predictive ability, but provides guidance for a quieter design of the launch pad and optimization of the water suppression system. This document contains the results of the NASA Engineering and Safety Center assessment.

Microphone array measurement system for analysis of directional and spatial variations of sound fields.

PubMed

Gover, Bradford N; Ryan, James G; Stinson, Michael R

2002-11-01

A measurement system has been developed that is capable of analyzing the directional and spatial variations in a reverberant sound field. A spherical, 32-element array of microphones is used to generate a narrow beam that is steered in 60 directions. Using an omnidirectional loudspeaker as excitation, the sound pressure arriving from each steering direction is measured as a function of time, in the form of pressure impulse responses. By subsequent analysis of these responses, the variation of arriving energy with direction is studied. The directional diffusion and directivity index of the arriving sound can be computed, as can the energy decay rate in each direction. An analysis of the 32 microphone responses themselves allows computation of the point-to-point variation of reverberation time and of sound pressure level, as well as the spatial cross-correlation coefficient, over the extent of the array. The system has been validated in simple sound fields in an anechoic chamber and in a reverberation chamber. The system characterizes these sound fields as expected, both quantitatively from the measures and qualitatively from plots of the arriving energy versus direction. It is anticipated that the system will be of value in evaluating the directional distribution of arriving energy and the degree and diffuseness of sound fields in rooms.

A reverberation-time-aware DNN approach leveraging spatial information for microphone array dereverberation

NASA Astrophysics Data System (ADS)

Wu, Bo; Yang, Minglei; Li, Kehuang; Huang, Zhen; Siniscalchi, Sabato Marco; Wang, Tong; Lee, Chin-Hui

2017-12-01

A reverberation-time-aware deep-neural-network (DNN)-based multi-channel speech dereverberation framework is proposed to handle a wide range of reverberation times (RT60s). There are three key steps in designing a robust system. First, to accomplish simultaneous speech dereverberation and beamforming, we propose a framework, namely DNNSpatial, by selectively concatenating log-power spectral (LPS) input features of reverberant speech from multiple microphones in an array and map them into the expected output LPS features of anechoic reference speech based on a single deep neural network (DNN). Next, the temporal auto-correlation function of received signals at different RT60s is investigated to show that RT60-dependent temporal-spatial contexts in feature selection are needed in the DNNSpatial training stage in order to optimize the system performance in diverse reverberant environments. Finally, the RT60 is estimated to select the proper temporal and spatial contexts before feeding the log-power spectrum features to the trained DNNs for speech dereverberation. The experimental evidence gathered in this study indicates that the proposed framework outperforms the state-of-the-art signal processing dereverberation algorithm weighted prediction error (WPE) and conventional DNNSpatial systems without taking the reverberation time into account, even for extremely weak and severe reverberant conditions. The proposed technique generalizes well to unseen room size, array geometry and loudspeaker position, and is robust to reverberation time estimation error.

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

PubMed

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

2006-04-01

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

Two-dimensional grid-free compressive beamforming.

PubMed

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.

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

PubMed

Moghaddasi, Hanie; Almasganj, Farshad; Zoroufian, Arezoo

2017-07-01

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

Monitoring of atmospheric nuclear explosions with infrasonic microphone arrays

NASA Astrophysics Data System (ADS)

Wilson, Charles R.

2002-11-01

A review is given of the various United States programs for the infrasonic monitoring of atmospheric nuclear explosions from their inception in 1946 to their termination in 1975. The US Atomic Energy Detection System (USAEDS) monitored all nuclear weapons tests that were conducted by the Soviet Union, France, China, and the US with arrays of sensitive microbarographs in a worldwide network of infrasonic stations. A discussion of the source mechanism for the creation and subsequent propagation around the globe of long wavelength infrasound from explosions (volcanic and nuclear) is given to show the efficacy of infrasonic monitoring for the detection of atmospheric nuclear weapons tests. The equipment that was used for infrasound detection, the design of the sensor arrays, and the data processing techniques that were used by USAEDS are all discussed.

Phased Array Noise Source Localization Measurements Made on a Williams International FJ44 Engine

NASA Technical Reports Server (NTRS)

Podboy, Gary G.; Horvath, Csaba

2010-01-01

A 48-microphone planar phased array system was used to acquire noise source localization data on a full-scale Williams International FJ44 turbofan engine. Data were acquired with the array at three different locations relative to the engine, two on the side and one in front of the engine. At the two side locations the planar microphone array was parallel to the engine centerline; at the front location the array was perpendicular to the engine centerline. At each of the three locations, data were acquired at eleven different engine operating conditions ranging from engine idle to maximum (take off) speed. Data obtained with the array off to the side of the engine were spatially filtered to separate the inlet and nozzle noise. Tones occurring in the inlet and nozzle spectra were traced to the low and high speed spools within the engine. The phased array data indicate that the Inflow Control Device (ICD) used during this test was not acoustically transparent; instead, some of the noise emanating from the inlet reflected off of the inlet lip of the ICD. This reflection is a source of error for far field noise measurements made during the test. The data also indicate that a total temperature rake in the inlet of the engine is a source of fan noise.

Design framework for spherical microphone and loudspeaker arrays in a multiple-input multiple-output system.

PubMed

Morgenstern, Hai; Rafaely, Boaz; Noisternig, Markus

2017-03-01

Spherical microphone arrays (SMAs) and spherical loudspeaker arrays (SLAs) facilitate the study of room acoustics due to the three-dimensional analysis they provide. More recently, systems that combine both arrays, referred to as multiple-input multiple-output (MIMO) systems, have been proposed due to the added spatial diversity they facilitate. The literature provides frameworks for designing SMAs and SLAs separately, including error analysis from which the operating frequency range (OFR) of an array is defined. However, such a framework does not exist for the joint design of a SMA and a SLA that comprise a MIMO system. This paper develops a design framework for MIMO systems based on a model that addresses errors and highlights the importance of a matched design. Expanding on a free-field assumption, errors are incorporated separately for each array and error bounds are defined, facilitating error analysis for the system. The dependency of the error bounds on the SLA and SMA parameters is studied and it is recommended that parameters should be chosen to assure matched OFRs of the arrays in MIMO system design. A design example is provided, demonstrating the superiority of a matched system over an unmatched system in the synthesis of directional room impulse responses.

DAMAS Processing for a Phased Array Study in the NASA Langley Jet Noise Laboratory

NASA Technical Reports Server (NTRS)

Brooks, Thomas F.; Humphreys, William M.; Plassman, Gerald e.

2010-01-01

A jet noise measurement study was conducted using a phased microphone array system for a range of jet nozzle configurations and flow conditions. The test effort included convergent and convergent/divergent single flow nozzles, as well as conventional and chevron dual-flow core and fan configurations. Cold jets were tested with and without wind tunnel co-flow, whereas, hot jets were tested only with co-flow. The intent of the measurement effort was to allow evaluation of new phased array technologies for their ability to separate and quantify distributions of jet noise sources. In the present paper, the array post-processing method focused upon is DAMAS (Deconvolution Approach for the Mapping of Acoustic Sources) for the quantitative determination of spatial distributions of noise sources. Jet noise is highly complex with stationary and convecting noise sources, convecting flows that are the sources themselves, and shock-related and screech noise for supersonic flow. The analysis presented in this paper addresses some processing details with DAMAS, for the array positioned at 90 (normal) to the jet. The paper demonstrates the applicability of DAMAS and how it indicates when strong coherence is present. Also, a new approach to calibrating the array focus and position is introduced and demonstrated.

Source sparsity control of sound field reproduction using the elastic-net and the lasso minimizers.

PubMed

Gauthier, P-A; Lecomte, P; Berry, A

2017-04-01

Sound field reproduction is aimed at the reconstruction of a sound pressure field in an extended area using dense loudspeaker arrays. In some circumstances, sound field reproduction is targeted at the reproduction of a sound field captured using microphone arrays. Although methods and algorithms already exist to convert microphone array recordings to loudspeaker array signals, one remaining research question is how to control the spatial sparsity in the resulting loudspeaker array signals and what would be the resulting practical advantages. Sparsity is an interesting feature for spatial audio since it can drastically reduce the number of concurrently active reproduction sources and, therefore, increase the spatial contrast of the solution at the expense of a difference between the target and reproduced sound fields. In this paper, the application of the elastic-net cost function to sound field reproduction is compared to the lasso cost function. It is shown that the elastic-net can induce solution sparsity and overcomes limitations of the lasso: The elastic-net solves the non-uniqueness of the lasso solution, induces source clustering in the sparse solution, and provides a smoother solution within the activated source clusters.

Identification of Noise Sources During Rocket Engine Test Firings and a Rocket Launch Using a Microphone Phased-Array

NASA Technical Reports Server (NTRS)

Panda, Jayanta; Mosher, Robert N.; Porter, Barry J.

2013-01-01

A 70 microphone, 10-foot by 10-foot, microphone phased array was built for use in the harsh environment of rocket launches. The array was setup at NASA Wallops launch pad 0A during a static test firing of Orbital Sciences' Antares engines, and again during the first launch of the Antares vehicle. It was placed 400 feet away from the pad, and was hoisted on a scissor lift 40 feet above ground. The data sets provided unprecedented insight into rocket noise sources. The duct exit was found to be the primary source during the static test firing; the large amount of water injected beneath the nozzle exit and inside the plume duct quenched all other sources. The maps of the noise sources during launch were found to be time-dependent. As the engines came to full power and became louder, the primary source switched from the duct inlet to the duct exit. Further elevation of the vehicle caused spilling of the hot plume, resulting in a distributed noise map covering most of the pad. As the entire plume emerged from the duct, and the ondeck water system came to full power, the plume itself became the loudest noise source. These maps of the noise sources provide vital insight for optimization of sound suppression systems for future Antares launches.

Population density estimated from locations of individuals on a passive detector array

USGS Publications Warehouse

Efford, Murray G.; Dawson, Deanna K.; Borchers, David L.

2009-01-01

The density of a closed population of animals occupying stable home ranges may be estimated from detections of individuals on an array of detectors, using newly developed methods for spatially explicit capture–recapture. Likelihood-based methods provide estimates for data from multi-catch traps or from devices that record presence without restricting animal movement ("proximity" detectors such as camera traps and hair snags). As originally proposed, these methods require multiple sampling intervals. We show that equally precise and unbiased estimates may be obtained from a single sampling interval, using only the spatial pattern of detections. This considerably extends the range of possible applications, and we illustrate the potential by estimating density from simulated detections of bird vocalizations on a microphone array. Acoustic detection can be defined as occurring when received signal strength exceeds a threshold. We suggest detection models for binary acoustic data, and for continuous data comprising measurements of all signals above the threshold. While binary data are often sufficient for density estimation, modeling signal strength improves precision when the microphone array is small.

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

PubMed Central

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

2009-01-01

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

Dragon Ears airborne acoustic array: CSP analysis applied to cross array to compute real-time 2D acoustic sound field

NASA Astrophysics Data System (ADS)

Cerwin, Steve; Barnes, Julie; Kell, Scott; Walters, Mark

2003-09-01

This paper describes development and application of a novel method to accomplish real-time solid angle acoustic direction finding using two 8-element orthogonal microphone arrays. The developed prototype system was intended for localization and signature recognition of ground-based sounds from a small UAV. Recent advances in computer speeds have enabled the implementation of microphone arrays in many audio applications. Still, the real-time presentation of a two-dimensional sound field for the purpose of audio target localization is computationally challenging. In order to overcome this challenge, a crosspower spectrum phase1 (CSP) technique was applied to each 8-element arm of a 16-element cross array to provide audio target localization. In this paper, we describe the technique and compare it with two other commonly used techniques; Cross-Spectral Matrix2 and MUSIC3. The results show that the CSP technique applied to two 8-element orthogonal arrays provides a computationally efficient solution with reasonable accuracy and tolerable artifacts, sufficient for real-time applications. Additional topics include development of a synchronized 16-channel transmitter and receiver to relay the airborne data to the ground-based processor and presentation of test data demonstrating both ground-mounted operation and airborne localization of ground-based gunshots and loud engine sounds.

Source Identification and Location Techniques

NASA Technical Reports Server (NTRS)

Weir, Donald; Bridges, James; Agboola, Femi; Dougherty, Robert

2001-01-01

Mr. Weir presented source location results obtained from an engine test as part of the Engine Validation of Noise Reduction Concepts program. Two types of microphone arrays were used in this program to determine the jet noise source distribution for the exhaust from a 4.3 bypass ratio turbofan engine. One was a linear array of 16 microphones located on a 25 ft. sideline and the other was a 103 microphone 3-D "cage" array in the near field of the jet. Data were obtained from a baseline nozzle and from numerous nozzle configuration using chevrons and/or tabs to reduce the jet noise. Mr. Weir presented data from two configurations: the baseline nozzle and a nozzle configuration with chevrons on both the core and bypass nozzles. This chevron configuration had achieved a jet noise reduction of 4 EPNdB in small scale tests conducted at the Glenn Research Center. IR imaging showed that the chevrons produced significant improvements in mixing and greatly reduced the length of the jet potential core. Comparison of source location data from the 1-D phased array showed a shift of the noise sources towards the nozzle and clear reductions of the sources due to the noise reduction devices. Data from the 3-D array showed a single source at a frequency of 125 Hz. located several diameters downstream from the nozzle exit. At 250 and 400 Hz., multiple sources, periodically spaced, appeared to exist downstream of the nozzle. The trend of source location moving toward the nozzle exit with increasing frequency was also observed. The 3-D array data also showed a reduction in source strength with the addition of chevrons. The overall trend of source location with frequency was compared for the two arrays and with classical experience. Similar trends were observed. Although overall trends with frequency and addition of suppression devices were consistent between the data from the 1-D and the 3-D arrays, a comparison of the details of the inferred source locations did show differences. A flight test is planned to determine if the hardware tested statically will achieve similar reductions in flight.

Characteristics of propeller noise on an aircraft fuselage related to interior noise transmission

NASA Technical Reports Server (NTRS)

Mixson, J. S.; Barton, C. K.; Piersol, A. G.; Wilby, J. F.

1979-01-01

Exterior noise was measured on the fuselage of a twin-engine, light aircraft at four values of engine rpm in ground static tests and at forward speeds up to 36 m/s in taxi tests. Propeller noise levels, spectra, and correlations were determined using a horizontal array of seven flush-mounted microphones and a vertical array of four flush-mounted microphones in the propeller plane. The measured levels and spectra are compared with predictions based on empirical and analytical methods for static and taxi conditions. Trace wavelengths of the propeller noise field, obtained from point-to-point correlations, are compared with the aircraft sidewall structural dimensions, and some analytical results are presented that suggest the sensitivity of interior noise transmission to variations of the propeller noise characteristics.

Wavevector-Frequency Analysis with Applications to Acoustics

DTIC Science & Technology

1994-01-01

Turbulent Boundary Layer Pressure Measured by Microphone Arrays," Journal of the Acoustical Society of America, vol. 49, no. 3, March 1971 , pp. 862-877. 1...ARplications of Green’s FuntionsinScie,.-and Enginlering, Prentice-Hall, Inc., Englewood Hills, NJ, 1971 . 9. 3. Ffowcs-Williams et al., Modern Methods for...variables of a random process are kalled Joint w.merit ,. The m,n-th joint moment of the random variables, v and w, iz flefined by E ,N 1 f (aB) do d- where

The Benefit of a Visually Guided Beamformer in a Dynamic Speech Task

PubMed Central

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

Experimental investigation into infrasonic emissions from atmospheric turbulence.

PubMed

Shams, Qamar A; Zuckerwar, Allan J; Burkett, Cecil G; Weistroffer, George R; Hugo, Derek R

2013-03-01

Clear air turbulence (CAT) is the leading cause of in-flight injuries and in severe cases can result in fatalities. The purpose of this work is to design and develop an infrasonic array network for early warning of clear air turbulence. The infrasonic system consists of an infrasonic three-microphone array, compact windscreens, and data management system. Past experimental efforts to detect acoustic emissions from CAT have been limited. An array of three infrasonic microphones, operating in the field at NASA Langley Research Center, on several occasions received signals interpreted as infrasonic emissions from CAT. Following comparison with current lidar and other past methods, the principle of operation, the experimental methods, and experimental data are presented for case studies and confirmed by pilot reports. The power spectral density of the received signals was found to fit a power law having an exponent of -6 to -7, which is found to be characteristics of infrasonic emissions from CAT, in contrast to findings of the past.

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.

NASA Test Flights Examine Effect of Atmospheric Turbulence on Sonic Booms

NASA Image and Video Library

2016-07-20

One of three microphone arrays positioned strategically along the ground at Edwards Air Force Base, California, sits ready to collect sound signatures from sonic booms created by a NASA F/A-18 during the SonicBAT flight series. The arrays collected the sound signatures of booms that had traveled through atmospheric turbulence before reaching the ground.

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.

Hydrogel microphones for stealthy underwater listening

PubMed Central

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

2016-01-01

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

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.

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.

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.

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.

High-bandwidth acoustic detection system (HBADS) for stripmap synthetic aperture acoustic imaging of canonical ground targets using airborne sound and a 16 element receiving array

NASA Astrophysics Data System (ADS)

Bishop, Steven S.; Moore, Timothy R.; Gugino, Peter; Smith, Brett; Kirkwood, Kathryn P.; Korman, Murray S.

2018-04-01

High Bandwidth Acoustic Detection System (HBADS) is an emerging active acoustic sensor technology undergoing study by the US Army's Night Vision and Electronic Sensors Directorate. Mounted on a commercial all-terrain type vehicle, it uses a single source pulse chirp while moving and a new array (two rows each containing eight microphones) mounted horizontally and oriented in a side scan mode. Experiments are performed with this synthetic aperture air acoustic (SAA) array to image canonical ground targets in clutter or foliage. A commercial audio speaker transmits a linear FM chirp having an effective frequency range of 2 kHz to 15 kHz. The system includes an inertial navigation system using two differential GPS antennas, an inertial measurement unit and a wheel coder. A web camera is mounted midway between the two horizontal microphone arrays and a meteorological unit acquires ambient, temperature, pressure and humidity information. A data acquisition system is central to the system's operation, which is controlled by a laptop computer. Recent experiments include imaging canonical targets located on the ground in a grassy field and similar targets camouflaged by natural vegetation along the side of a road. A recent modification involves implementing SAA stripmap mode interferometry for computing the reflectance of targets placed along the ground. Typical strip map SAA parameters are chirp pulse = 10 or 40 ms, slant range resolution c/(2*BW) = 0.013 m, microphone diameter D = 0.022 m, azimuthal resolution (D/2) = 0.01, air sound speed c ≍ 340 m/s and maximum vehicle speed ≍ 2 m/s.

Vehicle Counting and Moving Direction Identification Based on Small-Aperture Microphone Array.

PubMed

Zu, Xingshui; Zhang, Shaojie; Guo, Feng; Zhao, Qin; Zhang, Xin; You, Xing; Liu, Huawei; Li, Baoqing; Yuan, Xiaobing

2017-05-10

The varying trend of a moving vehicle's angles provides much important intelligence for an unattended ground sensor (UGS) monitoring system. The present study investigates the capabilities of a small-aperture microphone array (SAMA) based system to identify the number and moving direction of vehicles travelling on a previously established route. In this paper, a SAMA-based acoustic monitoring system, including the system hardware architecture and algorithm mechanism, is designed as a single node sensor for the application of UGS. The algorithm is built on the varying trend of a vehicle's bearing angles around the closest point of approach (CPA). We demonstrate the effectiveness of our proposed method with our designed SAMA-based monitoring system in various experimental sites. The experimental results in harsh conditions validate the usefulness of our proposed UGS monitoring system.

Design of an Acoustic Target Intrusion Detection System Based on Small-Aperture Microphone Array.

PubMed

Zu, Xingshui; Guo, Feng; Huang, Jingchang; Zhao, Qin; Liu, Huawei; Li, Baoqing; Yuan, Xiaobing

2017-03-04

Automated surveillance of remote locations in a wireless sensor network is dominated by the detection algorithm because actual intrusions in such locations are a rare event. Therefore, a detection method with low power consumption is crucial for persistent surveillance to ensure longevity of the sensor networks. A simple and effective two-stage algorithm composed of energy detector (ED) and delay detector (DD) with all its operations in time-domain using small-aperture microphone array (SAMA) is proposed. The algorithm analyzes the quite different velocities between wind noise and sound waves to improve the detection capability of ED in the surveillance area. Experiments in four different fields with three types of vehicles show that the algorithm is robust to wind noise and the probability of detection and false alarm are 96.67% and 2.857%, respectively.

Effectiveness of the Directional Microphone in the Baha® Divino™

PubMed Central

Oeding, Kristi; Valente, Michael; Kerckhoff, Jessica

2010-01-01

Background Patients with unilateral sensorineural hearing loss (USNHL) experience great difficulty listening to speech in noisy environments. A directional microphone (DM) could potentially improve speech recognition in this difficult listening environment. It is well known that DMs in behind-the-ear (BTE) and custom hearing aids can provide a greater signal-to-noise ratio (SNR) in comparison to an omnidirectional microphone (OM) to improve speech recognition in noise for persons with hearing impairment. Studies examining the DM in bone anchored auditory osseointegrated implants (Baha), however, have been mixed, with little to no benefit reported for the DM compared to an OM. Purpose The primary purpose of this study was to determine if there are statistically significant differences in the mean reception threshold for sentences (RTS in dB) in noise between the OM and DM in the Baha® Divino™. The RTS of these two microphone modes was measured utilizing two loudspeaker arrays (speech from 0° and noise from 180° or a diffuse eight-loudspeaker array) and with the better ear open or closed with an earmold impression and noise attenuating earmuff. Subjective benefit was assessed using the Abbreviated Profile of Hearing Aid Benefit (APHAB) to compare unaided and aided (Divino OM and DM combined) problem scores. Research Design A repeated measures design was utilized, with each subject counterbalanced to each of the eight treatment levels for three independent variables: (1) microphone (OM and DM), (2) loudspeaker array (180° and diffuse), and (3) better ear (open and closed). Study Sample Sixteen subjects with USNHL currently utilizing the Baha were recruited from Washington University’s Center for Advanced Medicine and the surrounding area. Data Collection and Analysis Subjects were tested at the initial visit if they entered the study wearing the Divino or after at least four weeks of acclimatization to a loaner Divino. The RTS was determined utilizing Hearing in Noise Test (HINT) sentences in the R-Space™ system, and subjective benefit was determined utilizing the APHAB. A three-way repeated measures analysis of variance (ANOVA) and a paired samples t-test were utilized to analyze results of the HINT and APHAB, respectively. Results Results revealed statistically significant differences within microphone (p < 0.001; directional advantage of 3.2 dB), loudspeaker array (p = 0.046; 180° advantage of 1.1 dB), and better ear conditions (p < 0.001; open ear advantage of 4.9 dB). Results from the APHAB revealed statistically and clinically significant benefit for the Divino relative to unaided on the subscales of Ease of Communication (EC) (p = 0.037), Background Noise (BN) (p < 0.001), and Reverberation (RV) (p = 0.005). Conclusions The Divino’s DM provides a statistically significant improvement in speech recognition in noise compared to the OM for subjects with USNHL. Therefore, it is recommended that audiologists consider selecting a Baha with a DM to provide improved speech recognition performance in noisy listening environments. PMID:21034701

Effectiveness of the directional microphone in the Baha® Divino™.

PubMed

Oeding, Kristi; Valente, Michael; Kerckhoff, Jessica

2010-09-01

Patients with unilateral sensorineural hearing loss (USNHL) experience great difficulty listening to speech in noisy environments. A directional microphone (DM) could potentially improve speech recognition in this difficult listening environment. It is well known that DMs in behind-the-ear (BTE) and custom hearing aids can provide a greater signal-to-noise ratio (SNR) in comparison to an omnidirectional microphone (OM) to improve speech recognition in noise for persons with hearing impairment. Studies examining the DM in bone anchored auditory osseointegrated implants (Baha), however, have been mixed, with little to no benefit reported for the DM compared to an OM. The primary purpose of this study was to determine if there are statistically significant differences in the mean reception threshold for sentences (RTS in dB) in noise between the OM and DM in the Baha® Divino™. The RTS of these two microphone modes was measured utilizing two loudspeaker arrays (speech from 0° and noise from 180° or a diffuse eight-loudspeaker array) and with the better ear open or closed with an earmold impression and noise attenuating earmuff. Subjective benefit was assessed using the Abbreviated Profile of Hearing Aid Benefit (APHAB) to compare unaided and aided (Divino OM and DM combined) problem scores. A repeated measures design was utilized, with each subject counterbalanced to each of the eight treatment levels for three independent variables: (1) microphone (OM and DM), (2) loudspeaker array (180° and diffuse), and (3) better ear (open and closed). Sixteen subjects with USNHL currently utilizing the Baha were recruited from Washington University's Center for Advanced Medicine and the surrounding area. Subjects were tested at the initial visit if they entered the study wearing the Divino or after at least four weeks of acclimatization to a loaner Divino. The RTS was determined utilizing Hearing in Noise Test (HINT) sentences in the R-Space™ system, and subjective benefit was determined utilizing the APHAB. A three-way repeated measures analysis of variance (ANOVA) and a paired samples t-test were utilized to analyze results of the HINT and APHAB, respectively. Results revealed statistically significant differences within microphone (p < 0.001; directional advantage of 3.2 dB), loudspeaker array (p = 0.046; 180° advantage of 1.1 dB), and better ear conditions (p < 0.001; open ear advantage of 4.9 dB). Results from the APHAB revealed statistically and clinically significant benefit for the Divino relative to unaided on the subscales of Ease of Communication (EC) (p = 0.037), Background Noise (BN) (p < 0.001), and Reverberation (RV) (p = 0.005). The Divino's DM provides a statistically significant improvement in speech recognition in noise compared to the OM for subjects with USNHL. Therefore, it is recommended that audiologists consider selecting a Baha with a DM to provide improved speech recognition performance in noisy listening environments. American Academy of Audiology.

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

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J. (Inventor)

1979-01-01

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

Expansions for infinite or finite plane circular time-reversal mirrors and acoustic curtains for wave-field-synthesis.

PubMed

Mellow, Tim; Kärkkäinen, Leo

2014-03-01

An acoustic curtain is an array of microphones used for recording sound which is subsequently reproduced through an array of loudspeakers in which each loudspeaker reproduces the signal from its corresponding microphone. Here the sound originates from a point source on the axis of symmetry of the circular array. The Kirchhoff-Helmholtz integral for a plane circular curtain is solved analytically as fast-converging expansions, assuming an ideal continuous array, to speed up computations and provide insight. By reversing the time sequence of the recording (or reversing the direction of propagation of the incident wave so that the point source becomes an "ideal" point sink), the curtain becomes a time reversal mirror and the analytical solution for this is given simultaneously. In the case of an infinite planar array, it is demonstrated that either a monopole or dipole curtain will reproduce the diverging sound field of the point source on the far side. However, although the real part of the sound field of the infinite time-reversal mirror is reproduced, the imaginary part is an approximation due to the missing singularity. It is shown that the approximation may be improved by using the appropriate combination of monopole and dipole sources in the mirror.

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

PubMed Central

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

2008-01-01

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

Acoustic Source Elevation Angle Estimates Using Two Microphones

DTIC Science & Technology

2014-06-01

elevated. Elevation angles are successfully estimated, under certain conditions, for a loudspeaker broadcasting band limited white noise. 15. SUBJECT...INTENTIONALLY LEFT BLANK. 1 1. Introduction The U.S. Army uses acoustic arrays to track and locate various sources including...ground and airborne vehicles, small arms, mortars, and rockets. The tracking and locating algorithms often used with these acoustic arrays perform

Jet Noise Source Localization Using Linear Phased Array

NASA Technical Reports Server (NTRS)

Agboola, Ferni A.; Bridges, James

2004-01-01

A study was conducted to further clarify the interpretation and application of linear phased array microphone results, for localizing aeroacoustics sources in aircraft exhaust jet. Two model engine nozzles were tested at varying power cycles with the array setup parallel to the jet axis. The array position was varied as well to determine best location for the array. The results showed that it is possible to resolve jet noise sources with bypass and other components separation. The results also showed that a focused near field image provides more realistic noise source localization at low to mid frequencies.

Improving acoustic beamforming maps in a reverberant environment by modifying the cross-correlation matrix

NASA Astrophysics Data System (ADS)

Fischer, J.; Doolan, C.

2017-12-01

A method to improve the quality of acoustic beamforming in reverberant environments is proposed in this paper. The processing is based on a filtering of the cross-correlation matrix of the microphone signals obtained using a microphone array. The main advantage of the proposed method is that it does not require information about the geometry of the reverberant environment and thus it can be applied to any configuration. The method is applied to the particular example of aeroacoustic testing in a hard-walled low-speed wind tunnel; however, the technique can be used in any reverberant environment. Two test cases demonstrate the technique. The first uses a speaker placed in the hard-walled working section with no wind tunnel flow. In the second test case, an airfoil is placed in a flow and acoustic beamforming maps are obtained. The acoustic maps have been improved, as the reflections observed in the conventional maps have been removed after application of the proposed method.

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.

The Detection of Radiated Modes from Ducted Fan Engines

NASA Technical Reports Server (NTRS)

Farassat, F.; Nark, Douglas M.; Thomas, Russell H.

2001-01-01

The bypass duct of an aircraft engine is a low-pass filter allowing some spinning modes to radiate outside the duct. The knowledge of the radiated modes can help in noise reduction, as well as the diagnosis of noise generation mechanisms inside the duct. We propose a nonintrusive technique using a circular microphone array outside the engine measuring the complex noise spectrum on an arc of a circle. The array is placed at various axial distances from the inlet or the exhaust of the engine. Using a model of noise radiation from the duct, an overdetermined system of linear equations is constructed for the complex amplitudes of the radial modes for a fixed circumferential mode. This system of linear equations is generally singular, indicating that the problem is illposed. Tikhonov regularization is employed to solve this system of equations for the unknown amplitudes of the radiated modes. An application of our mode detection technique using measured acoustic data from a circular microphone array is presented. We show that this technique can reliably detect radiated modes with the possible exception of modes very close to cut-off.

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

NASA Astrophysics Data System (ADS)

Yousefian Jazi, Nima

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

Acoustic Imaging of Snowpack Physical Properties

NASA Astrophysics Data System (ADS)

Kinar, N. J.; Pomeroy, J. W.

2011-12-01

Measurements of snowpack depth, density, structure and temperature have often been conducted by the use of snowpits and invasive measurement devices. Previous research has shown that acoustic waves passing through snow are capable of measuring these properties. An experimental observation device (SAS2, System for the Acoustic Sounding of Snow) was used to autonomously send audible sound waves into the top of the snowpack and to receive and process the waves reflected from the interior and bottom of the snowpack. A loudspeaker and microphone array separated by an offset distance was suspended in the air above the surface of the snowpack. Sound waves produced from a loudspeaker as frequency-swept sequences and maximum length sequences were used as source signals. Up to 24 microphones measured the audible signal from the snowpack. The signal-to-noise ratio was compared between sequences in the presence of environmental noise contributed by wind and reflections from vegetation. Beamforming algorithms were used to reject spurious reflections and to compensate for movement of the sensor assembly during the time of data collection. A custom-designed circuit with digital signal processing hardware implemented an inversion algorithm to relate the reflected sound wave data to snowpack physical properties and to create a two-dimensional image of snowpack stratigraphy. The low power consumption circuit was powered by batteries and through WiFi and Bluetooth interfaces enabled the display of processed data on a mobile device. Acoustic observations were logged to an SD card after each measurement. The SAS2 system was deployed at remote field locations in the Rocky Mountains of Alberta, Canada. Acoustic snow properties data was compared with data collected from gravimetric sampling, thermocouple arrays, radiometers and snowpit observations of density, stratigraphy and crystal structure. Aspects for further research and limitations of the acoustic sensing system are also discussed.

Infrasonic Stethoscope for Monitoring Physiological Processes

NASA Technical Reports Server (NTRS)

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

2018-01-01

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

Infrasonic Stethoscope for Monitoring Physiological Processes

NASA Technical Reports Server (NTRS)

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

2016-01-01

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

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...

Assessment of Railroad Locomotive Noise

DOT National Transportation Integrated Search

1976-08-01

Measurements of the noise generated by an SD40-2 diesel electric locomotive are described. The noise was measured in three types of moving tests: the first with the locomotive passing a 6-microphone array while under maximum power acceleration, the s...

Passive wake acoustics measurements at Denver International Airport

DOT National Transportation Integrated Search

2004-04-26

From August to September 2003, NASA conducted an extensive measurement campaign to characterize the acoustic signal of wake vortices. A large, both spatially as well as in number of elements, phased microphone array was deployed at Denver Internation...

Design and Implementation of Sound Searching Robots in Wireless Sensor Networks

PubMed Central

Han, Lianfu; Shen, Zhengguang; Fu, Changfeng; Liu, Chao

2016-01-01

A sound target-searching robot system which includes a 4-channel microphone array for sound collection, magneto-resistive sensor for declination measurement, and a wireless sensor networks (WSN) for exchanging information is described. It has an embedded sound signal enhancement, recognition and location method, and a sound searching strategy based on a digital signal processor (DSP). As the wireless network nodes, three robots comprise the WSN a personal computer (PC) in order to search the three different sound targets in task-oriented collaboration. The improved spectral subtraction method is used for noise reduction. As the feature of audio signal, Mel-frequency cepstral coefficient (MFCC) is extracted. Based on the K-nearest neighbor classification method, we match the trained feature template to recognize sound signal type. This paper utilizes the improved generalized cross correlation method to estimate time delay of arrival (TDOA), and then employs spherical-interpolation for sound location according to the TDOA and the geometrical position of the microphone array. A new mapping has been proposed to direct the motor to search sound targets flexibly. As the sink node, the PC receives and displays the result processed in the WSN, and it also has the ultimate power to make decision on the received results in order to improve their accuracy. The experiment results show that the designed three-robot system implements sound target searching function without collisions and performs well. PMID:27657088

Design and Implementation of Sound Searching Robots in Wireless Sensor Networks.

PubMed

Han, Lianfu; Shen, Zhengguang; Fu, Changfeng; Liu, Chao

2016-09-21

A sound target-searching robot system which includes a 4-channel microphone array for sound collection, magneto-resistive sensor for declination measurement, and a wireless sensor networks (WSN) for exchanging information is described. It has an embedded sound signal enhancement, recognition and location method, and a sound searching strategy based on a digital signal processor (DSP). As the wireless network nodes, three robots comprise the WSN a personal computer (PC) in order to search the three different sound targets in task-oriented collaboration. The improved spectral subtraction method is used for noise reduction. As the feature of audio signal, Mel-frequency cepstral coefficient (MFCC) is extracted. Based on the K-nearest neighbor classification method, we match the trained feature template to recognize sound signal type. This paper utilizes the improved generalized cross correlation method to estimate time delay of arrival (TDOA), and then employs spherical-interpolation for sound location according to the TDOA and the geometrical position of the microphone array. A new mapping has been proposed to direct the motor to search sound targets flexibly. As the sink node, the PC receives and displays the result processed in the WSN, and it also has the ultimate power to make decision on the received results in order to improve their accuracy. The experiment results show that the designed three-robot system implements sound target searching function without collisions and performs well.

Echolocating bats use future-target information for optimal foraging.

PubMed

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.

Phased-Array Study of Dual-Flow Jet Noise: Effect of Nozzles and Mixers

NASA Technical Reports Server (NTRS)

Soo Lee, Sang; Bridges, James

2006-01-01

A 16-microphone linear phased-array installed parallel to the jet axis and a 32-microphone azimuthal phased-array installed in the nozzle exit plane have been applied to identify the noise source distributions of nozzle exhaust systems with various internal mixers (lobed and axisymmetric) and nozzles (three different lengths). Measurements of velocity were also obtained using cross-stream stereo particle image velocimetry (PIV). Among the three nozzle lengths tested, the medium length nozzle was the quietest for all mixers at high frequency on the highest speed flow condition. Large differences in source strength distributions between nozzles and mixers occurred at or near the nozzle exit for this flow condition. The beamforming analyses from the azimuthal array for the 12-lobed mixer on the highest flow condition showed that the core flow and the lobe area were strong noise sources for the long and short nozzles. The 12 noisy spots associated with the lobe locations of the 12-lobed mixer with the long nozzle were very well detected for the frequencies 5 KHz and higher. Meanwhile, maps of the source strength of the axisymmetric splitter show that the outer shear layer was the most important noise source at most flow conditions. In general, there was a good correlation between the high turbulence regions from the PIV tests and the high noise source regions from the phased-array measurements.

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.

A Robust Sound Source Localization Approach for Microphone Array with Model Errors

NASA Astrophysics Data System (ADS)

Xiao, Hua; Shao, Huai-Zong; Peng, Qi-Cong

In this paper, a robust sound source localization approach is proposed. The approach retains good performance even when model errors exist. Compared with previous work in this field, the contributions of this paper are as follows. First, an improved broad-band and near-field array model is proposed. It takes array gain, phase perturbations into account and is based on the actual positions of the elements. It can be used in arbitrary planar geometry arrays. Second, a subspace model errors estimation algorithm and a Weighted 2-Dimension Multiple Signal Classification (W2D-MUSIC) algorithm are proposed. The subspace model errors estimation algorithm estimates unknown parameters of the array model, i. e., gain, phase perturbations, and positions of the elements, with high accuracy. The performance of this algorithm is improved with the increasing of SNR or number of snapshots. The W2D-MUSIC algorithm based on the improved array model is implemented to locate sound sources. These two algorithms compose the robust sound source approach. The more accurate steering vectors can be provided for further processing such as adaptive beamforming algorithm. Numerical examples confirm effectiveness of this proposed approach.

Acoustic Localization with Infrasonic Signals

NASA Astrophysics Data System (ADS)

Threatt, Arnesha; Elbing, Brian

2015-11-01

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

Wake Vortex Avoidance System and Method

NASA Technical Reports Server (NTRS)

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

2017-01-01

A wake vortex avoidance system includes a microphone array configured to detect low frequency sounds. A signal processor determines a geometric mean coherence based on the detected low frequency sounds. A display displays wake vortices based on the determined geometric mean coherence.

Jet-Surface Interaction Test: Phased Array Noise Source Localization Results

NASA Technical Reports Server (NTRS)

Podboy, Gary G.

2013-01-01

An experiment was conducted to investigate the effect that a planar surface located near a jet flow has on the noise radiated to the far-field. Two different configurations were tested: 1) a shielding configuration in which the surface was located between the jet and the far-field microphones, and 2) a reflecting configuration in which the surface was mounted on the opposite side of the jet, and thus the jet noise was free to reflect off the surface toward the microphones. Both conventional far-field microphone and phased array noise source localization measurements were obtained. This paper discusses phased array results, while a companion paper (Brown, C.A., "Jet-Surface Interaction Test: Far-Field Noise Results," ASME paper GT2012-69639, June 2012.) discusses far-field results. The phased array data show that the axial distribution of noise sources in a jet can vary greatly depending on the jet operating condition and suggests that it would first be necessary to know or be able to predict this distribution in order to be able to predict the amount of noise reduction to expect from a given shielding configuration. The data obtained on both subsonic and supersonic jets show that the noise sources associated with a given frequency of noise tend to move downstream, and therefore, would become more difficult to shield, as jet Mach number increases. The noise source localization data obtained on cold, shock-containing jets suggests that the constructive interference of sound waves that produces noise at a given frequency within a broadband shock noise hump comes primarily from a small number of shocks, rather than from all the shocks at the same time. The reflecting configuration data illustrates that the law of reflection must be satisfied in order for jet noise to reflect off of a surface to an observer, and depending on the relative locations of the jet, the surface, and the observer, only some of the jet noise sources may satisfy this requirement.

Jet-Surface Interaction Test: Phased Array Noise Source Localization Results

NASA Technical Reports Server (NTRS)

Podboy, Gary G.

2012-01-01

An experiment was conducted to investigate the effect that a planar surface located near a jet flow has on the noise radiated to the far-field. Two different configurations were tested: 1) a shielding configuration in which the surface was located between the jet and the far-field microphones, and 2) a reflecting configuration in which the surface was mounted on the opposite side of the jet, and thus the jet noise was free to reflect off the surface toward the microphones. Both conventional far-field microphone and phased array noise source localization measurements were obtained. This paper discusses phased array results, while a companion paper discusses far-field results. The phased array data show that the axial distribution of noise sources in a jet can vary greatly depending on the jet operating condition and suggests that it would first be necessary to know or be able to predict this distribution in order to be able to predict the amount of noise reduction to expect from a given shielding configuration. The data obtained on both subsonic and supersonic jets show that the noise sources associated with a given frequency of noise tend to move downstream, and therefore, would become more difficult to shield, as jet Mach number increases. The noise source localization data obtained on cold, shock-containing jets suggests that the constructive interference of sound waves that produces noise at a given frequency within a broadband shock noise hump comes primarily from a small number of shocks, rather than from all the shocks at the same time. The reflecting configuration data illustrates that the law of reflection must be satisfied in order for jet noise to reflect off of a surface to an observer, and depending on the relative locations of the jet, the surface, and the observer, only some of the jet noise sources may satisfy this requirement.

Detection of infrasound generated by Clear Air Turbulence (CAT)

NASA Technical Reports Server (NTRS)

Meredith, R.; Badavi, F.; Becher, J.

1981-01-01

A unified data acquisiton system which is an AM carrier system consisting of a converter signal, conditioning electronics, and peripheral equipment was checked and calibrated and installed in a mobile van. A microphone array in the shape of a 244-m equilateral triangle was connected to the data acquisition system using 457-m long cables. Development of techniques for signal processing for interpreting the infrasonic signature (differentiating between CAT and other sources of infrasound) is summarized. Once patches of CAT are located in the atmosphere, corroboration can be achieved through test flights of aircraft into the suspected region.

Design and Evaluation of a Scalable and Reconfigurable Multi-Platform System for Acoustic Imaging

PubMed Central

Izquierdo, Alberto; Villacorta, Juan José; del Val Puente, Lara; Suárez, Luis

2016-01-01

This paper proposes a scalable and multi-platform framework for signal acquisition and processing, which allows for the generation of acoustic images using planar arrays of MEMS (Micro-Electro-Mechanical Systems) microphones with low development and deployment costs. Acoustic characterization of MEMS sensors was performed, and the beam pattern of a module, based on an 8 × 8 planar array and of several clusters of modules, was obtained. A flexible framework, formed by an FPGA, an embedded processor, a computer desktop, and a graphic processing unit, was defined. The processing times of the algorithms used to obtain the acoustic images, including signal processing and wideband beamforming via FFT, were evaluated in each subsystem of the framework. Based on this analysis, three frameworks are proposed, defined by the specific subsystems used and the algorithms shared. Finally, a set of acoustic images obtained from sound reflected from a person are presented as a case study in the field of biometric identification. These results reveal the feasibility of the proposed system. PMID:27727174

Methods for determining infrasound phase velocity direction with an array of line sensors.

PubMed

Walker, Kristoffer T; Zumberge, Mark A; Hedlin, Michael A H; Shearer, Peter M

2008-10-01

Infrasound arrays typically consist of several microbarometers separated by distances that provide predictable signal time separations, forming the basis for processing techniques that estimate the phase velocity direction. The directional resolution depends on the noise level and is proportional to the number of these point sensors; additional sensors help attenuate noise and improve direction resolution. An alternative approach is to form an array of directional line sensors, each of which emulates a line of many microphones that instantaneously integrate pressure change. The instrument response is a function of the orientation of the line with respect to the signal wavefront. Real data recorded at the Piñon Flat Observatory in southern California and synthetic data show that this spectral property can be exploited with multiple line sensors to determine the phase velocity direction with a precision comparable to a larger aperture array of microbarometers. Three types of instrument-response-dependent beamforming and an array deconvolution technique are evaluated. The results imply that an array of five radial line sensors, with equal azimuthal separation and an aperture that depends on the frequency band of interest, provides directional resolution while requiring less space compared to an equally effective array of five microbarometers with rosette wind filters.

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

NASA Astrophysics Data System (ADS)

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

2018-01-01

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

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

PubMed

Beck, Christoph; Garreau, Guillaume; Georgiou, Julius

2016-01-01

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

Acoustic 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.

Active room compensation for sound reinforcement using sound field separation techniques.

PubMed

Heuchel, Franz M; Fernandez-Grande, Efren; Agerkvist, Finn T; Shabalina, Elena

2018-03-01

This work investigates how the sound field created by a sound reinforcement system can be controlled at low frequencies. An indoor control method is proposed which actively absorbs the sound incident on a reflecting boundary using an array of secondary sources. The sound field is separated into incident and reflected components by a microphone array close to the secondary sources, enabling the minimization of reflected components by means of optimal signals for the secondary sources. The method is purely feed-forward and assumes constant room conditions. Three different sound field separation techniques for the modeling of the reflections are investigated based on plane wave decomposition, equivalent sources, and the Spatial Fourier transform. Simulations and an experimental validation are presented, showing that the control method performs similarly well at enhancing low frequency responses with the three sound separation techniques. Resonances in the entire room are reduced, although the microphone array and secondary sources are confined to a small region close to the reflecting wall. Unlike previous control methods based on the creation of a plane wave sound field, the investigated method works in arbitrary room geometries and primary source positions.

Spherical beamforming for spherical array with impedance surface

NASA Astrophysics Data System (ADS)

Tontiwattanakul, Khemapat

2018-01-01

Spherical microphone array beamforming has been a popular research topic for recent years. Due to their isotropic beam in three dimensional spaces as well as a certain frequency range, the arrays are widely used in many applications such as sound field recording, acoustic beamforming, and noise source localisation. The body of a spherical array is usually considered perfectly rigid. A sound field captured by the sensors on spherical array can be decomposed into a series of spherical harmonics. In noise source localisation, the amplitude density of sound sources is estimated and illustrated by mean of colour maps. In this work, a rigid spherical array covered by fibrous materials is studied via numerical simulation and the performance of the spherical beamforming is discussed.

Theoretical and experimental study of DOA estimation using AML algorithm for an isotropic and non-isotropic 3D array

NASA Astrophysics Data System (ADS)

Asgari, Shadnaz; Ali, Andreas M.; Collier, Travis C.; Yao, Yuan; Hudson, Ralph E.; Yao, Kung; Taylor, Charles E.

2007-09-01

The focus of most direction-of-arrival (DOA) estimation problems has been based mainly on a two-dimensional (2D) scenario where we only need to estimate the azimuth angle. But in various practical situations we have to deal with a three-dimensional scenario. The importance of being able to estimate both azimuth and elevation angles with high accuracy and low complexity is of interest. We present the theoretical and the practical issues of DOA estimation using the Approximate-Maximum-Likelihood (AML) algorithm in a 3D scenario. We show that the performance of the proposed 3D AML algorithm converges to the Cramer-Rao Bound. We use the concept of an isotropic array to reduce the complexity of the proposed algorithm by advocating a decoupled 3D version. We also explore a modified version of the decoupled 3D AML algorithm which can be used for DOA estimation with non-isotropic arrays. Various numerical results are presented. We use two acoustic arrays each consisting of 8 microphones to do some field measurements. The processing of the measured data from the acoustic arrays for different azimuth and elevation angles confirms the effectiveness of the proposed methods.

Implementation of the Phase Difference Trace Function for a Circular Array.

DTIC Science & Technology

1980-06-01

15 B. HICROPHONES............... ... .. ..... ....... 16 C. PREAMPLIFIER /FILTER ................. o....... 16 D. PHASE SHIFrER...Photgraphs ......................... 31 5. Microphone Specifications ........................... 32 6. Preamplifier /Filter Schematic...33 7. Preamplifier /Filter Printel Cir-cait Board ........... 34 8. Phase Shifter Schemitic ............................. 35 9. Phase Shifter Print.l

Inconspicuous echolocation in hoary bats (Lasiurus cinereus)

Treesearch

Aaron J. Corcoran; Theodore J. Weller

2018-01-01

Echolocation allows bats to occupy diverse nocturnal niches. Bats almost always use echolocation, even when other sensory stimuli are available to guide navigation. Here, using arrays of calibrated infrared cameras and ultrasonic microphones, we demonstrate that hoary bats (Lasiurus cinereus) use previously unknown echolocation behaviours that...

Lightning characterization through acoustic and electromagnetic measurements recorded during the HyMeX SOP1 and simulation of the acoustic nonlinear propagation in realistic thunderstorm meteorological conditions

NASA Astrophysics Data System (ADS)

Gallin, L.; Coulouvrat, F.; Farges, T.; Marchiano, R.; Defer, E.; Rison, W.; Schulz, W.; Nuret, M.

2013-12-01

The goal is to study the transformation of the thunder (amplitude, spectrum) during its travel from the lightning channel towards a detector (microphone, microbarometer), considering propagation distances of less than 50 km and complex local meteorological properties. Inside the European HyMeX project, the SOP1 campaign (2012) took place from September 2012 to November 2012 in South of France. An acoustic station (center: 4.39° E, 44.08° N) composed of a microphone array placed inside a microbarometer array was installed by CEA near city of Uzès. It was located in the center of an LMA network coming with two slow antennas. This network was deployed in France for the first time by the New Mexico Tech and LERMA laboratory. The detections from the European lightning location system EUCLID complete this dataset. During the SOP1 period several storms passed over the station. The post-processings of the records point out days with interesting thunderstorms. Especially during the 26th of October 2012 in the evening (around 8 pm) a thunderstorm passed just over the acoustic station. Not too many lightning strokes are detected by EUCLID, the corresponding flashes are then well characterized by the LMA network. Slow antennas present good electric field measurements. The acoustic records have excellent quality. We present for some selected flashes a comparative study of the different measurements (LMA, slow antenna, EUCLID, microphones, microbarometers): focusing on amplitude and spectrum of the thunder waveforms, and on propagation effects due to the meteorological conditions. To quantify the impact of these meteorological conditions on the propagating thunder (from the lightning sources to the acoustic array), a code named Flhoward is used [Dagrau et al., J. Acoust. Soc. Am., 130, 20-32, 2011][Coulouvrat, Wave Motion, 49, 50--63, 2012]. It is designed to simulate the nonlinear propagation of acoustic shock waves through a realistic atmosphere model (including temperature gradients, rigid ground, and wind flows). The meteorological conditions are extracted from the data calculated by Météo-France weather forecast model AROME-WMED for the chosen days. Some cases where numerical simulation helps to understand the observations are presented.

Speech Acquisition and Automatic Speech Recognition for Integrated Spacesuit Audio Systems

NASA Technical Reports Server (NTRS)

Huang, Yiteng; Chen, Jingdong; Chen, Shaoyan

2010-01-01

A voice-command human-machine interface system has been developed for spacesuit extravehicular activity (EVA) missions. A multichannel acoustic signal processing method has been created for distant speech acquisition in noisy and reverberant environments. This technology reduces noise by exploiting differences in the statistical nature of signal (i.e., speech) and noise that exists in the spatial and temporal domains. As a result, the automatic speech recognition (ASR) accuracy can be improved to the level at which crewmembers would find the speech interface useful. The developed speech human/machine interface will enable both crewmember usability and operational efficiency. It can enjoy a fast rate of data/text entry, small overall size, and can be lightweight. In addition, this design will free the hands and eyes of a suited crewmember. The system components and steps include beam forming/multi-channel noise reduction, single-channel noise reduction, speech feature extraction, feature transformation and normalization, feature compression, model adaption, ASR HMM (Hidden Markov Model) training, and ASR decoding. A state-of-the-art phoneme recognizer can obtain an accuracy rate of 65 percent when the training and testing data are free of noise. When it is used in spacesuits, the rate drops to about 33 percent. With the developed microphone array speech-processing technologies, the performance is improved and the phoneme recognition accuracy rate rises to 44 percent. The recognizer can be further improved by combining the microphone array and HMM model adaptation techniques and using speech samples collected from inside spacesuits. In addition, arithmetic complexity models for the major HMMbased ASR components were developed. They can help real-time ASR system designers select proper tasks when in the face of constraints in computational resources.

Assessment of Systematic Measurement Errors for Acoustic Travel-Time Tomography of the Atmosphere

DTIC Science & Technology

2013-01-01

measurements include assess- ment of the time delays in electronic circuits and mechanical hardware (e.g., drivers and microphones) of a tomography array ...hardware and electronic circuits of the tomography array and errors in synchronization of the transmitted and recorded signals. For example, if...coordinates can be as large as 30 cm. These errors are equivalent to the systematic errors in the travel times of 0.9 ms. Third, loudspeakers which are used

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.

A Comparative Study of Simulated and Measured Main Landing Gear Noise for Large Civil Transports

NASA Technical Reports Server (NTRS)

Konig, Benedikt; Fares, Ehab; Ravetta, Patricio; Khorrami, Mehdi R.

2017-01-01

Computational results for the NASA 26%-scale model of a six-wheel main landing gear with and without a toboggan-shaped noise reduction fairing are presented. The model is a high-fidelity representation of a Boeing 777-200 aircraft main landing gear. A lattice Boltzmann method was used to simulate the unsteady flow around the model in isolation. The computations were conducted in free-air at a Mach number of 0.17, matching a recent acoustic test of the same gear model in the Virginia Tech Stability Wind Tunnel in its anechoic configuration. Results obtained on a set of grids with successively finer spatial resolution demonstrate the challenge in resolving/capturing the flow field for the smaller components of the gear and their associated interactions, and the resulting effects on the high-frequency segment of the farfield noise spectrum. Farfield noise spectra were computed based on an FWH integral approach, with simulated pressures on the model solid surfaces or flow-field data extracted on a set of permeable surfaces enclosing the model as input. Comparison of these spectra with microphone array measurements obtained in the tunnel indicated that, for the present complex gear model, the permeable surfaces provide a more accurate representation of farfield noise, suggesting that volumetric effects are not negligible. The present study also demonstrates that good agreement between simulated and measured farfield noise can be achieved if consistent post-processing is applied to both physical and synthetic pressure records at array microphone locations.

On the ability of consumer electronics microphones for environmental noise monitoring.

PubMed

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.

Langley Aerospace Research Summer Scholars (LARSS) Scholars Pres

NASA Image and Video Library

2013-08-07

250 students participated in the Langley Aerospace Research Summer Scholars (LARSS) Presentations focused on 3D modeling of STARBUKS calibration components in the National Transonic Facility, hypersonic aerodynamic inflatable decelerator, and optimization of a microphone-based array for flight testing. Reid Center LaRC Hampton, VA

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

PubMed

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

2014-01-01

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

Optical microphone with fiber Bragg grating and signal processing techniques

NASA Astrophysics Data System (ADS)

Tosi, Daniele; Olivero, Massimo; Perrone, Guido

2008-06-01

In this paper, we discuss the realization of an optical microphone array using fiber Bragg gratings as sensing elements. The wavelength shift induced by acoustic waves perturbing the sensing Bragg grating is transduced into an intensity modulation. The interrogation unit is based on a fixed-wavelength laser source and - as receiver - a photodetector with proper amplification; the system has been implemented using devices for standard optical communications, achieving a low-cost interrogator. One of the advantages of the proposed approach is that no voltage-to-strain calibration is required for tracking dynamic shifts. The optical sensor is complemented by signal processing tools, including a data-dependent frequency estimator and adaptive filters, in order to improve the frequency-domain analysis and mitigate the effects of disturbances. Feasibility and performances of the optical system have been tested measuring the output of a loudspeaker. With this configuration, the sensor is capable of correctly detecting sounds up to 3 kHz, with a frequency response that exhibits a top sensitivity within the range 200-500 Hz; single-frequency input sounds inducing an axial strain higher than ~10nɛ are correctly detected. The repeatability range is ~0.1%. The sensor has also been applied for the detection of pulsed stimuli generated from a metronome.

Low-frequency interaural cross correlation discrimination in stereophonic reproduction of musical tones

NASA Astrophysics Data System (ADS)

Kim, Sungyoung; Martens, William L.

2005-04-01

By industry standard (ITU-R. Recommendation BS.775-1), multichannel stereophonic signals within the frequency range of up to 80 or 120 Hz may be mixed and delivered via a single driver (e.g., a subwoofer) without significant impairment of stereophonic sound quality. The assumption that stereophonic information within such low-frequency content is not significant was tested by measuring discrimination thresholds for changes in interaural cross-correlation (IACC) within spectral bands containing the lowest frequency components of low-pitch musical tones. Performances were recorded for three different musical instruments playing single notes ranging in fundamental frequency from 41 Hz to 110 Hz. The recordings, made using a multichannel microphone array composed of five DPA 4006 pressure microphones, were processed to produce a set of stimuli that varied in interaural cross-correlation (IACC) within a low-frequency band, but were otherwise identical in a higher-frequency band. This correlation processing was designed to have minimal effect upon other psychoacoustic variables such as loudness and timbre. The results show that changes in interaural cross correlation (IACC) within low-frequency bands of low-pitch musical tones are most easily discriminated when decorrelated signals are presented via subwoofers positioned at extreme lateral angles (far from the median plane). [Work supported by VRQ.

Maneuver Acoustic Flight Test of the Bell 430 Helicopter Data Report

NASA Technical Reports Server (NTRS)

Watts, Michael E.; Greenwood, Eric; Smith, Charles D.; Snider, Royce; Conner, David A.

2014-01-01

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

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…

Infrasonic emissions from local meteorological events: A summary of data taken throughout 1984

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J.

1986-01-01

Records of infrasonic signals, propagating through the Earth's atmosphere in the frequency band 2 to 16 Hz, were gathered on a three microphone array at Langley Research Center throughout the year 1984. Digital processing of these records fulfilled three functions: time delay estimation, based on an adaptive filter; source location, determined from the time delay estimates; and source identification, based on spectral analysis. Meteorological support was provided by significant meteorological advisories, lightning locator plots, and daily reports from the Air Weather Service. The infrasonic data are organized into four characteristic signatures, one of which is believed to contain emissions from local meteorological sources. This class of signature prevailed only on those days when major global meteorological events appeared in or near to eastern United States. Eleven case histories are examined. Practical application of the infrasonic array in a low level wing shear alert system is discussed.

SoundCompass: A Distributed MEMS Microphone Array-Based Sensor for Sound Source Localization

PubMed Central

Tiete, Jelmer; Domínguez, Federico; da Silva, Bruno; Segers, Laurent; Steenhaut, Kris; Touhafi, Abdellah

2014-01-01

Sound source localization is a well-researched subject with applications ranging from localizing sniper fire in urban battlefields to cataloging wildlife in rural areas. One critical application is the localization of noise pollution sources in urban environments, due to an increasing body of evidence linking noise pollution to adverse effects on human health. Current noise mapping techniques often fail to accurately identify noise pollution sources, because they rely on the interpolation of a limited number of scattered sound sensors. Aiming to produce accurate noise pollution maps, we developed the SoundCompass, a low-cost sound sensor capable of measuring local noise levels and sound field directionality. Our first prototype is composed of a sensor array of 52 Microelectromechanical systems (MEMS) microphones, an inertial measuring unit and a low-power field-programmable gate array (FPGA). This article presents the SoundCompass’s hardware and firmware design together with a data fusion technique that exploits the sensing capabilities of the SoundCompass in a wireless sensor network to localize noise pollution sources. Live tests produced a sound source localization accuracy of a few centimeters in a 25-m2 anechoic chamber, while simulation results accurately located up to five broadband sound sources in a 10,000-m2 open field. PMID:24463431

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

PubMed

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

1998-01-01

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

An integrated analysis-synthesis array system for spatial sound fields.

PubMed

Bai, Mingsian R; Hua, Yi-Hsin; Kuo, Chia-Hao; Hsieh, Yu-Hao

2015-03-01

An integrated recording and reproduction array system for spatial audio is presented within a generic framework akin to the analysis-synthesis filterbanks in discrete time signal processing. In the analysis stage, a microphone array "encodes" the sound field by using the plane-wave decomposition. Direction of arrival of plane-wave components that comprise the sound field of interest are estimated by multiple signal classification. Next, the source signals are extracted by using a deconvolution procedure. In the synthesis stage, a loudspeaker array "decodes" the sound field by reconstructing the plane-wave components obtained in the analysis stage. This synthesis stage is carried out by pressure matching in the interior domain of the loudspeaker array. The deconvolution problem is solved by truncated singular value decomposition or convex optimization algorithms. For high-frequency reproduction that suffers from the spatial aliasing problem, vector panning is utilized. Listening tests are undertaken to evaluate the deconvolution method, vector panning, and a hybrid approach that combines both methods to cover frequency ranges below and above the spatial aliasing frequency. Localization and timbral attributes are considered in the subjective evaluation. The results show that the hybrid approach performs the best in overall preference. In addition, there is a trade-off between reproduction performance and the external radiation.

Drive-by large-region acoustic noise-source mapping via sparse beamforming tomography.

PubMed

Tuna, Cagdas; Zhao, Shengkui; Nguyen, Thi Ngoc Tho; Jones, Douglas L

2016-10-01

Environmental noise is a risk factor for human physical and mental health, demanding an efficient large-scale noise-monitoring scheme. The current technology, however, involves extensive sound pressure level (SPL) measurements at a dense grid of locations, making it impractical on a city-wide scale. This paper presents an alternative approach using a microphone array mounted on a moving vehicle to generate two-dimensional acoustic tomographic maps that yield the locations and SPLs of the noise-sources sparsely distributed in the neighborhood traveled by the vehicle. The far-field frequency-domain delay-and-sum beamforming output power values computed at multiple locations as the vehicle drives by are used as tomographic measurements. The proposed method is tested with acoustic data collected by driving an electric vehicle with a rooftop-mounted microphone array along a straight road next to a large open field, on which various pre-recorded noise-sources were produced by a loudspeaker at different locations. The accuracy of the tomographic imaging results demonstrates the promise of this approach for rapid, low-cost environmental noise-monitoring.

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.

Evaluating the Acoustic Effect of Over-the-Rotor Foam-Metal Liner Installed on a Low Speed Fan Using Virtual Rotating Microphone Imaging

NASA Technical Reports Server (NTRS)

Sutliff, Daniel L.; Dougherty, Robert P.; Walker, Bruce E.

2010-01-01

An in-duct beamforming technique for imaging rotating broadband fan sources has been used to evaluate the acoustic characteristics of a Foam-Metal Liner installed over-the-rotor of a low-speed fan. The NASA Glenn Research Center s Advanced Noise Control Fan was used as a test bed. A duct wall-mounted phased array consisting of several rings of microphones was employed. The data are mathematically resampled in the fan rotating reference frame and subsequently used in a conventional beamforming technique. The steering vectors for the beamforming technique are derived from annular duct modes, so that effects of reflections from the duct walls are reduced.

Phase-Based Adaptive Estimation of Magnitude-Squared Coherence Between Turbofan Internal Sensors and Far-Field Microphone Signals

NASA Technical Reports Server (NTRS)

Miles, Jeffrey Hilton

2015-01-01

A cross-power spectrum phase based adaptive technique is discussed which iteratively determines the time delay between two digitized signals that are coherent. The adaptive delay algorithm belongs to a class of algorithms that identifies a minimum of a pattern matching function. The algorithm uses a gradient technique to find the value of the adaptive delay that minimizes a cost function based in part on the slope of a linear function that fits the measured cross power spectrum phase and in part on the standard error of the curve fit. This procedure is applied to data from a Honeywell TECH977 static-engine test. Data was obtained using a combustor probe, two turbine exit probes, and far-field microphones. Signals from this instrumentation are used estimate the post-combustion residence time in the combustor. Comparison with previous studies of the post-combustion residence time validates this approach. In addition, the procedure removes the bias due to misalignment of signals in the calculation of coherence which is a first step in applying array processing methods to the magnitude squared coherence data. The procedure also provides an estimate of the cross-spectrum phase-offset.

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

PubMed

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

2015-01-01

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

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

PubMed Central

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

2016-01-01

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

Infrasound research at Kola Regional Seismological Centre, Russia

NASA Astrophysics Data System (ADS)

Asming, Vladimir; Kremenetskaya, Elena

2013-04-01

A small-aperture infrasound array has been installed in Kola Peninsula, Russia 17 km far from the town of Apatity in the year 2000. It comprises 3 Chaparral V microbarographs placed closely to the APA seismic array sensors and equipped with pipe wind reducing filters. The data are digitized at the array site and transmitted in real time to a processing center in Apatity. To search for infrasound events (arrivals of coherent signals) a beamforming-style detector has been developed. Now it works in near real time. We analyzed the detecting statistics for different frequency bands. Most man-made events are detected in 1-5 Hz band, microbaromes are typically detected in 0.2-1 Hz band. In lower frequencies we record mostly a wind noise. A data base of samples of infrasound signals of different natures has been collected. It contains recordings of microbaromes, industrial and military explosions, airplane shock waves, infrasound of airplanes, thunders, rocket launches and reentries, bolides etc. The most distant signals we have detected are associated with Kursk Magnetic Anomaly explosions (1700 km far from Apatity). We implemented an algorithm for association of infrasound signals and preliminary location of infrasound events by several arrays. It was tested with Apatity data together with data of Sweden - Finnish infrasound network operated by the Institute of Space Physics in Umea (Sweden). By agreement with NORSAR we have a real-time access to the data of Norwegian experimental infrasound installation situated in Karasjok (North Norway). Currently our detection and location programs work both with Apatity and Norwegian data. The results are available in Internet. Finnish militaries routinely destroy out-of-date weapon in autumns at the same compact site in North Finland. This is a great source of repeating infrasound signals of the same magnitude and origin. We recorded several hundreds of such explosions. The signals have been used for testing our location routines. Some factors were observed enabling or disabling first (tropospheric) arrivals of such signals depending on weather conditions. Systematic backazimuth deviations for stratospheric arrivals have been observed caused by strong stratospheric winds. In 2009 mobile infrasound arrays were developed in KRSC. Each array comprises 3 low-frequency microphones, GPS, digitizer and PC with data acquisition system. Aperture of such arrays is about 250 m, deployment time is less than 1 hour. These arrays are used in experimental work with Roskosmos space agency to search space debris reentering places. In 2012 a wireless version of such mobile array was created. Each acquisition point comprises a microphone, GPS and ADC chips, microcontroller and radio modem to send data to a central unit. This enabled us to increase aperture (up to 500 m) and decrease deployment time.

KSC-00pp1758

NASA Image and Video Library

2000-11-27

After arriving at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Pilot Michael Bloomfield. Behind him can be seen Mission Specialists Joseph Tanner and Carlos Noriega. Mission STS-97 is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

Preliminary Analysis of Acoustic Measurements from the NASA-Gulfstream Airframe Noise Flight Test

NASA Technical Reports Server (NTRS)

Khorrami, Mehdi R.; Lockhard, David D.; Humphreys, Willliam M.; Choudhari, Meelan M.; Van De Ven, Thomas

2008-01-01

The NASA-Gulfstream joint Airframe Noise Flight Test program was conducted at the NASA Wallops Flight Facility during October, 2006. The primary objective of the AFN flight test was to acquire baseline airframe noise data on a regional jet class of transport in order to determine noise source strengths and distributions for model validation. To accomplish this task, two measuring systems were used: a ground-based microphone array and individual microphones. Acoustic data for a Gulfstream G550 aircraft were acquired over the course of ten days. Over twenty-four test conditions were flown. The test matrix was designed to provide an acoustic characterization of both the full aircraft and individual airframe components and included cruise to landing configurations. Noise sources were isolated by selectively deploying individual components (flaps, main landing gear, nose gear, spoilers, etc.) and altering the airspeed, glide path, and engine settings. The AFN flight test program confirmed that the airframe is a major contributor to the noise from regional jets during landing operations. Sound pressure levels from the individual microphones on the ground revealed the flap system to be the dominant airframe noise source for the G550 aircraft. The corresponding array beamform maps showed that most of the radiated sound from the flaps originates from the side edges. Using velocity to the sixth power and Strouhal scaling of the sound pressure spectra obtained at different speeds failed to collapse the data into a single spectrum. The best data collapse was obtained when the frequencies were left unscaled.

Mic Flocks in the Cloud: Harnessing Mobile Ubiquitous Sensor Networks

NASA Astrophysics Data System (ADS)

Garces, M. A.; Christe, A.

2015-12-01

Smartphones provide a commercial, off-the-shelf solution to capture, store, analyze, and distribute infrasound using on-board or external microphones (mics) as well as on-board barometers. Free iOS infrasound apps can be readily downloaded from the Apple App Store, and Android versions are in progress. Infrasound propagates for great distances, has low sample rates, and provides a tractable pilot study scenario for open distributed sensor networks at regional and global scales using one of the most ubiquitous sensors on Earth - microphones. Data collection is no longer limited to selected vendors at exclusive prices: anybody on Earth can record and stream infrasound, and the diversity of recording systems and environments is rapidly expanding. Global deployment may be fast and easy (www.redvox.io), but comes with the cost of increasing data volume, velocity, variety, and complexity. Flocking - the collective motion of mobile agents - is a natural human response to threats or events of interest. Anticipating, modeling and harnessing flocking sensor topologies will be necessary for adaptive array and network processing. The increasing data quantity and complexity will exceed the processing capacity of human analysts and most research servers. We anticipate practical real-time applications will require the on-demand adaptive scalability and resources of the Cloud. Cloud architectures for such heterogeneous sensor networks will consider eventual integration into the Global Earth Observation System of Systems (GEOSS).

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.

Comparison of Different Measurement Technologies for the In-Flight Assessment of Radiated Acoustic Intensity

NASA Technical Reports Server (NTRS)

Klos, Jacob; Palumbo, Daniel L.; Buehrle, Ralph D.; Williams, Earl G.; Valdivia, Nicolas; Herdic, Peter C.; Sklanka, Bernard

2005-01-01

A series of tests was planned and conducted in the Interior Noise Test Facility at Boeing Field, on the NASA Aries 757 flight research aircraft, and in the Structural Acoustic Loads and Transmission Facility at NASA Langley Research Center. These tests were designed to answer several questions concerning the use of array methods in flight. One focus of the tests was determining whether and to what extent array methods could be used to identify the effects of an acoustical treatment applied to a limited portion of an aircraft fuselage. Another focus of the tests was to verify that the arrays could be used to localize and quantify a known source purposely placed in front of the arrays. Thus the issues related to backside sources and flanking paths present in the complicated sound field were addressed during these tests. These issues were addressed through the use of reference transducers, both accelerometers mounted to the fuselage and microphones in the cabin, that were used to correlate the pressure holograms. measured by the microphone arrays using either SVD methods or partial coherence methods. This correlation analysis accepts only energy that is coherent with the sources sensed by the reference transducers, allowing a noise control engineer to only identify and study those vibratory sources of interest. The remainder of this paper will present a detailed description of the test setups that were used in this test sequence and typical results of the NAH/IBEM analysis used to reconstruct the sound fields. Also, a comparison of data obtained in the laboratory environments and during flights of the 757 aircraft will be made.

Multidirectional seismo-acoustic wavefield of strombolian explosions at Yasur, Vanuatu using a broadband seismo-acoustic network, infrasound arrays, and infrasonic sensors on tethered balloons

NASA Astrophysics Data System (ADS)

Matoza, R. S.; Jolly, A. D.; Fee, D.; Johnson, R.; Kilgour, G.; Christenson, B. W.; Garaebiti, E.; Iezzi, A. M.; Austin, A.; Kennedy, B.; Fitzgerald, R.; Key, N.

2016-12-01

Seismo-acoustic wavefields at volcanoes contain rich information on shallow magma transport and subaerial eruption processes. Acoustic wavefields from eruptions are predicted to be directional, but sampling this wavefield directivity is challenging because infrasound sensors are usually deployed on the ground surface. We attempt to overcome this observational limitation using a novel deployment of infrasound sensors on tethered balloons in tandem with a suite of dense ground-based seismo-acoustic, geochemical, and eruption imaging instrumentation. We present preliminary results from a field experiment at Yasur Volcano, Vanuatu from July 26th to August 4th 2016. Our observations include data from a temporary network of 11 broadband seismometers, 6 single infrasonic microphones, 7 small-aperture 3-element infrasound arrays, 2 infrasound sensor packages on tethered balloons, an FTIR, a FLIR, 2 scanning Flyspecs, and various visual imaging data. An introduction to the dataset and preliminary analysis of the 3D seismo-acoustic wavefield and source process will be presented. This unprecedented dataset should provide a unique window into processes operating in the shallow magma plumbing system and their relation to subaerial eruption dynamics.

Flaperon Modification Effect on Jet-Flap Interaction Noise Reduction for Chevron Nozzles

NASA Technical Reports Server (NTRS)

Thomas, Russell H.; Mengle, Vinod G.; Stoker, Robert W.; Brusniak, Leon; Elkoby, Ronen

2007-01-01

Jet-flap interaction (JFI) noise can become an important component of far field noise when a flap is immersed in the engine propulsive stream or is in its entrained region, as in approach conditions for under-the-wing engine configurations. We experimentally study the effect of modifying the flaperon, which is a high speed aileron between the inboard and outboard flaps, at both approach and take-off conditions using scaled models in a free jet. The flaperon modifications were of two types: sawtooth trailing edge and mini vortex generators (vg s). Parametric variations of these two concepts were tested with a round coaxial nozzle and an advanced chevron nozzle, with azimuthally varying fan chevrons, using both far field microphone arrays and phased microphone arrays for source diagnostics purposes. In general, the phased array results corroborated the far field results in the upstream quadrant pointing to JFI near the flaperon trailing edge as the origin of the far field noise changes. Specific sawtooth trailing edges in conjunction with the round nozzle gave marginal reduction in JFI noise at approach, and parallel co-rotating mini-vg s were somewhat more beneficial over a wider range of angles, but both concepts were noisier at take-off conditions. These two concepts had generally an adverse JFI effect when used in conjunction with the advanced chevron nozzle at both approach and take-off conditions.

Modeling and analysis of secondary sources coupling for active sound field reduction in confined spaces

NASA Astrophysics Data System (ADS)

Montazeri, Allahyar; Taylor, C. James

2017-10-01

This article addresses the coupling of acoustic secondary sources in a confined space in a sound field reduction framework. By considering the coupling of sources in a rectangular enclosure, the set of coupled equations governing its acoustical behavior are solved. The model obtained in this way is used to analyze the behavior of multi-input multi-output (MIMO) active sound field control (ASC) systems, where the coupling of sources cannot be neglected. In particular, the article develops the analytical results to analyze the effect of coupling of an array of secondary sources on the sound pressure levels inside an enclosure, when an array of microphones is used to capture the acoustic characteristics of the enclosure. The results are supported by extensive numerical simulations showing how coupling of loudspeakers through acoustic modes of the enclosure will change the strength and hence the driving voltage signal applied to the secondary loudspeakers. The practical significance of this model is to provide a better insight on the performance of the sound reproduction/reduction systems in confined spaces when an array of loudspeakers and microphones are placed in a fraction of wavelength of the excitation signal to reduce/reproduce the sound field. This is of particular importance because the interaction of different sources affects their radiation impedance depending on the electromechanical properties of the loudspeakers.

Electronic filters, hearing aids and methods

NASA Technical Reports Server (NTRS)

Engebretson, A. Maynard (Inventor)

1995-01-01

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

Electronic filters, hearing aids and methods

NASA Technical Reports Server (NTRS)

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

1991-01-01

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

Wavespace-Based Coherent Deconvolution

NASA Technical Reports Server (NTRS)

Bahr, Christopher J.; Cattafesta, Louis N., III

2012-01-01

Array deconvolution is commonly used in aeroacoustic analysis to remove the influence of a microphone array's point spread function from a conventional beamforming map. Unfortunately, the majority of deconvolution algorithms assume that the acoustic sources in a measurement are incoherent, which can be problematic for some aeroacoustic phenomena with coherent, spatially-distributed characteristics. While several algorithms have been proposed to handle coherent sources, some are computationally intractable for many problems while others require restrictive assumptions about the source field. Newer generalized inverse techniques hold promise, but are still under investigation for general use. An alternate coherent deconvolution method is proposed based on a wavespace transformation of the array data. Wavespace analysis offers advantages over curved-wave array processing, such as providing an explicit shift-invariance in the convolution of the array sampling function with the acoustic wave field. However, usage of the wavespace transformation assumes the acoustic wave field is accurately approximated as a superposition of plane wave fields, regardless of true wavefront curvature. The wavespace technique leverages Fourier transforms to quickly evaluate a shift-invariant convolution. The method is derived for and applied to ideal incoherent and coherent plane wave fields to demonstrate its ability to determine magnitude and relative phase of multiple coherent sources. Multi-scale processing is explored as a means of accelerating solution convergence. A case with a spherical wave front is evaluated. Finally, a trailing edge noise experiment case is considered. Results show the method successfully deconvolves incoherent, partially-coherent, and coherent plane wave fields to a degree necessary for quantitative evaluation. Curved wave front cases warrant further investigation. A potential extension to nearfield beamforming is proposed.

Experimental study using Nearfield Acoustical Holography of sound transmission fuselage sidewall structures

NASA Technical Reports Server (NTRS)

Maynard, J. D.

1983-01-01

This project involves the development of the Nearfield Acoustic Holography (NAH) technique (in particular its extension from single frequency to wideband noise measurement) and its application in a detailed study of the noise radiation characteristics of several samples of aircraft sidewall panels. With the extensive amount of information provided by the NAH technique, the properties of the sound field radiated by the panels may be correlated with their structure, mounting, and excitation (single frequency or wideband, spatially correlated or uncorrelated, structure-borne). The work accomplished at the beginning of this grant period included: (1) Calibration of the 256 microphone array and test of its accuracy. (2) extension of the facility to permit measurements on wideband noise sources. The extensions incuded the addition of high-speed data acquisition hardware and an array processor, and the development of new software. (3) Installation of motion picture graphics for correlating panel motion with structure, mounting, radiation, etc. (4) Development of new holographic data processing techniques.

Multi-criteria anomaly detection in urban noise sensor networks.

PubMed

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

2014-01-01

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

Signal Processing and Interpretation Using Multilevel Signal Abstractions.

DTIC Science & Technology

1986-06-01

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

A static acoustic signature system for the analysis of dynamic flight information

NASA Technical Reports Server (NTRS)

Ramer, D. J.

1978-01-01

The Army family of helicopters was analyzed to measure the polar octave band acoustic signature in various modes of flight. A static array of calibrated microphones was used to simultaneously acquire the signature and differential times required to mathematically position the aircraft in space. The signature was then reconstructed, mathematically normalized to a fixed radius around the aircraft.

Synchronized and noise-robust audio recordings during realtime magnetic resonance imaging scans.

PubMed

Bresch, Erik; Nielsen, Jon; Nayak, Krishna; Narayanan, Shrikanth

2006-10-01

This letter describes a data acquisition setup for recording, and processing, running speech from a person in a magnetic resonance imaging (MRI) scanner. The main focus is on ensuring synchronicity between image and audio acquisition, and in obtaining good signal to noise ratio to facilitate further speech analysis and modeling. A field-programmable gate array based hardware design for synchronizing the scanner image acquisition to other external data such as audio is described. The audio setup itself features two fiber optical microphones and a noise-canceling filter. Two noise cancellation methods are described including a novel approach using a pulse sequence specific model of the gradient noise of the MRI scanner. The setup is useful for scientific speech production studies. Sample results of speech and singing data acquired and processed using the proposed method are given.

Synchronized and noise-robust audio recordings during realtime magnetic resonance imaging scans (L)

PubMed Central

Bresch, Erik; Nielsen, Jon; Nayak, Krishna; Narayanan, Shrikanth

2007-01-01

This letter describes a data acquisition setup for recording, and processing, running speech from a person in a magnetic resonance imaging (MRI) scanner. The main focus is on ensuring synchronicity between image and audio acquisition, and in obtaining good signal to noise ratio to facilitate further speech analysis and modeling. A field-programmable gate array based hardware design for synchronizing the scanner image acquisition to other external data such as audio is described. The audio setup itself features two fiber optical microphones and a noise-canceling filter. Two noise cancellation methods are described including a novel approach using a pulse sequence specific model of the gradient noise of the MRI scanner. The setup is useful for scientific speech production studies. Sample results of speech and singing data acquired and processed using the proposed method are given. PMID:17069275

An evaluation of talker localization based on direction of arrival estimation and statistical sound source identification

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.

Acoustic imaging of a duct spinning mode by the use of an in-duct circular microphone array.

PubMed

Wei, Qingkai; Huang, Xun; Peers, Edward

2013-06-01

An imaging method of acoustic spinning modes propagating within a circular duct simply with surface pressure information is introduced in this paper. The proposed method is developed in a theoretical way and is demonstrated by a numerical simulation case. Nowadays, the measurements within a duct have to be conducted using in-duct microphone array, which is unable to provide information of complete acoustic solutions across the test section. The proposed method can estimate immeasurable information by forming a so-called observer. The fundamental idea behind the testing method was originally developed in control theory for ordinary differential equations. Spinning mode propagation, however, is formulated in partial differential equations. A finite difference technique is used to reduce the associated partial differential equations to a classical form in control. The observer method can thereafter be applied straightforwardly. The algorithm is recursive and, thus, could be operated in real-time. A numerical simulation for a straight circular duct is conducted. The acoustic solutions on the test section can be reconstructed with good agreement to analytical solutions. The results suggest the potential and applications of the proposed method.

Acoustic Measurements of a Large Civil Transport Main Landing Gear Model

NASA Technical Reports Server (NTRS)

Ravetta, Patricio A.; Khorrami, Mehdi R.; Burdisso, Ricardo A.; Wisda, David M.

2016-01-01

Microphone phased array acoustic measurements of a 26 percent-scale, Boeing 777-200 main landing gear model with and without noise reduction fairings installed were obtained in the anechoic configuration of the Virginia Tech Stability Tunnel. Data were acquired at Mach numbers of 0.12, 0.15, and 0.17 with the latter speed used as the nominal test condition. The fully and partially dressed gear with the truck angle set at 13 degrees toe-up landing configuration were the two most extensively tested configurations, serving as the baselines for comparison purposes. Acoustic measurements were also acquired for the same two baseline configurations with the truck angle set at 0 degrees. In addition, a previously tested noise reducing, toboggan-shaped fairing was re-evaluated extensively to address some of the lingering questions regarding the extent of acoustic benefit achievable with this device. The integrated spectra generated from the acoustic source maps reconfirm, in general terms, the previously reported noise reduction performance of the toboggan fairing as installed on an isolated gear. With the recent improvements to the Virginia Tech tunnel acoustic quality and microphone array capabilities, the present measurements provide an additional, higher quality database to the acoustic information available for this gear model.

Micropower Mixed-signal VLSI Independent Component Analysis for Gradient Flow Acoustic Source Separation.

PubMed

Stanaćević, Milutin; Li, Shuo; Cauwenberghs, Gert

2016-07-01

A parallel micro-power mixed-signal VLSI implementation of independent component analysis (ICA) with reconfigurable outer-product learning rules is presented. With the gradient sensing of the acoustic field over a miniature microphone array as a pre-processing method, the proposed ICA implementation can separate and localize up to 3 sources in mild reverberant environment. The ICA processor is implemented in 0.5 µm CMOS technology and occupies 3 mm × 3 mm area. At 16 kHz sampling rate, ASIC consumes 195 µW power from a 3 V supply. The outer-product implementation of natural gradient and Herault-Jutten ICA update rules demonstrates comparable performance to benchmark FastICA algorithm in ideal conditions and more robust performance in noisy and reverberant environment. Experiments demonstrate perceptually clear separation and precise localization over wide range of separation angles of two speech sources presented through speakers positioned at 1.5 m from the array on a conference room table. The presented ASIC leads to a extreme small form factor and low power consumption microsystem for source separation and localization required in applications like intelligent hearing aids and wireless distributed acoustic sensor arrays.

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

PubMed

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

2018-03-01

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

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

PubMed

Picou, Erin M; Ricketts, Todd A

2018-06-01

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

Optimization of actuator arrays for aircraft interior noise control

NASA Technical Reports Server (NTRS)

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

1993-01-01

A numerical procedure for grouping actuators in order to reduce the number of degrees of freedom in an active noise control system is evaluated using experimental data. Piezoceramic actuators for reducing aircraft interior noise are arranged into groups using a nonlinear optimization routine and clustering algorithm. An actuator group is created when two or more actuators are driven with the same control input. This procedure is suitable for active control applications where actuators are already mounted on a structure. The feasibility of this technique is demonstrated using measured data from the aft cabin of a Douglas DC-9 fuselage. The measured data include transfer functions between 34 piezoceramic actuators and 29 interior microphones and microphone responses due to the primary noise produced by external speakers. Control inputs for the grouped actuators were calculated so that a cost function, defined as a quadratic pressure term and a penalty term, was a minimum. The measured transfer functions and microphone responses are checked by comparing calculated noise reductions with measured noise reductions for four frequencies. The grouping procedure is then used to determine actuator groups that improve overall interior noise reductions by 5.3 to 15 dB, compared to the baseline experimental configuration.

The effect of frequency-dependent microphone directionality on horizontal localization performance in hearing-aid users.

PubMed

Keidser, Gitte; O'Brien, Anna; Hain, Jens-Uwe; McLelland, Margot; Yeend, Ingrid

2009-11-01

Frequency-dependent microphone directionality alters the spectral shape of sound as a function of arrival azimuth. The influence of this on horizontal-plane localization performance was investigated. Using a 360 degrees loudspeaker array and five stimuli with different spectral characteristics, localization performance was measured on 21 hearing-impaired listeners when wearing no hearing aids and aided with no directionality, partial (from 1 and 2 kHz) directionality, and full directionality. The test schemes were also evaluated in everyday life. Without hearing aids, localization accuracy was significantly poorer than normative data. Due to inaudibility of high-frequency energy, front/back reversals were prominent. Front/back reversals remained prominent when aided with omnidirectional microphones. For stimuli with low-frequency emphasis, directionality had no further effect on localization. For stimuli with sufficient mid- and high-frequency information, full directionality had a small positive effect on front/back localization but a negative effect on left/right localization. Partial directionality further improved front/back localization and had no significant effect on left/right localization. The field test revealed no significant effects. The alternative spectral cues provided by frequency-dependent directionality improve front/back localization in hearing-aid users.

Improved Phased Array Imaging of a Model Jet

NASA Technical Reports Server (NTRS)

Dougherty, Robert P.; Podboy, Gary G.

2010-01-01

An advanced phased array system, OptiNav Array 48, and a new deconvolution algorithm, TIDY, have been used to make octave band images of supersonic and subsonic jet noise produced by the NASA Glenn Small Hot Jet Acoustic Rig (SHJAR). The results are much more detailed than previous jet noise images. Shock cell structures and the production of screech in an underexpanded supersonic jet are observed directly. Some trends are similar to observations using spherical and elliptic mirrors that partially informed the two-source model of jet noise, but the radial distribution of high frequency noise near the nozzle appears to differ from expectations of this model. The beamforming approach has been validated by agreement between the integrated image results and the conventional microphone data.

Airborne ultrasonic phased arrays using ferroelectrets: a new fabrication approach.

PubMed

Ealo, Joao L; Camacho, Jorge J; Fritsch, Carlos

2009-04-01

In this work, a novel procedure that considerably simplifies the fabrication process of ferroelectret-based multielement array transducers is proposed and evaluated. Also, the potential of ferroelectrets being used as active material for air-coupled ultrasonic transducer design is demonstrated. The new construction method of multi-element transducers introduces 2 distinctive improvements. First, active ferroelectret material is not discretized into elements, and second, the need of structuring upper and/or lower electrodes in advance of the permanent polarization of the film is removed. The aperture discretization and the mechanical connection are achieved in one step using a through-thickness conductive tape. To validate the procedure, 2 linear array prototypes of 32 elements, with a pitch of 3.43 mm and a wide usable frequency range from 30 to 300 kHz, were built and evaluated using a commercial phased-array system. A low crosstalk among elements, below -30 dB, was measured by interferometry. Likewise, a homogeneous response of the array elements, with a maximum deviation of +/-1.8 dB, was obtained. Acoustic beam steering measurements were accomplished at different deflection angles using a calibrated microphone. The ultrasonic beam parameters, namely, lateral resolution, side lobe level, grating lobes, and focus depth, were congruent with theory. Acoustic images of a single reflector were obtained using one of the array elements as the receiver. Resulting images are also in accordance with numerical simulation, demonstrating the feasibility of using these arrays in pulse-echo mode. The proposed procedure simplifies the manufacturing of multidimensional arrays with arbitrary shape elements and not uniformly distributed. Furthermore, this concept can be extended to nonflat arrays as long as the transducer substrate conforms to a developable surface.

Identification and tracking of particular speaker in noisy environment

NASA Astrophysics Data System (ADS)

Sawada, Hideyuki; Ohkado, Minoru

2004-10-01

Human is able to exchange information smoothly using voice under different situations such as noisy environment in a crowd and with the existence of plural speakers. We are able to detect the position of a source sound in 3D space, extract a particular sound from mixed sounds, and recognize who is talking. By realizing this mechanism with a computer, new applications will be presented for recording a sound with high quality by reducing noise, presenting a clarified sound, and realizing a microphone-free speech recognition by extracting particular sound. The paper will introduce a realtime detection and identification of particular speaker in noisy environment using a microphone array based on the location of a speaker and the individual voice characteristics. The study will be applied to develop an adaptive auditory system of a mobile robot which collaborates with a factory worker.

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

PubMed

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

2013-06-01

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

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.

Mach stem formation in outdoor measurements of acoustic shocks.

PubMed

Leete, Kevin M; Gee, Kent L; Neilsen, Tracianne B; Truscott, Tadd T

2015-12-01

Mach stem formation during outdoor acoustic shock propagation is investigated using spherical oxyacetylene balloons exploded above pavement. The location of the transition point from regular to irregular reflection and the path of the triple point are experimentally resolved using microphone arrays and a high-speed camera. The transition point falls between recent analytical work for weak irregular reflections and an empirical relationship derived from large explosions.

Use of Pattern Classification Algorithms to Interpret Passive and Active Data Streams from a Walking-Speed Robotic Sensor Platform

NASA Astrophysics Data System (ADS)

Dieckman, Eric Allen

In order to perform useful tasks for us, robots must have the ability to notice, recognize, and respond to objects and events in their environment. This requires the acquisition and synthesis of information from a variety of sensors. Here we investigate the performance of a number of sensor modalities in an unstructured outdoor environment, including the Microsoft Kinect, thermal infrared camera, and coffee can radar. Special attention is given to acoustic echolocation measurements of approaching vehicles, where an acoustic parametric array propagates an audible signal to the oncoming target and the Kinect microphone array records the reflected backscattered signal. Although useful information about the target is hidden inside the noisy time domain measurements, the Dynamic Wavelet Fingerprint process (DWFP) is used to create a time-frequency representation of the data. A small-dimensional feature vector is created for each measurement using an intelligent feature selection process for use in statistical pattern classification routines. Using our experimentally measured data from real vehicles at 50 m, this process is able to correctly classify vehicles into one of five classes with 94% accuracy. Fully three-dimensional simulations allow us to study the nonlinear beam propagation and interaction with real-world targets to improve classification results.

Simulation Study of the Localization of a Near-Surface Crack Using an Air-Coupled Ultrasonic Sensor Array

PubMed Central

Delrue, Steven; Aleshin, Vladislav; Sørensen, Mikael; De Lathauwer, Lieven

2017-01-01

The importance of Non-Destructive Testing (NDT) to check the integrity of materials in different fields of industry has increased significantly in recent years. Actually, industry demands NDT methods that allow fast (preferably non-contact) detection and localization of early-stage defects with easy-to-interpret results, so that even a non-expert field worker can carry out the testing. The main challenge is to combine as many of these requirements into one single technique. The concept of acoustic cameras, developed for low frequency NDT, meets most of the above-mentioned requirements. These cameras make use of an array of microphones to visualize noise sources by estimating the Direction Of Arrival (DOA) of the impinging sound waves. Until now, however, because of limitations in the frequency range and the lack of integrated nonlinear post-processing, acoustic camera systems have never been used for the localization of incipient damage. The goal of the current paper is to numerically investigate the capabilities of locating incipient damage by measuring the nonlinear airborne emission of the defect using a non-contact ultrasonic sensor array. We will consider a simple case of a sample with a single near-surface crack and prove that after efficient excitation of the defect sample, the nonlinear defect responses can be detected by a uniform linear sensor array. These responses are then used to determine the location of the defect by means of three different DOA algorithms. The results obtained in this study can be considered as a first step towards the development of a nonlinear ultrasonic camera system, comprising the ultrasonic sensor array as the hardware and nonlinear post-processing and source localization software. PMID:28441738

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.

Single and Multiple Microphone Noise Reduction Strategies in Cochlear Implants

PubMed Central

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

2012-01-01

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

Active noise canceling system for mechanically cooled germanium radiation detectors

DOEpatents

Nelson, Karl Einar; Burks, Morgan T

2014-04-22

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

KSC-00padig100

NASA Image and Video Library

2000-11-27

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Mission Specialist Carlos Noriega. Behind him stand Commander Brent Jett, Pilot Michael Bloomfield and Mission Specialists Joseph Tanner and Marc Garneau, who is with the Canadian Space Agency. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

KSC-00pp1759

NASA Image and Video Library

2000-11-27

After arriving at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Mission Specialist Marc Garneau, who is with the Canadian Space Agency. Behind him can be seen Mission Specialists Joseph Tanner (left) and Carlos Noriega. Mission STS-97 is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

KSC-00padig102

NASA Image and Video Library

2000-11-27

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Mission Specialist Marc Garneau, who is with the Canadian Space Agency. Behind him stand Commander Brent Jett, Pilot Michael Bloomfield and Mission Specialists Joseph Tanner and Carlos Noriega. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

KSC-00padig101

NASA Image and Video Library

2000-11-27

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Mission Specialist Joseph Tanner. Behind him stand Commander Brent Jett, Pilot Michael Bloomfield and Mission Specialists Marc Garneau, who is with the Canadian Space Agency, and Carlos Noriega. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

KSC-00padig099

NASA Image and Video Library

2000-11-27

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Pilot Michael Bloomfield. Behind him stand Commander Brent Jett and Mission Specialists Joseph Tanner, Carolos Noriega and Marc Garneau, who is with the Canadian Space Agency. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

Partial differential equation-based localization of a monopole source from a circular array.

PubMed

Ando, Shigeru; Nara, Takaaki; Levy, Tsukassa

2013-10-01

Wave source localization from a sensor array has long been the most active research topics in both theory and application. In this paper, an explicit and time-domain inversion method for the direction and distance of a monopole source from a circular array is proposed. The approach is based on a mathematical technique, the weighted integral method, for signal/source parameter estimation. It begins with an exact form of the source-constraint partial differential equation that describes the unilateral propagation of wide-band waves from a single source, and leads to exact algebraic equations that include circular Fourier coefficients (phase mode measurements) as their coefficients. From them, nearly closed-form, single-shot and multishot algorithms are obtained that is suitable for use with band-pass/differential filter banks. Numerical evaluation and several experimental results obtained using a 16-element circular microphone array are presented to verify the validity of the proposed method.

The effect of operations on the ground noise footprints associated with a large multibladed, nonbanging helicopter

NASA Technical Reports Server (NTRS)

Hilton, D. A.; Henderson, H. R.; Maglieri, D. J.; Bigler, W. B., II

1978-01-01

In order to expand the data base of helicopter external noise characteristics, a flyover noise measurement program was conducted utilizing the NASA Civil Helicopter Research Aircraft. The remotely operated multiple array acoustics range (ROMAAR) and a 2560-m linear microphone array were utilized for the purpose of documenting the noise characteristics of the test helicopter during flyby and landing operations. By utilizing both ROMAAR concept and the linear array, the data necessary to plot the ground noise footprints and noise radiation patterns were obtained. Examples of the measured noise signature of the test helicopter, the ground noise footprint or contours, and the directivity patterns measured during level flyby and landing operations of a large, multibladed, nonbanging helicopter, the CH-53, are presented.

Sensing of Particular Speakers for the Construction of Voice Interface Utilized in Noisy Environment

NASA Astrophysics Data System (ADS)

Sawada, Hideyuki; Ohkado, Minoru

Human is able to exchange information smoothly using voice under different situations such as noisy environment in a crowd and with the existence of plural speakers. We are able to detect the position of a source sound in 3D space, extract a particular sound from mixed sounds, and recognize who is talking. By realizing this mechanism with a computer, new applications will be presented for recording a sound with high quality by reducing noise, presenting a clarified sound, and realizing a microphone-free speech recognition by extracting particular sound. The paper will introduce a realtime detection and identification of particular speaker in noisy environment using a microphone array based on the location of a speaker and the individual voice characteristics. The study will be applied to develop an adaptive auditory system of a mobile robot which collaborates with a factory worker.

Apparatus and Methods for Photoacoustic Measurement of Light Absorption of Particulate and Gaseous Species

NASA Technical Reports Server (NTRS)

Brown, William (Inventor); Yu, Zhenhong (Inventor); Kebabian, Paul L. (Inventor); Assif, James (Inventor)

2017-01-01

In one embodiment, a photoacoustic effect measurement instrument for measuring a species (e.g., a species of PM) in a gas employs a pair of differential acoustic cells including a sample cell that receives sample gas including the species, and a reference cell that receives a filtered version of the sample gas from which the species has been substantially removed. An excitation light source provides an amplitude modulated beam to each of the acoustic cells. An array of multiple microphones is mounted to each of the differential acoustic cells, and measures an acoustic wave generated in the respective acoustic cell by absorption of light by sample gas therein to produce a respective signal. The microphones are isolated from sample gas internal to the acoustic cell by a film. A preamplifier determines a differential signal and a controller calculates concentration of the species based on the differential signal.

Antarctic atmospheric infrasound. Final technical report, 1 July 1981-30 September 1984

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

Wilson, C.R.; McKibben, B.N.

1986-11-01

In order to monitor atmospheric infrasonic waves in the passband from 0.1 to 0.01 Hz a digital infrasonic detection system was installed in Antarctica on the Ross Ice shelf near McMurdo Station on McMurdo Sound. An array of seven infrasonic microphones subtending an area of about 35 sg km was operated in Windless Bight. The analog microphone data were telemetered to McMurdo station where the infrasonic date were digitized and subjected to on-line real-time analysis to detect traveling infrasonic waves with periods from 10 to 100 seconds. During the period of operation of the Antartic infrasonic observatory, hundreds of infrasonicmore » signals were detected in association with many natural sources such as the aurora australis, marine storm sea-air interactions, volcanic eruptions, mountain generated lee-wave effects, large meteors and auroral electrojet supersonic motions.« less

Method of fan sound mode structure determination

NASA Technical Reports Server (NTRS)

Pickett, G. F.; Sofrin, T. G.; Wells, R. W.

1977-01-01

A method for the determination of fan sound mode structure in the Inlet of turbofan engines using in-duct acoustic pressure measurements is presented. The method is based on the simultaneous solution of a set of equations whose unknowns are modal amplitude and phase. A computer program for the solution of the equation set was developed. An additional computer program was developed which calculates microphone locations the use of which results in an equation set that does not give rise to numerical instabilities. In addition to the development of a method for determination of coherent modal structure, experimental and analytical approaches are developed for the determination of the amplitude frequency spectrum of randomly generated sound models for use in narrow annulus ducts. Two approaches are defined: one based on the use of cross-spectral techniques and the other based on the use of an array of microphones.

Sonic boom signature data from cruciform microphone array experiments during the 1966-1967 EAFB national sonic boom evaluation program

NASA Technical Reports Server (NTRS)

Hubbard, H. H.; Maglieri, D. J.

1990-01-01

Tables are provided of measured sonic boom signature data derived from supersonic flyover tests of the XB-70, B-58 and F-104 aircraft for ranges of altitude and Mach number. These tables represent a convenient hard copy version of available electronic files and complement preliminary information included in a reference National Sonic Boom Evaluation Office document.

Sub-Surface Windscreen for the Measurement of Outdoor Infrasound

NASA Technical Reports Server (NTRS)

Shams, Qamar A.; Burkett, Cecil G., Jr.; Comeaux, Toby; Zuckerwar, Allan J.; Weistroffer, George R.

2008-01-01

A windscreen has been developed that features two advantages favorable for the measurement of outdoor infrasound. First, the sub-surface location, with the top of the windscreen flush with the ground surface, minimizes the mean velocity of the impinging wind. Secondly, the windscreen material (closed cell polyurethane foam) has a sufficiently low acoustic impedance (222 times that of air) and wall thickness (0.0127 m) to provide a transmission coefficient of nearly unity over the infrasonic frequency range (0-20 Hz). The windscreen, a tightly-sealed box having internal dimensions of 0.3048 x 0.3048 x 0.3556 m, contains a microphone, preamplifier, and a cable feed thru to an external power supply. Provisions are made for rain drainage and seismic isolation. A three-element array, configured as an equilateral triangle with 30.48 m spacing and operating continuously in the field, periodically receives highly coherent signals attributed to emissions from atmospheric turbulence. The time delays between infrasonic signals received at the microphones permit determination of the bearing and elevation of the sources, which correlate well with locations of pilot reports (PIREPS) within a 320 km radius about the array. The test results are interpreted to yield spectral information on infrasonic emissions from clear air turbulence.

The effect of hearing aid technologies on listening in an automobile

PubMed Central

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

2014-01-01

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

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.

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…

Localization of short-range acoustic and seismic wideband sources: Algorithms and experiments

NASA Astrophysics Data System (ADS)

Stafsudd, J. Z.; Asgari, S.; Hudson, R.; Yao, K.; Taciroglu, E.

2008-04-01

We consider the determination of the location (source localization) of a disturbance source which emits acoustic and/or seismic signals. We devise an enhanced approximate maximum-likelihood (AML) algorithm to process data collected at acoustic sensors (microphones) belonging to an array of, non-collocated but otherwise identical, sensors. The approximate maximum-likelihood algorithm exploits the time-delay-of-arrival of acoustic signals at different sensors, and yields the source location. For processing the seismic signals, we investigate two distinct algorithms, both of which process data collected at a single measurement station comprising a triaxial accelerometer, to determine direction-of-arrival. The direction-of-arrivals determined at each sensor station are then combined using a weighted least-squares approach for source localization. The first of the direction-of-arrival estimation algorithms is based on the spectral decomposition of the covariance matrix, while the second is based on surface wave analysis. Both of the seismic source localization algorithms have their roots in seismology; and covariance matrix analysis had been successfully employed in applications where the source and the sensors (array) are typically separated by planetary distances (i.e., hundreds to thousands of kilometers). Here, we focus on very-short distances (e.g., less than one hundred meters) instead, with an outlook to applications in multi-modal surveillance, including target detection, tracking, and zone intrusion. We demonstrate the utility of the aforementioned algorithms through a series of open-field tests wherein we successfully localize wideband acoustic and/or seismic sources. We also investigate a basic strategy for fusion of results yielded by acoustic and seismic arrays.

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

PubMed

Jenkins, Herman A; Uhler, Kristin

2012-01-01

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

Experimental investigation of an inversion technique for the determination of broadband duct mode amplitudes by the use of near-field sensor arrays.

PubMed

Castres, Fabrice O; Joseph, Phillip F

2007-08-01

This paper is an experimental investigation of an inverse technique for deducing the amplitudes of the modes radiated from a turbofan engine, including schemes for stablizing the solution. The detection of broadband modes generated by a laboratory-scaled fan inlet is performed using a near-field array of microphones arranged in a geodesic geometry. This array geometry is shown to allow a robust and accurate modal inversion. The sound power radiated from the fan inlet and the coherence function between different modal amplitudes are also presented. The knowledge of such modal content is useful in helping to characterize the source mechanisms of fan broadband noise generation, for determining the most appropriate mode distribution model for duct liner predictions, and for making sound power measurements of the radiated sound field.

Cryogenic Design of the Setup for MARE-1 in Milan

NASA Astrophysics Data System (ADS)

Schaeffer, D.; Arnaboldi, C.; Ceruti, G.; Ferri, E.; Kilbourne, C.; Kraft-Bermuth, S.; Margesin, B.; McCammon, D.; Monfardini, A.; Nucciotti, A.; Pessina, G.; Previtali, E.; Sisti, M.

2008-05-01

A large worldwide collaboration is growing around the project of Micro-calorimeter Arrays for a Rhenium Experiment (MARE) for a direct calorimetric measurement of the neutrino mass. To validate the use of cryogenic detectors by checking the presence of unexpected systematic errors, two first experiments are planned using the available techniques composed of arrays of 300 detectors to measure 1010 events in a reasonable time of 3 years (step MARE-1) to reach a sensitivity on the neutrino mass of ˜2 eV/c2. Our experiment in Milan is based on compensated doped silicon implanted thermistor arrays made in NASA/GSFC and on AgReO4 crystals. We present here the design of the cryogenic system that integrates all the requirements for such experiment (electronics for high impedances, low parasitic capacitances, low micro-phonic noise).

KSC00pp1757

NASA Image and Video Library

2000-11-27

After arriving at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone, Commander Brent Jett praises the efforts of the KSC workers to get ready for the launch. Behind Jett are Pilot Michael Bloomfield and Mission Specialists Joseph Tanner, Carlos Noriega and Marc Garneau, who is with the Canadian Space Agency. Mission STS-97 is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

KSC-00padig098

NASA Image and Video Library

2000-11-27

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone, Commander Brent Jett praises the efforts of the KSC workers to get ready for the launch. Behind Jett are Pilot Michael Bloomfield and Mission Specialists Joseph Tanner, Carolos Noriega and Marc Garneau, who is with the Canadian Space Agency. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

KSC-00pp1757

NASA Image and Video Library

2000-11-27

After arriving at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone, Commander Brent Jett praises the efforts of the KSC workers to get ready for the launch. Behind Jett are Pilot Michael Bloomfield and Mission Specialists Joseph Tanner, Carlos Noriega and Marc Garneau, who is with the Canadian Space Agency. Mission STS-97 is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST

STS-97 Mission Specialist Noriega talks to media after arrival for launch

NASA Technical Reports Server (NTRS)

2000-01-01

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Mission Specialist Carlos Noriega. Behind him stand Commander Brent Jett, Pilot Michael Bloomfield and Mission Specialists Joseph Tanner and Marc Garneau, who is with the Canadian Space Agency. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST.

STS-97 Mission Specialist Tanner talks to media after arrival for launch

NASA Technical Reports Server (NTRS)

2000-01-01

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Mission Specialist Joseph Tanner. Behind him stand Commander Brent Jett, Pilot Michael Bloomfield and Mission Specialists Marc Garneau, who is with the Canadian Space Agency, and Carlos Noriega. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST.

STS-97 Mission Specialist Garneau talks to media after arrival for launch

NASA Technical Reports Server (NTRS)

2000-01-01

After their arrival at the Shuttle Landing Facility, the STS-97 crew gather to address the media. At the microphone is Mission Specialist Marc Garneau, who is with the Canadian Space Agency. Behind him stand Commander Brent Jett, Pilot Michael Bloomfield and Mission Specialists Joseph Tanner and Carlos Noriega. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:06 p.m. EST.

a Study of Ultrasonic Wave Propagation Through Parallel Arrays of Immersed Tubes

NASA Astrophysics Data System (ADS)

Cocker, R. P.; Challis, R. E.

1996-06-01

Tubular array structures are a very common component in industrial heat exchanging plant and the non-destructive testing of these arrays is essential. Acoustic methods using microphones or ultrasound are attractive but require a thorough understanding of the acoustic properties of tube arrays. This paper details the development and testing of a small-scale physical model of a tube array to verify the predictions of a theoretical model for acoustic propagation through tube arrays developed by Heckl, Mulholland, and Huang [1-5] as a basis for the consideration of small-scale physical models in the development of non-destructive testing procedures for tube arrays. Their model predicts transmission spectra for plane waves incident on an array of tubes arranged in straight rows. Relative transmission is frequency dependent with bands of high and low attenuation caused by resonances within individual tubes and between tubes in the array. As the number of rows in the array increases the relative transmission spectrum becomes more complex, with increasingly well-defined bands of high and low attenuation. Diffraction of acoustic waves with wavelengths less than the tube spacing is predicted and appears as step reductions in the transmission spectrum at frequencies corresponding to integer multiples of the tube spacing. Experiments with the physical model confirm the principle features of the theoretical treatment.

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

NASA Technical Reports Server (NTRS)

Horne, William C.; Burnside, Nathan J.

2013-01-01

The AMELIA Cruise-Efficient Short Take-off and Landing (CESTOL) configuration concept was developed to meet future requirements of reduced field length, noise, and fuel burn by researchers at Cal Poly, San Luis Obispo and Georgia Tech Research Institute under sponsorship by the NASA Fundamental Aeronautics Program (FAP), Subsonic Fixed Wing Project. The novel configuration includes leading- and trailing-edge circulation control wing (CCW), over-wing podded turbine propulsion simulation (TPS). Extensive aerodynamic measurements of forces, surfaces pressures, and wing surface skin friction measurements were recently measured over a wide range of test conditions in the Arnold Engineering Development Center(AEDC) National Full-Scale Aerodynamics Complex (NFAC) 40- by 80-Ft Wind Tunnel. Acoustic measurements of the model were also acquired for each configuration with 7 fixed microphones on a line under the left wing, and with a 48-element, 40-inch diameter phased microphone array under the right wing. This presentation will discuss acoustic characteristics of the CCW system for a variety of tunnel speeds (0 to 120 kts), model configurations (leading edge(LE) and/or trailing-edge(TE) slot blowing, and orientations (incidence and yaw) based on acoustic measurements acquired concurrently with the aerodynamic measurements. The flow coefficient, Cmu= mVSLOT/qSW varied from 0 to 0.88 at 40 kts, and from 0 to 0.15 at 120 kts. Here m is the slot mass flow rate, VSLOT is the slot exit velocity, q is dynamic pressure, and SW is wing surface area. Directivities at selected 1/3 octave bands will be compared with comparable measurements of a 2-D wing at GTRI, as will as microphone array near-field measurements of the right wing at maximum flow rate. The presentation will include discussion of acoustic sensor calibrations as well as characterization of the wind tunnel background noise environment.

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

PubMed Central

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

2011-01-01

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

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.

Factors influencing individual variation in perceptual directional microphone benefit.

PubMed

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

2013-01-01

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

Infrasonic monitoring of snow avalanches in the Alps

NASA Astrophysics Data System (ADS)

Marchetti, E.; Ulivieri, G.; Ripepe, M.; Chiambretti, I.; Segor, V.

2012-04-01

Risk assessment of snow avalanches is mostly related to weather conditions and snow cover. However a robust risk validation requires to identify all avalanches occurring, in order to compare predictions to real effects. For this purpose on December 2010 we installed a permanent 4-element, small aperture (100 m), infrasound array in the Alps, after a pilot experiment carried out in Gressonay during the 2009-2010 winter season. The array has been deployed in the Ayas Valley, at an elevation of 2000 m a.s.l., where natural avalanches are expected and controlled events are regularly performed. The array consists into 4 Optimic 2180 infrasonic microphones, with a sensitivity of 10-3 Pa in the 0.5-50 Hz frequency band and a 4 channel Guralp CMG-DM24 A/D converter, sampling at 100 Hz. Timing is achieved with a GPS receiver. Data are transmitted to the Department of Earth Sciences of the University of Firenze, where data is recorded and processed in real-time. A multi-channel semblance is carried out on the continuous data set as a function of slowness, back-azimuth and frequency of recorded infrasound in order to detect all avalanches occurring from the back-ground signal, strongly affected by microbarom and mountain induced gravity waves. This permanent installation in Italy will allow to verify the efficiency of the system in short-to-medium range (2-8 km) avalanche detection, and might represent an important validation to model avalanches activity during this winter season. Moreover, the real-time processing of infrasonic array data, might strongly contribute to avalanche risk assessments providing an up-to-description of ongoing events.

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.

Probe Microphone Measurements: 20 Years of Progress

PubMed Central

Mueller, H. Gustav

2001-01-01

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

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.

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

PubMed

Crukley, Jeffery; Scollie, Susan D

2014-03-01

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

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.

Volumetric Acoustic Vector Intensity Probe

NASA Technical Reports Server (NTRS)

Klos, Jacob

2006-01-01

A new measurement tool capable of imaging the acoustic intensity vector throughout a large volume is discussed. This tool consists of an array of fifty microphones that form a spherical surface of radius 0.2m. A simultaneous measurement of the pressure field across all the microphones provides time-domain near-field holograms. Near-field acoustical holography is used to convert the measured pressure into a volumetric vector intensity field as a function of frequency on a grid of points ranging from the center of the spherical surface to a radius of 0.4m. The volumetric intensity is displayed on three-dimensional plots that are used to locate noise sources outside the volume. There is no restriction on the type of noise source that can be studied. The sphere is mobile and can be moved from location to location to hunt for unidentified noise sources. An experiment inside a Boeing 757 aircraft in flight successfully tested the ability of the array to locate low-noise-excited sources on the fuselage. Reference transducers located on suspected noise source locations can also be used to increase the ability of this device to separate and identify multiple noise sources at a given frequency by using the theory of partial field decomposition. The frequency range of operation is 0 to 1400Hz. This device is ideal for the study of noise sources in commercial and military transportation vehicles in air, on land and underwater.

Aircraft Wake Vortex Measurements at Denver International Airport

NASA Technical Reports Server (NTRS)

Dougherty, Robert P.; Wang, Frank Y.; Booth, Earl R.; Watts, Michael E.; Fenichel, Neil; D'Errico, Robert E.

2004-01-01

Airport capacity is constrained, in part, by spacing requirements associated with the wake vortex hazard. NASA's Wake Vortex Avoidance Project has a goal to establish the feasibility of reducing this spacing while maintaining safety. Passive acoustic phased array sensors, if shown to have operational potential, may aid in this effort by detecting and tracking the vortices. During August/September 2003, NASA and the USDOT sponsored a wake acoustics test at the Denver International Airport. The central instrument of the test was a large microphone phased array. This paper describes the test in general terms and gives an overview of the array hardware. It outlines one of the analysis techniques that is being applied to the data and gives sample results. The technique is able to clearly resolve the wake vortices of landing aircraft and measure their separation, height, and sinking rate. These observations permit an indirect estimate of the vortex circulation. The array also provides visualization of the vortex evolution, including the Crow instability.

Wavenumber-domain separation of rail contribution to pass-by noise

NASA Astrophysics Data System (ADS)

Zea, Elias; Manzari, Luca; Squicciarini, Giacomo; Feng, Leping; Thompson, David; Arteaga, Ines Lopez

2017-11-01

In order to counteract the problem of railway noise and its environmental impact, passing trains in Europe must be tested in accordance to a noise legislation that demands the quantification of the noise generated by the vehicle alone. However, for frequencies between about 500 Hz and 1600 Hz, it has been found that a significant part of the measured noise is generated by the rail, which behaves like a distributed source and radiates plane waves as a result of the contact with the train's wheels. Thus the need arises for separating the rail contribution to the pass-by noise in that particular frequency range. To this end, the present paper introduces a wavenumber-domain filtering technique, referred to as wave signature extraction, which requires a line microphone array parallel to the rail, and two accelerometers on the rail in the vertical and lateral direction. The novel contributions of this research are: (i) the introduction and application of wavenumber (or plane-wave) filters to pass-by data measured with a microphone array located in the near-field of the rail, and (ii) the design of such filters without prior information of the structural properties of the rail. The latter is achieved by recording the array pressure, as well as the rail vibrations with the accelerometers, before and after the train pass-by. The performance of the proposed method is investigated with a set of pass-by measurements performed in Germany. The results seem to be promising when compared to reference data from TWINS, and the largest discrepancies occur above 1600 Hz and are attributed to plane waves radiated by the rail that so far have not been accounted for in the design of the filters.

Blind source separation and localization using microphone arrays

NASA Astrophysics Data System (ADS)

Sun, Longji

The blind source separation and localization problem for audio signals is studied using microphone arrays. Pure delay mixtures of source signals typically encountered in outdoor environments are considered. Our proposed approach utilizes the subspace methods, including multiple signal classification (MUSIC) and estimation of signal parameters via rotational invariance techniques (ESPRIT) algorithms, to estimate the directions of arrival (DOAs) of the sources from the collected mixtures. Since audio signals are generally considered broadband, the DOA estimates at frequencies with the large sum of squared amplitude values are combined to obtain the final DOA estimates. Using the estimated DOAs, the corresponding mixing and demixing matrices are computed, and the source signals are recovered using the inverse short time Fourier transform. Subspace methods take advantage of the spatial covariance matrix of the collected mixtures to achieve robustness to noise. While the subspace methods have been studied for localizing radio frequency signals, audio signals have their special properties. For instance, they are nonstationary, naturally broadband and analog. All of these make the separation and localization for the audio signals more challenging. Moreover, our algorithm is essentially equivalent to the beamforming technique, which suppresses the signals in unwanted directions and only recovers the signals in the estimated DOAs. Several crucial issues related to our algorithm and their solutions have been discussed, including source number estimation, spatial aliasing, artifact filtering, different ways of mixture generation, and source coordinate estimation using multiple arrays. Additionally, comprehensive simulations and experiments have been conducted to examine various aspects of the algorithm. Unlike the existing blind source separation and localization methods, which are generally time consuming, our algorithm needs signal mixtures of only a short duration and therefore supports real-time implementation.

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

PubMed Central

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

2011-01-01

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

Noise propagation from a four-engine, propeller-driven airplane

NASA Technical Reports Server (NTRS)

Willshire, William L., Jr.

1987-01-01

A flight experiment was conducted to investigate the propagation of periodic low-frequency noise from a propeller-driven airplane. The test airplane was a large four-engine, propeller-driven airplane flown at altitudes from 15 to 500 m over the end of an 1800-m-long, 22-element microphone array. The acoustic data were reduced by a one-third octave-band analysis. The primary propagation quantities computed were lateral attenuation and ground effects, both of which become significant at shallow elevation angles. Scatter in the measured results largely obscured the physics of the low-frequency noise propagation. Variability of the noise source, up to 9.5 dB over a 2-sec interval, was the major contributor to the data scatter. The microphones mounted at ground level produced more consistent results with less scatter than those mounted 1.2 m above ground. The ground noise levels were found to be greater on the port side than on the starboard side.

Analysis of ground reflection of jet noise obtained with various microphone arrays over an asphalt surface

NASA Technical Reports Server (NTRS)

Miles, J. H.

1975-01-01

Ground reflection effects on the propagation of jet noise over an asphalt surface are discussed for data obtained using a 33.02-cm diameter nozzle with microphones at several heights and distances from the nozzle axis. Ground reflection effects are analyzed using the concept of a reflected signal transfer function which represents the influence of both the reflecting surface and the atmosphere on the propagation of the reflected signal in a mathematical model. The mathematical model used as a basis for the computer program was successful in significantly reducing the ground reflection effects. The range of values of the single complex number used to define the reflected signal transfer function was larger than expected when determined only by the asphalt surface. This may indicate that the atmosphere is affecting the propagation of the reflected signal more than the asphalt surface. The selective placement of the reinforcements and cancellations in the design of an experiment to minimize ground reflection effects is also discussed.

Analysis of ground reflection of jet noise obtained with various microphone arrays over an asphalt surface

NASA Technical Reports Server (NTRS)

Miles, J. H.

1975-01-01

Ground reflection effects on the propagation of jet noise over an asphalt surface are discussed for data obtained using a 33.02 cm (13-in.) diameter nozzle with microphones at several heights and distances from the nozzle axis. Analysis of ground reflection effects is accomplished using the concept of a reflected signal transfer function which represents the influence of both the reflecting surface and the atmosphere on the propagation of the reflected signal in a mathematical model. The mathematical model used as a basis for the computer program was successful in significantly reducing the ground reflection effects. The range of values of the single complex number used to define the reflected signal transfer function was larger than expected when determined only by the asphalt surface. This may indicate that the atmosphere is affecting the propagation of the reflected signal more than the asphalt surface. Also discussed is the selective placement of the reinforcements and cancellations in the design of an experiment to minimize ground reflection effects.

Engine Installation Effects of Four Civil Transport Airplanes: Wallops Flight Facility Study

NASA Technical Reports Server (NTRS)

Fleming, Gregg G.; Senzig, David A.; McCurdy, David A.; Roof, Christopher J.; Rapoza, Amanda S.

2003-01-01

The National Aeronautics and Space Administration (NASA), Langley Research Center (LaRC), the Environmental Measurement and Modeling Division of the United States Department of Transportation s John A. Volpe National Transportation Systems Center (Volpe), and several other organizations (see Appendix A for a complete list of participating organizations and individuals) conducted a noise measurement study at NASA s Wallops Flight Facility (Wallops) near Chincoteague, Virginia during September 2000. This test was intended to determine engine installation effects on four civil transport airplanes: a Boeing 767-400, a McDonnell-Douglas DC9, a Dassault Falcon 2000, and a Beechcraft King Air. Wallops was chosen for this study because of the relatively low ambient noise of the site and the degree of control over airplane operating procedures enabled by operating over a runway closed to other uses during the test period. Measurements were conducted using a twenty microphone U-shaped array oriented perpendicular to the flight path; microphones were mounted such that ground effects were minimized and low elevation angles were observed.

Phased Acoustic Array Measurements of a 5.75 Percent Hybrid Wing Body Aircraft

NASA Technical Reports Server (NTRS)

Burnside, Nathan J.; Horne, William C.; Elmer, Kevin R.; Cheng, Rui; Brusniak, Leon

2016-01-01

Detailed acoustic measurements of the noise from the leading-edge Krueger flap of a 5.75 percent Hybrid Wing Body (HWB) aircraft model were recently acquired with a traversing phased microphone array in the AEDC NFAC (Arnold Engineering Development Complex, National Full Scale Aerodynamics Complex) 40- by 80-Foot Wind Tunnel at NASA Ames Research Center. The spatial resolution of the array was sufficient to distinguish between individual support brackets over the full-scale frequency range of 100 to 2875 Hertz. For conditions representative of landing and take-off configuration, the noise from the brackets dominated other sources near the leading edge. Inclusion of flight-like brackets for select conditions highlights the importance of including the correct number of leading-edge high-lift device brackets with sufficient scale and fidelity. These measurements will support the development of new predictive models.

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

PubMed

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

2016-08-01

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

Improvement in airborne position measurements based on an ultrasonic linear-period-modulated wave by 1-bit signal processing

NASA Astrophysics Data System (ADS)

Thong-un, Natee; Hirata, Shinnosuke; Kurosawa, Minoru K.

2015-07-01

In this paper, we describe an expansion of the airborne ultrasonic systems for object localization in the three-dimensional spaces of navigation. A system, which revises the microphone arrangement and algorithm, can expand the object-position measurement from +90° in a previous method up to +180° for both the elevation and azimuth angles. The proposed system consists of a sound source and four acoustical receivers. Moreover, the system is designed to utilize low-cost devices, and low-cost computation relying on 1-bit signal processing is used to support the real-time application on a field-programmable gate array (FPGA). An object location is identified using spherical coordinates. A spherical object, which has a curved surface, is considered a target for this system. The transmit pulse to the target is a linear-period-modulated ultrasonic wave with a chirp rate of 50-20 kHz. Statistical evaluation of this work is the experimental investigation under repeatability.

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…

Preliminary Study on Acoustic Detection of Faults Experienced by a High-Bypass Turbofan Engine

NASA Technical Reports Server (NTRS)

Boyle, Devin K.

2014-01-01

The vehicle integrated propulsion research (VIPR) effort conducted by NASA and several partners provided an unparalleled opportunity to test a relatively low TRL concept regarding the use of far field acoustics to identify faults occurring in a high bypass turbofan engine. Though VIPR Phase II ground based aircraft installed engine testing wherein a multitude of research sensors and methods were evaluated, an array of acoustic microphones was used to determine the viability of such an array to detect failures occurring in a commercially representative high bypass turbofan engine. The failures introduced during VIPR testing included commanding the engine's low pressure compressor (LPC) exit and high pressure compressor (HPC) 14th stage bleed values abruptly to their failsafe positions during steady state

Aeroacoustic Experiments in the NASA Langley Low-Turbulence Pressure Tunnel

NASA Technical Reports Server (NTRS)

Choudhari, Meelan M.; Lockard, David P.; Macaraeg, Michele G.; Singer, Bart A.; Streett, Craig L.; Neubert, Guy R.; Stoker, Robert W.; Underbrink, James R.; Berkman, Mert E.; Khorrami, Mehdi R.

2002-01-01

A phased microphone array was used in the NASA Langley Low-Turbulence Pressure Tunnel to obtain acoustic data radiating from high-lift wing configurations. The data included noise localization plots and acoustic spectra. The tests were performed at Reynolds numbers based on the cruise-wing chord, ranging from 3.6 x 10(exp 6) to 19.2 x 10(exp 6). The effects of Reynolds number were small and monotonic for Reynolds numbers above 7.2 x 10(exp 6).

Acoustic measurements from a rotor blade-vortex interaction noise experiment in the German-Dutch Wind Tunnel (DNW)

NASA Technical Reports Server (NTRS)

Martin, Ruth M.; Splettstoesser, W. R.; Elliott, J. W.; Schultz, K.-J.

1988-01-01

Acoustic data are presented from a 40 percent scale model of the 4-bladed BO-105 helicopter main rotor, measured in the large European aeroacoustic wind tunnel, the DNW. Rotor blade-vortex interaction (BVI) noise data in the low speed flight range were acquired using a traversing in-flow microphone array. The experimental apparatus, testing procedures, calibration results, and experimental objectives are fully described. A large representative set of averaged acoustic signals is presented.

Optical Fiber Infrasound Sensor Arrays: An Improved Alternative to Arrays of Rosette Wind Filters

NASA Astrophysics Data System (ADS)

Walker, Kristoffer; Zumberge, Mark; Dewolf, Scott; Berger, Jon; Hedlin, Michael

2010-05-01

A key difficulty in infrasound signal detection is the noise created by spatially incoherent turbulence that is usually present in wind. Increasing wind speeds correlate with increasing noise levels across the entire infrasound band. Optical fiber infrasound sensors (OFIS) are line microphones that instantaneously integrate pressure along their lengths with laser interferometry. Although the sensor has a very low noise floor, the promise of the sensor is in its effectiveness at reducing wind noise without the need for a network of interconnected pipes. We have previously shown that a single 90 m OFIS (spanning a line) is just as effective at reducing wind noise as a 70 m diameter rosette (covering a circular area). We have also empirically measured the infrasound response of the OFIS as a function of back azimuth, showing that it is well predicted by an analytical solution; the response is flat for broadside signals and similar to the rosette response for endfire signals. Using that analytical solution, we have developed beamforming techniques that permit the estimation of back azimuth using an array of OFIS arms as well as an array deconvolution technique that accurately stacks weighted versions of the recordings to obtain the original infrasound signal. We show how a slight modification to traditional array processing techniques can also be used with OFIS arrays to determine back azimuth, even for signal-to-noise ratios much less than 1. Recently several improvements to the OFIS instrumentation have been achieved. We have made an important modification to our interferometric technique that makes the interferometer insensitive to ambient temperature fluctuation. We are also developing a continuous real-time calibration system, which may eliminate the need for periodic array calibration efforts. We also report progress in comparing a newly installed 270 m long OFIS at Piñon Flat Observatory (PFO) to a collocated 70 m rosette of the I57US array. Specifically, we compare hundreds of wind noise spectra and two Vandenberg rocket launch infrasound signals that were recorded by both systems. The 70-m diameter rosette (L2) was connected to a Chaparral Physics Model 50 microphone, which is usually more sensitive than the MB2000 microbarometer in the 1-10 Hz band. The data show that in low wind, the noise floor of the OFIS is the same as the Chaparral. However, in moderate wind (5 m/s) the OFIS attenuates wind noise at 1 Hz by 10 dB better than L2. Similarly, the two rocket launch signals that were recorded in the presence of 3-4 m/s wind confirm that the signal-to-noise ratio improvement is 10 dB at 1 Hz. This confirms that at each signal frequency and direction, there exists an OFIS length threshold above which an OFIS wind filter will always outperform a rosette in terms recorded signal-to-noise ratio. The OFIS technology is proven and mature for observatory installations. Work is underway to make the technology more portable for remote, DC-powered deployments. A DC-powered, ruggedized OFIS array will be installed for microbarom research in Northern California during the spring of 2010. We seek collaborations with other researchers that are interested in evaluating or assisting in the further development of the OFIS technology.

Investigation of the infrasound produced by geophysical events such as volcanoes, thunder, and avalanches: the case for local infrasound monitoring (Invited)

NASA Astrophysics Data System (ADS)

Johnson, J. B.; Marcillo, O. E.; Arechiga, R. O.; Johnson, R.; Edens, H. E.; Marshall, H.; Havens, S.; Waite, G. P.

2010-12-01

Volcanoes, lightning, and mass wasting events generate substantial infrasonic energy that propagates for long distances through the atmosphere with generally low intrinsic attenuation. Although such sources are often studied with regional infrasound arrays that provide important records of their occurrence, position, and relative magnitudes these signals recorded at tens to hundreds of kilometers are often significantly affected by propagation effects. Complex atmospheric structure, due to heterogeneous winds and temperatures, and intervening topography can be responsible for multi-pathing, signal attenuation, and focusing or, alternatively, information loss (i.e., a shadow zone). At far offsets, geometric spreading diminishes signal amplitude requiring low noise recording sites and high fidelity microphones. In contrast recorded excess pressures at local distances are much higher in amplitude and waveforms are more representative of source phenomena. We report on recent studies of volcanoes, thunder, and avalanches made with networks and arrays of infrasound sensors deployed local (within a few km) to the source. At Kilauea Volcano (Hawaii) we deployed a network of ~50 infrasound sensitive sensors (flat from 50 s to 50 Hz) to track the coherence of persistent infrasonic tremor signals in the near-field (out to a few tens of kilometers). During periods of high winds (> 5-10 m/s) we found significant atmospheric influence for signals recorded at stations only a few kilometers from the source. Such observations have encouraged us to conduct a range of volcano, thunder, and snow avalanche studies with networks of small infrasound arrays (~30 m aperture) deployed close to the source region. We present results from local microphone deployments (12 sensors) at Santiaguito Volcano (Guatemala) where we are able to precisely (~10 m resolution) locate acoustic sources from explosions and rock falls. We also present results from our thunder mapping acoustic arrays (15 sensors) in the Magdalena Mountains of New Mexico capable of mapping lightning channels more than 10 km in extent whose positions are corroborated by the radio wave detecting lightning mapping array. We also discuss the recent implementation of a network of snow avalanche detection arrays (12 sensors) in Idaho that are used to monitor and track and map infrasound produced by moving sources. We contend that local infrasound deployment is analogous to local seismic monitoring in that it enables precision localization and interpretation of source phenomena.

Estimation of source location and ground impedance using a hybrid multiple signal classification and Levenberg-Marquardt approach

NASA Astrophysics Data System (ADS)

Tam, Kai-Chung; Lau, Siu-Kit; Tang, Shiu-Keung

2016-07-01

A microphone array signal processing method for locating a stationary point source over a locally reactive ground and for estimating ground impedance is examined in detail in the present study. A non-linear least square approach using the Levenberg-Marquardt method is proposed to overcome the problem of unknown ground impedance. The multiple signal classification method (MUSIC) is used to give the initial estimation of the source location, while the technique of forward backward spatial smoothing is adopted as a pre-processer of the source localization to minimize the effects of source coherence. The accuracy and robustness of the proposed signal processing method are examined. Results show that source localization in the horizontal direction by MUSIC is satisfactory. However, source coherence reduces drastically the accuracy in estimating the source height. The further application of Levenberg-Marquardt method with the results from MUSIC as the initial inputs improves significantly the accuracy of source height estimation. The present proposed method provides effective and robust estimation of the ground surface impedance.

A self-teaching image processing and voice-recognition-based, intelligent and interactive system to educate visually impaired children

NASA Astrophysics Data System (ADS)

Iqbal, Asim; Farooq, Umar; Mahmood, Hassan; Asad, Muhammad Usman; Khan, Akrama; Atiq, Hafiz Muhammad

2010-02-01

A self teaching image processing and voice recognition based system is developed to educate visually impaired children, chiefly in their primary education. System comprises of a computer, a vision camera, an ear speaker and a microphone. Camera, attached with the computer system is mounted on the ceiling opposite (on the required angle) to the desk on which the book is placed. Sample images and voices in the form of instructions and commands of English, Urdu alphabets, Numeric Digits, Operators and Shapes are already stored in the database. A blind child first reads the embossed character (object) with the help of fingers than he speaks the answer, name of the character, shape etc into the microphone. With the voice command of a blind child received by the microphone, image is taken by the camera which is processed by MATLAB® program developed with the help of Image Acquisition and Image processing toolbox and generates a response or required set of instructions to child via ear speaker, resulting in self education of a visually impaired child. Speech recognition program is also developed in MATLAB® with the help of Data Acquisition and Signal Processing toolbox which records and process the command of the blind child.

Aeroacoustic Measurements of a Wing/Slat Model

NASA Astrophysics Data System (ADS)

Mendoza, Jeff M.; Brooks, Thomas F.; Humphreys, William M.

2002-01-01

Aeroacoustic evaluations of high-lift devices have been carried out in the Quiet Flow Facility of the NASA Langley Research Center. The present paper deals with detailed flow and acoustic measurements that have been made to understand, and to possibly predict and reduce, the noise from a wing leading edge slat configuration. The acoustic database is obtained by a moveable Small Aperture Directional Array (SADA) of microphones designed to electronically steer to different portions of models under study. The slat is shown to be a uniform distributed noise source. The data was processed such that spectra and directivity were determined with respect to a one-foot span of slat. The spectra are normalized in various fashions to demonstrate slat noise character. In order to equate portions of the spectra to different slat noise components, trailing edge noise predictions using measured slat boundary layer parameters as inputs are compared to the measured slat noise spectra.

Volcanic Thunder From Explosive Eruptions at Bogoslof Volcano, Alaska

NASA Astrophysics Data System (ADS)

Haney, Matthew M.; Van Eaton, Alexa R.; Lyons, John J.; Kramer, Rebecca L.; Fee, David; Iezzi, Alexandra M.

2018-04-01

Lightning often occurs during ash-producing eruptive activity, and its detection is now being used in volcano monitoring for rapid alerts. We report on infrasonic and sonic recordings of the related, but previously undocumented, phenomenon of volcanic thunder. We observe volcanic thunder during the waning stages of two explosive eruptions at Bogoslof volcano, Alaska, on a microphone array located 60 km away. Thunder signals arrive from a different direction than coeruptive infrasound generated at the vent following an eruption on 10 June 2017, consistent with locations from lightning networks. For the 8 March 2017 eruption, arrival times and amplitudes of high-frequency thunder signals correlate well with the timing and strength of lightning detections. In both cases, the thunder is associated with lightning that continues after significant eruptive activity has ended. Infrasonic and sonic observations of volcanic thunder offer a new avenue for studying electrification processes in volcanic plumes.

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.

Time-Frequency Masking for Speech Separation and Its Potential for Hearing Aid Design

PubMed Central

Wang, DeLiang

2008-01-01

A new approach to the separation of speech from speech-in-noise mixtures is the use of time-frequency (T-F) masking. Originated in the field of computational auditory scene analysis, T-F masking performs separation in the time-frequency domain. This article introduces the T-F masking concept and reviews T-F masking algorithms that separate target speech from either monaural or binaural mixtures, as well as microphone-array recordings. The review emphasizes techniques that are promising for hearing aid design. This article also surveys recent studies that evaluate the perceptual effects of T-F masking techniques, particularly their effectiveness in improving human speech recognition in noise. An assessment is made of the potential benefits of T-F masking methods for the hearing impaired in light of the processing constraints of hearing aids. Finally, several issues pertinent to T-F masking are discussed. PMID:18974204

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

NASA Astrophysics Data System (ADS)

Kaiser, Zachary David Epping

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

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.

Fast PSP measurements of wall-pressure fluctuation in low-speed flows: improvements using proper orthogonal decomposition

NASA Astrophysics Data System (ADS)

Peng, Di; Wang, Shaofei; Liu, Yingzheng

2016-04-01

Fast pressure-sensitive paint (PSP) is very useful in flow diagnostics due to its fast response and high spatial resolution, but its applications in low-speed flows are usually challenging due to limitations of paint's pressure sensitivity and the capability of high-speed imagers. The poor signal-to-noise ratio in low-speed cases makes it very difficult to extract useful information from the PSP data. In this study, unsteady PSP measurements were made on a flat plate behind a cylinder in a low-speed wind tunnel (flow speed from 10 to 17 m/s). Pressure fluctuations (Δ P) on the plate caused by vortex-plate interaction were recorded continuously by fast PSP (using a high-speed camera) and a microphone array. Power spectrum of pressure fluctuations and phase-averaged Δ P obtained from PSP and microphone were compared, showing good agreement in general. Proper orthogonal decomposition (POD) was used to reduce noise in PSP data and extract the dominant pressure features. The PSP results reconstructed from selected POD modes were then compared to the pressure data obtained simultaneously with microphone sensors. Based on the comparison of both instantaneous Δ P and root-mean-square of Δ P, it was confirmed that POD analysis could effectively remove noise while preserving the instantaneous pressure information with good fidelity, especially for flows with strong periodicity. This technique extends the application range of fast PSP and can be a powerful tool for fundamental fluid mechanics research at low speed.

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.

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.

Speech intelligibility in noise using throat and acoustic microphones.

PubMed

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.

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

PubMed

Desjardins, Jamie L

2016-01-01

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

Acoustical Direction Finding with Time-Modulated Arrays

PubMed Central

Clark, Ben; Flint, James A.

2016-01-01

Time-Modulated Linear Arrays (TMLAs) offer useful efficiency savings over conventional phased arrays when applied in parameter estimation applications. The present paper considers the application of TMLAs to acoustic systems and proposes an algorithm for efficiently deriving the arrival angle of a signal. The proposed technique is applied in the frequency domain, where the signal and harmonic content is captured. Using a weighted average method on harmonic amplitudes and their respective main beam angles, it is possible to determine an estimate for the signal’s direction of arrival. The method is demonstrated and evaluated using results from both numerical and practical implementations and performance data is provided. The use of Micro-Electromechanical Systems (MEMS) sensors allows time-modulation techniques to be applied at ultrasonic frequencies. Theoretical predictions for an array of five isotropic elements with half-wavelength spacing and 1000 data samples suggest an accuracy of ±1∘ within an angular range of approximately ±50∘. In experiments of a 40 kHz five-element microphone array, a Direction of Arrival (DoA) estimation within ±2.5∘ of the target signal is readily achieved inside a ±45∘ range using a single switched input stage and a simple hardware setup. PMID:27973432

Origami acoustics: using principles of folding structural acoustics for simple and large focusing of sound energy

NASA Astrophysics Data System (ADS)

Harne, Ryan L.; Lynd, Danielle T.

2016-08-01

Fixed in spatial distribution, arrays of planar, electromechanical acoustic transducers cannot adapt their wave energy focusing abilities unless each transducer is externally controlled, creating challenges for the implementation and portability of such beamforming systems. Recently, planar, origami-based structural tessellations are found to facilitate great versatility in system function and properties through kinematic folding. In this research we bridge the physics of acoustics and origami-based design to discover that the simple topological reconfigurations of a Miura-ori-based acoustic array yield many orders of magnitude worth of reversible change in wave energy focusing: a potential for acoustic field morphing easily obtained through deployable, tessellated architectures. Our experimental and theoretical studies directly translate the roles of folding the tessellated array to the adaptations in spectral and spatial wave propagation sensitivities for far field energy transmission. It is shown that kinematic folding rules and flat-foldable tessellated arrays collectively provide novel solutions to the long-standing challenges of conventional, electronically-steered acoustic beamformers. While our examples consider sound radiation from the foldable array in air, linear acoustic reciprocity dictates that the findings may inspire new innovations for acoustic receivers, e.g. adaptive sound absorbers and microphone arrays, as well as concepts that include water-borne waves.

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.

Analysis of noise from reusable solid rocket motor firings

NASA Astrophysics Data System (ADS)

Jerome, Trevor W.; Gee, Kent L.; Neilsen, Tracianne B.

2012-10-01

As part of investigations into the design of next-generation launch vehicles, near and far-field data were collected during horizontal static firings of reusable solid rocket motors. Spatial variation of overall and one-third octave band pressure levels at sideline and polar arc arrays is analyzed. Spectra at individual microphone locations were analyzed. Positively-skewed pressure waveforms were observed in the probability density functions. Extreme skewness in the first-order estimate of the time derivative was found as a result of the presence of significant acoustic shocks.

How hummingbirds hum: Acoustic holography of hummingbirds during maneuvering flight

NASA Astrophysics Data System (ADS)

Hightower, Ben; Wijnings, Patrick; Ingersoll, Rivers; Chin, Diana; Scholte, Rick; Lentink, David

2017-11-01

Hummingbirds make a characteristic humming sound when they flap their wings. The physics and the biological significance of hummingbird aeroacoustics is still poorly understood. We used acoustic holography and high-speed cameras to determine the acoustic field of six hummingbirds while they either hovered stationary in front of a flower or maneuvered to track flower motion. We used a robotic flower that oscillated either laterally or longitudinally with a linear combination of 20 different frequencies between 0.2 and 20 Hz, a range that encompasses natural flower vibration frequencies in wind. We used high-speed marker tracking to dissect the transfer function between the moving flower, the head, and body of the bird. We also positioned four acoustic arrays equipped with 2176 microphones total above, below, and in front of the hummingbird. Acoustic data from the microphones were back-propagated to planes adjacent to the hummingbird to create the first real-time holograms of the pressure field a hummingbird generates in vivo. Integration of all this data offers insight into how hummingbirds modulate the acoustic field during hovering and maneuvering flight.

Coherent active methods for applications in room acoustics.

PubMed

Guicking, D; Karcher, K; Rollwage, M

1985-10-01

An adjustment of reverberation time in rooms is often desired, even for low frequencies where passive absorbers fail. Among the active (electroacoustic) systems, incoherent ones permit lengthening of reverberation time only, whereas coherent active methods will allow sound absorption as well. A coherent-active wall lining consists of loudspeakers with microphones in front and adjustable control electronics. The microphones pick up the incident sound and drive the speakers in such a way that the reflection coefficient takes on prescribed values. An experimental device for the one-dimensional case allows reflection coefficients between almost zero and about 1.5 to be realized below 1000 Hz. The extension to three dimensions presents problems, especially by nearfield effects. Experiments with a 3 X 3 loudspeaker array and computer simulations proved that the amplitude reflection coefficient can be adjusted between 10% and 200% for sinusoidal waves at normal and oblique incidence. Future developments have to make the system work with broadband excitation and in more diffuse sound fields. It is also planned to combine the active reverberation control with active diffusion control.

Listening for bats: the hearing range of the bushcricket Phaneroptera falcata for bat echolocation calls measured in the field.

PubMed Central

Schul, J; Matt, F; von Helversen, O

2000-01-01

The hearing range of the tettigoniid Phaneropterafalcata for the echolocation calls of freely flying mouseeared bats (Myotis myotis) was determined in the field. The hearing of the insect was monitored using hook electrode recordings from an auditory interneuron, which is as sensitive as the hearing organ for frequencies above 16 kHz. The flight path of the bat relative to the insect's position was tracked by recording the echolocation calls with two microphone arrays, and calculating the bat's position from the arrival time differences of the calls at each microphone. The hearing distances ranged from 13 to 30 m. The large variability appeared both between different insects and between different bat approaches to an individual insect. The escape time of the bushcricket, calculated from the detection distance of the insect and the instantaneous flight speed of the bat, ranged from 1.5 to more than 4s. The hearing ranges of bushcrickets suggest that the insect hears the approaching bat long before the bat can detect an echo from the flying insect. PMID:12233766

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.

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.

Development of an Experimental Rig for Investigation of Higher Order Modes in Ducts

NASA Technical Reports Server (NTRS)

Gerhold, Carl H.; Cabell, Randolph H.; Brown, Martha C.

2006-01-01

Continued progress to reduce fan noise emission from high bypass ratio engine ducts in aircraft increasingly relies on accurate description of the sound propagation in the duct. A project has been undertaken at NASA Langley Research Center to investigate the propagation of higher order modes in ducts with flow. This is a two-pronged approach, including development of analytic models (the subject of a separate paper) and installation of a laboratory-quality test rig. The purposes of the rig are to validate the analytical models and to evaluate novel duct acoustic liner concepts, both passive and active. The dimensions of the experimental rig test section scale to between 25% and 50% of the aft bypass ducts of most modern engines. The duct is of rectangular cross section so as to provide flexibility to design and fabricate test duct liner samples. The test section can accommodate flow paths that are straight through or offset from inlet to discharge, the latter design allowing investigation of the effect of curvature on sound propagation and duct liner performance. The maximum air flow rate through the duct is Mach 0.3. Sound in the duct is generated by an array of 16 high-intensity acoustic drivers. The signals to the loudspeaker array are generated by a multi-input/multi-output feedforward control system that has been developed for this project. The sound is sampled by arrays of flush-mounted microphones and a modal decomposition is performed at the frequency of sound generation. The data acquisition system consists of two arrays of flush-mounted microphones, one upstream of the test section and one downstream. The data are used to determine parameters such as the overall insertion loss of the test section treatment as well as the effect of the treatment on a modal basis such as mode scattering. The methodology used for modal decomposition is described, as is a description of the mode generation control system. Data are presented which demonstrate the performance of the controller to generate the desired mode while suppressing all other cut on modes in the duct.

Towards a sub 15-dBA optical micromachined microphone

PubMed Central

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

Benefits of the fiber optic versus the electret microphone in voice amplification.

PubMed

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.

Truck acoustic data analyzer system

DOEpatents

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.

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

NASA Astrophysics Data System (ADS)

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

2015-03-01

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

Adaptive Suppression of Noise in Voice Communications

NASA Technical Reports Server (NTRS)

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

2003-01-01

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

Investigation of Twin Jet Aeroacoustic Properties in the Presence of a Hybrid Wing Body Shield

NASA Technical Reports Server (NTRS)

Doty, Michael J.

2012-01-01

In preparation for upcoming wind tunnel acoustic experiments of a Hybrid Wing Body (HWB) vehicle with two jet engine simulator units, a series of twin jet aeroacoustic investigations were conducted leading to increased understanding and risk mitigation. A previously existing twin jet nozzle system and a fabricated HWB aft deck fuselage are combined for a 1.9% model scale study of jet nozzle spacing and jet cant angle effects, elevon deflection into the jet plume, and acoustic shielding by the fuselage body. Linear and phased array microphone measurements are made, and data processing includes the use of DAMAS (Deconvolution Approach for the Mapping of Acoustic Sources). Closely-spaced twin jets with a 5 inward cant angle exhibit reduced noise levels compared to their parallel flow counterparts at similar and larger nozzle spacings. A 40 elevon deflection into the twin jet plume, which is required for HWB ground rotation, can significantly increase upstream noise levels (more than 5 dB OASPL) with only minimal increases in the downstream direction. Lastly, DAMAS processing can successfully measure the noise source distribution of multiple shielded jet sources.

In situ calibration of atmospheric-infrasound sensors including the effects of wind-noise-reduction pipe systems.

PubMed

Gabrielson, Thomas B

2011-09-01

A worldwide network of more than 40 infrasound monitoring stations has been established as part of the effort to ensure compliance with the Comprehensive Nuclear Test Ban Treaty. Each station has four to eight individual infrasound elements in a kilometer-scale array for detection and bearing determination of acoustic events. The frequency range of interest covers a three-decade range-roughly from 0.01 to 10 Hz. A typical infrasound array element consists of a receiving transducer connected to a multiple-inlet pipe network to average spatially over the short-wavelength turbulence-associated "wind noise." Although the frequency response of the transducer itself may be known, the wind-noise reduction system modifies that response. In order to understand the system's impact on detection and identification of acoustical events, the overall frequency response must be determined. This paper describes a technique for measuring the absolute magnitude and phase of the frequency response of an infrasound element including the wind-noise-reduction piping by comparison calibration using ambient noise and a reference-microphone system. Measured coherence between the reference and the infrasound element and the consistency between the magnitude and the phase provide quality checks on the process. © 2011 Acoustical Society of America

En route noise levels from propfan test assessment airplane

NASA Technical Reports Server (NTRS)

Garber, Donald P.; Willshire, William L., Jr.

1994-01-01

The en route noise test was designed to characterize propagation of propfan noise from cruise altitudes to the ground. In-flight measurements of propfan source levels and directional patterns were made by a chase plane flying in formation with the propfan test assessment (PTA) airplane. Ground noise measurements were taken during repeated flights over a distributed microphone array. The microphone array on the ground was used to provide ensemble-averaged estimates of mean flyover noise levels, establish confidence limits for those means, and measure propagation-induced noise variability. Even for identical nominal cruise conditions, peak sound levels for individual overflights varied substantially about the average, particularly when overflights were performed on different days. Large day-to-day variations in peak level measurements appeared to be caused by large day-to-day differences in propagation conditions and tended to obscure small variations arising from operating conditions. A parametric evaluation of the sensitivity of this prediction method to weather measurement and source level uncertainties was also performed. In general, predictions showed good agreement with measurements. However, the method was unable to predict short-term variability of ensemble-averaged data within individual overflights. Although variations in absorption appear to be the dominant factor in variations of peak sound levels recorded on the ground, accurate predictions of those levels require that a complete description of operational conditions be taken into account. The comprehensive and integrated methods presented in this paper have adequately predicted ground-measured sound levels. On average, peak sound levels were predicted within 3 dB for each of the three different cruise conditions.

Preliminary results from the White Sands Missile Range sonic boom propagation experiment

NASA Technical Reports Server (NTRS)

Willshire, William L., Jr.; Devilbiss, David W.

1992-01-01

Sonic boom bow shock amplitude and rise time statistics from a recent sonic boom propagation experiment are presented. Distributions of bow shock overpressure and rise time measured under different atmospheric turbulence conditions for the same test aircraft are quite different. The peak overpressure distributions are skewed positively, indicating a tendency for positive deviations from the mean to be larger than negative deviations. Standard deviations of overpressure distributions measured under moderate turbulence were 40 percent larger than those measured under low turbulence. As turbulence increased, the difference between the median and the mean increased, indicating increased positive overpressure deviations. The effect of turbulence was more readily seen in the rise time distributions. Under moderate turbulence conditions, the rise time distribution means were larger by a factor of 4 and the standard deviations were larger by a factor of 3 from the low turbulence values. These distribution changes resulted in a transition from a peaked appearance of the rise time distribution for the morning to a flattened appearance for the afternoon rise time distributions. The sonic boom propagation experiment consisted of flying three types of aircraft supersonically over a ground-based microphone array with concurrent measurements of turbulence and other meteorological data. The test aircraft were a T-38, an F-15, and an F-111, and they were flown at speeds of Mach 1.2 to 1.3, 30,000 feet above a 16 element, linear microphone array with an inter-element spacing of 200 ft. In two weeks of testing, 57 supersonic passes of the test aircraft were flown from early morning to late afternoon.

Location of acoustic radiators and inversion for energy density using radio-frequency sources and thunder recordings

NASA Astrophysics Data System (ADS)

Anderson, J.; Johnson, J. B.; Arechiga, R. O.; Edens, H. E.; Thomas, R. J.

2011-12-01

We use radio frequency (VHF) pulse locations mapped with the New Mexico Tech Lightning Mapping Array (LMA) to study the distribution of thunder sources in lightning channels. A least squares inversion is used to fit channel acoustic energy radiation with broadband (0.01 to 500 Hz) acoustic recordings using microphones deployed local (< 10 km) to the lightning. We model the thunder (acoustic) source as a superposition of line segments connecting the LMA VHF pulses. An optimum branching algorithm is used to reconstruct conductive channels delineated by VHF sources, which we discretize as a superposition of finely-spaced (0.25 m) acoustic point sources. We consider total radiated thunder as a weighted superposition of acoustic waves from individual channels, each with a constant current along its length that is presumed to be proportional to acoustic energy density radiated per unit length. Merged channels are considered as a linear sum of current-carrying branches and radiate proportionally greater acoustic energy. Synthetic energy time series for a given microphone location are calculated for each independent channel. We then use a non-negative least squares inversion to solve for channel energy densities to match the energy time series determined from broadband acoustic recordings across a 4-station microphone network. Events analyzed by this method have so far included 300-1000 VHF sources, and correlations as high as 0.5 between synthetic and recorded thunder energy were obtained, despite the presence of wind noise and 10-30 m uncertainty in VHF source locations.

Method of fan sound mode structure determination computer program user's manual: Microphone location program

NASA Technical Reports Server (NTRS)

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

1977-01-01

A computer user's manual describing the operation and the essential features of the 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.

The impact of listening with directional microphone technology on self-perceived localization disabilities and handicaps.

PubMed

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.

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

PubMed

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

2018-04-01

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

Energy and Power Spectra of Thunder in the Magdalena Mountains, Central New Mexico

NASA Astrophysics Data System (ADS)

Johnson, R. L.; Johnson, J. B.; Arechiga, R. O.; Michnovicz, J. C.; Edens, H. E.; Rison, W.

2011-12-01

Thunder is generated primarily by heating and expansion of the atmosphere around a lightning channel and by charge relaxation within a cloud. Broadband acoustic studies are important for inferring dynamic charge behavior during and after lightning events. During the Summer monsoon seasons of 2009-2011, we deployed networks of 3-5 stations consisting of broadband (0.01 to 500 Hz) acoustic arrays and audio microphones in the Magdalena Mountains in central New Mexico. We utilize Lightning Mapping Array (LMA) data for accurate timing of lightning events within a 10 km radius of our network. Unlike the LMA, which detects VHF signals from breakdown processes, thunder signals may be used to observe charge dynamics and thermal shocking of the atmosphere. Previous investigations show that thunder spectral content may distinguish between electrostatic and thermal heating processes. We collected extensive datasets in terms of number of independent broadband sensors (up to 20), number of observed flashes (hundreds from multiple storms), and available coincident LMA data. We use infrasound and audio data to quantify total acoustic energy produced at lightning sources in various frequency bands. We attribute the spectral content and intensity of thunder signals to source characteristics, sensor locations, propagation effects, and noise. We observe variations in acoustic energy for both entire storm systems and individual lightning flashes. We propose that some variations may be related to the type of lightning flash and that spectral content is important for distinguishing between thunder generation mechanisms.

Long-Term Stability of One-Inch Condenser Microphones Calibrated at the National Institute of Standards and Technology

PubMed Central

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

Acoustic Detection Of Loose Particles In Pressure Sensors

NASA Technical Reports Server (NTRS)

Kwok, Lloyd C.

1995-01-01

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

Frequency multiplexed superconducting quantum interference device readout of large bolometer arrays for cosmic microwave background measurements.

PubMed

Dobbs, M A; Lueker, M; Aird, K A; Bender, A N; Benson, B A; Bleem, L E; Carlstrom, J E; Chang, C L; Cho, H-M; Clarke, J; Crawford, T M; Crites, A T; Flanigan, D I; de Haan, T; George, E M; Halverson, N W; Holzapfel, W L; Hrubes, J D; Johnson, B R; Joseph, J; Keisler, R; Kennedy, J; Kermish, Z; Lanting, T M; Lee, A T; Leitch, E M; Luong-Van, D; McMahon, J J; Mehl, J; Meyer, S S; Montroy, T E; Padin, S; Plagge, T; Pryke, C; Richards, P L; Ruhl, J E; Schaffer, K K; Schwan, D; Shirokoff, E; Spieler, H G; Staniszewski, Z; Stark, A A; Vanderlinde, K; Vieira, J D; Vu, C; Westbrook, B; Williamson, R

2012-07-01

A technological milestone for experiments employing transition edge sensor bolometers operating at sub-Kelvin temperature is the deployment of detector arrays with 100s-1000s of bolometers. One key technology for such arrays is readout multiplexing: the ability to read out many sensors simultaneously on the same set of wires. This paper describes a frequency-domain multiplexed readout system which has been developed for and deployed on the APEX-SZ and South Pole Telescope millimeter wavelength receivers. In this system, the detector array is divided into modules of seven detectors, and each bolometer within the module is biased with a unique ∼MHz sinusoidal carrier such that the individual bolometer signals are well separated in frequency space. The currents from all bolometers in a module are summed together and pre-amplified with superconducting quantum interference devices operating at 4 K. Room temperature electronics demodulate the carriers to recover the bolometer signals, which are digitized separately and stored to disk. This readout system contributes little noise relative to the detectors themselves, is remarkably insensitive to unwanted microphonic excitations, and provides a technology pathway to multiplexing larger numbers of sensors.

A wavenumber approach to analysing the active control of plane waves with arrays of secondary sources

NASA Astrophysics Data System (ADS)

Elliott, Stephen J.; Cheer, Jordan; Bhan, Lam; Shi, Chuang; Gan, Woon-Seng

2018-04-01

The active control of an incident sound field with an array of secondary sources is a fundamental problem in active control. In this paper the optimal performance of an infinite array of secondary sources in controlling a plane incident sound wave is first considered in free space. An analytic solution for normal incidence plane waves is presented, indicating a clear cut-off frequency for good performance, when the separation distance between the uniformly-spaced sources is equal to a wavelength. The extent of the near field pressure close to the source array is also quantified, since this determines the positions of the error microphones in a practical arrangement. The theory is also extended to oblique incident waves. This result is then compared with numerical simulations of controlling the sound power radiated through an open aperture in a rigid wall, subject to an incident plane wave, using an array of secondary sources in the aperture. In this case the diffraction through the aperture becomes important when its size is compatible with the acoustic wavelength, in which case only a few sources are necessary for good control. When the size of the aperture is large compared to the wavelength, and diffraction is less important but more secondary sources need to be used for good control, the results then become similar to those for the free field problem with an infinite source array.

Response Identification in the Extremely Low Frequency Region of an Electret Condenser Microphone

PubMed Central

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

Response identification in the extremely low frequency region of an electret condenser microphone.

PubMed

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.

Application of acoustic imaging techniques on snowmobile pass-by noise.

PubMed

Padois, Thomas; Berry, Alain

2017-02-01

Snowmobile manufacturers invest important efforts to reduce the noise emission of their products. The noise sources of snowmobiles are multiple and closely spaced, leading to difficult source separation in practice. In this study, source imaging results for snowmobile pass-by noise are discussed. The experiments involve a 193-microphone Underbrink array, with synchronization of acoustic with video data provided by a high-speed camera. Both conventional beamforming and Clean-SC deconvolution are implemented to provide noise source maps of the snowmobile. The results clearly reveal noise emission from the engine, exhaust, and track depending on the frequency range considered.

Evaluation of aero commander propeller acoustic data: Taxi operations

NASA Technical Reports Server (NTRS)

Piersol, A. G.; Wilby, E. G.; Wilby, J. F.

1979-01-01

The acoustic data from ground tests performed on an Aero Commander propeller driven aircraft are analyzed. An array of microphones flush mounted on the side of the fuselage were used to record data. The propeller blade passage noise during operations at several different taxi speeds is considered and calculations of the magnitude and phase of the blade passage tones, the amplitude stability of the tones, and the spatial phase and coherence of the tones are included. The measured results are compared to theoretical predictions for propeller noise and various evaluations which reveal important details of propeller noise characteristics are presented.

[An implantable microphone for electronic hearing aids].

PubMed

Leysieffer, H; Müller, G; Zenner, H P

1997-10-01

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

Aeroacoustic Measurements of a Wing/Slat Model. [Research conducted at the Quiet Flow Facility of the NASA Langley Research Center

NASA Technical Reports Server (NTRS)

Mendoza, Jeff M.; Brooks, Thomas F.; Humphreys, William M.

2002-01-01

Aeroacoustic evaluations of high-lift devices have been carried out in the Quiet Flow Facility of the NASA Langley Research Center. The present paper deals with detailed flow and acoustic measurements that have been made to understand, and to possibly predict and reduce, the noise from a wing leading edge slat configuration. The acoustic database is obtained by a moveable Small Aperture Directional Array (SADA) of microphones designed to electronically steer to different portions of models under study. The slat is shown to be a uniform distributed noise source. The data was processed such that spectra and directivity were determined with respect to a one-foot span of slat. The spectra are normalized in various fashions to demonstrate slat noise character. In order to equate portions of the spectra to different slat noise components, trailing edge noise predictions using measured slat boundary layer parameters as inputs are compared to the measured slat noise spectra.

Experimental investigation of trailing edge noise from stationary and rotating airfoils

PubMed Central

Zajamsek, Branko; Doolan, Con J.; Moreau, Danielle J.; Fischer, Jeoffrey; Prime, Zebb

2017-01-01

Trailing edge noise from stationary and rotating NACA 0012 airfoils is characterised and compared with a noise prediction based on the semi-empirical Brooks, Pope, and Marcolini (BPM) model. The NACA 0012 is symmetrical airfoil with no camber and 12% thickness to chord length ratio. Acoustic measurements were conducted in an anechoic wind tunnel using a stationary NACA 0012 airfoil at 0° pitch angle. Airfoil self-noise emissions from rotating NACA 0012 airfoils mounted at 0° and 10° pitch angles on a rotor-rig are studied in an anechoic room. The measurements were carried out using microphone arrays for noise localisation and magnitude estimation using beamforming post-processing. Results show good agreement between peak radiating trailing edge noise emissions of stationary and rotating NACA 0012 airfoils in terms of the Strouhal number. Furthermore, it is shown that noise predictions based on the BPM model considering only two dimensional flow effects, are in good agreement with measurements for rotating airfoils, at these particular conditions. PMID:28599535

Experimental investigation of trailing edge noise from stationary and rotating airfoils.

PubMed

Zajamsek, Branko; Doolan, Con J; Moreau, Danielle J; Fischer, Jeoffrey; Prime, Zebb

2017-05-01

Trailing edge noise from stationary and rotating NACA 0012 airfoils is characterised and compared with a noise prediction based on the semi-empirical Brooks, Pope, and Marcolini (BPM) model. The NACA 0012 is symmetrical airfoil with no camber and 12% thickness to chord length ratio. Acoustic measurements were conducted in an anechoic wind tunnel using a stationary NACA 0012 airfoil at 0° pitch angle. Airfoil self-noise emissions from rotating NACA 0012 airfoils mounted at 0° and 10° pitch angles on a rotor-rig are studied in an anechoic room. The measurements were carried out using microphone arrays for noise localisation and magnitude estimation using beamforming post-processing. Results show good agreement between peak radiating trailing edge noise emissions of stationary and rotating NACA 0012 airfoils in terms of the Strouhal number. Furthermore, it is shown that noise predictions based on the BPM model considering only two dimensional flow effects, are in good agreement with measurements for rotating airfoils, at these particular conditions.

An Aeroacoustic Study of a Leading Edge Slat Configuration

NASA Technical Reports Server (NTRS)

Mendoza, J. M.; Brooks, T. F.; Humphreys, W. M., Jr.

2002-01-01

Aeroacoustic evaluations of high-lift devices have been carried out in the Quiet Flow Facility of the NASA Langley Research Center. The present paper describes detailed flow and acoustic measurements that have been made in order to better understand the noise generated from airflow over a wing leading edge slat configuration, and to possibly predict and reduce this noise source. The acoustic database is obtained by a moveable Small Aperture Directional Array of microphones designed to electronically steer to different portions of models under study. The slat is shown to be a uniform distributed noise source. The data was processed such that spectra and directivity were determined with respect to a one-foot span of slat. The spectra are normalized in various fashions to demonstrate slat noise character. In order to equate portions of the spectra to different slat noise components, trailing edge noise predictions using measured slat boundary layer parameters as inputs are compared to the measured slat noise spectra.

F-16XL and F-18 High Speed Acoustic Flight Test Databases

NASA Technical Reports Server (NTRS)

Kelly, J. J.; Wilson, M. R.; Rawls, J., Jr.; Norum, T. D.; Golub, R. A.

1999-01-01

This report presents the recorded acoustic data and the computed narrow-band and 1/3-octave band spectra produced by F-18 and F-16XL aircraft in subsonic flight over an acoustic array. Both broadband-shock noise and turbulent mixing noise are observed in the spectra. Radar and c-band tracking systems provided the aircraft position which enabled directivity and smear angles from the aircraft to each microphone to be computed. These angles are based on source emission time and thus give some idea about the directivity of the radiated sound field due to jet noise. A follow-on static test was also conducted where acoustic and engine data were obtained. The acoustic data described in the report has application to community noise analysis, noise source characterization and validation of prediction models. A detailed description of the signal processing procedures is provided. Follow-on static tests of each aircraft were also conducted for which engine data and far-field acoustic data are presented.

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

ERIC Educational Resources Information Center

Janota, Claus P.; Janota, Jeanette Olach

1991-01-01

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

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.

Infrasound in the middle stratosphere measured with a free-flying acoustic array

NASA Astrophysics Data System (ADS)

Bowman, Daniel C.; Lees, Jonathan M.

2015-11-01

Infrasound recorded in the middle stratosphere suggests that the acoustic wavefield above the Earth's surface differs dramatically from the wavefield near the ground. In contrast to nearby surface stations, the balloon-borne infrasound array detected signals from turbulence, nonlinear ocean wave interactions, building ventilation systems, and other sources that have not been identified yet. Infrasound power spectra also bore little resemblance to spectra recorded on the ground at the same time. Thus, sensors on the Earth's surface likely capture a fraction of the true diversity of acoustic waves in the atmosphere. Future studies building upon this experiment may quantify the acoustic energy flux from the surface to the upper atmosphere, extend the capability of the International Monitoring System to detect nuclear explosions, and lay the observational groundwork for a recently proposed mission to detect earthquakes on Venus using free-flying microphones.

Beamforming synthesis of binaural responses from computer simulations of acoustic spaces.

PubMed

Poletti, Mark A; Svensson, U Peter

2008-07-01

Auditorium designs can be evaluated prior to construction by numerical modeling of the design. High-accuracy numerical modeling produces the sound pressure on a rectangular grid, and subjective assessment of the design requires auralization of the sampled sound field at a desired listener position. This paper investigates the production of binaural outputs from the sound pressure at a selected number of grid points by using a least squares beam forming approach. Low-frequency axisymmetric emulations are derived by assuming a solid sphere model of the head, and a spherical array of 640 microphones is used to emulate ten measured head-related transfer function (HRTF) data sets from the CIPIC database for half the audio bandwidth. The spherical array can produce high-accuracy band-limited emulation of any human subject's measured HRTFs for a fixed listener position by using individual sets of beam forming impulse responses.

Infrasound from thunder: A natural seismic source

NASA Astrophysics Data System (ADS)

Lin, Ting-L.; Langston, Charles A.

2007-07-01

A small array consisting of five three-component short-period surface seismometers, a three-component borehole seismometer, and five infrasound microphones was built to investigate thunder-induced ground motions. Data from two thunder events with similar N-wave waveforms but different horizontal slownesses are chosen as examples of data collected by the array. These impulsive acoustic waves excited P and S reverberations in the near surface that depend on both the incident wave horizontal slowness and the velocity structure in the upper 30 meters at the site. Although the depth of the borehole is relatively shallow compared to a seismic wave wavelength, velocity amplitude in the radial component decays as much as 63 percent with depth but vertical component amplitudes are unaffected consistent with air-coupled Rayleigh wave excitation. Naturally occurring thunder appears to be a useful seismic source to empirically determine site resonance characteristics for hazards assessments.

Imaging thunder

NASA Astrophysics Data System (ADS)

Johnson, J. B.; Arechiga, R. O.; Thomas, R. J.; Edens, H. E.; Anderson, J.; Johnson, R. L.

2011-12-01

We use a network of broadband microphones, including a 4-element array, to locate the sources of thunder occurring during an electrical storm in central New Mexico on July 24th, 2009. Combined slowness search and distance ranging are used to identify thunder regions in three dimensions (out to 12 km) and for two overlapping frequency bands (1-10 and 4-40 Hz). Distinct thunder pulses are locatable and used to predict time-of-arrival to neighboring stations and to identify correlated phases across the network. Spatial correlation is also found between the thunder source regions and regions of VHF radiation as located by the New Mexico Lightning Mapping Array (LMA). Some of the misfit between the LMA and thunder locations is attributable to differences in excitation mechanisms of the respective radiation, which is related to current impulses in lightning channels (for thunder) and incremental ionization of the atmosphere (for VHF emissions).

Imaging thunder

NASA Astrophysics Data System (ADS)

Johnson, J. B.; Arechiga, R. O.; Thomas, R. J.; Edens, H. E.; Anderson, J.; Johnson, R.

2011-10-01

We use a network of broadband microphones, including a 4-element array, to locate the sources of thunder occurring during an electrical storm in central New Mexico on July 24th, 2009. Combined slowness search and distance ranging are used to identify thunder regions in three dimensions (out to 12 km) and for two overlapping frequency bands (1-10 and 4-40 Hz). Distinct thunder pulses are locatable and used to predict time-of-arrival to neighboring stations and to identify correlated phases across the network. Spatial correlation is also found between the thunder source regions and regions of very high frequency (VHF) radiation as located by the New Mexico Lightning Mapping Array (LMA). Some of the misfit between the LMA and thunder locations is attributable to differences in excitation mechanisms of the respective radiation, which is related to current impulses in lightning channels (for thunder) and incremental ionization of the atmosphere (for VHF emissions).

STS-97 crew meets with the media at Launch Pad 39B

NASA Technical Reports Server (NTRS)

2000-01-01

From the slidewire landing zone at Launch Pad 39B, STS-97 Mission Specialist Carlos Noriega (at right, with microphone) describes the mission for the media. Next to him are Mission Specialists Joe Tanner (left) and Marc Garneau (center). The crew is at KSC to take part in Terminal Countdown Demonstration Test activities that include emergency egress training, familiarization with the payload, and a simulated launch countdown. The other crew members are Commander Brent Jett and Pilot Mike Bloomfield. Mission STS- 97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:05 p.m. EST.

Design and Evaluation of Modifications to the NASA Langley Flow Impedance Tube

NASA Technical Reports Server (NTRS)

Jones, Michael G.; Watson, Willie R.; Parrott, Tony L.; Smith, Charles D.

2004-01-01

The need to minimize fan noise radiation from commercial aircraft engine nacelles continues to provide an impetus for developing new acoustic liner concepts. If the full value of such concepts is to be attained, an understanding of grazing flow effects is crucial. Because of this need for improved understanding of grazing flow effects, the NASA Langley Research Center Liner Physics Group has invested a large effort over the past decade into the development of a 2-D finite element method that characterizes wave propagation through a lined duct. The original test section in the Langley Grazing IncidenceTube was used to acquire data needed for implementation of this finite element method. This test section employed a stepper motor-driven axial-traversing bar, embedded in the wall opposite the test liner, to position a flush-mounted microphone at pre-selected locations. Complex acoustic pressure data acquired with this traversing microphone were used to educe the acoustic impedance of test liners using this 2-D finite element method and a local optimization technique. Results acquired in this facility have been extensively reported, and were compared with corresponding results from various U.S. aeroacoustics laboratories in the late 1990 s. Impedance data comparisons acquired from this multi-laboratory study suggested that it would be valuable to incorporate more realistic 3-D aeroacoustic effects into the impedance eduction methodology. This paper provides a description of modifications that have been implemented to facilitate studies of 3-D effects. The two key features of the modified test section are (1) the replacement of the traversing bar and its flush-mounted microphone with an array of 95 fixed-location microphones that are flush-mounted in all four walls of the duct, and (2) the inclusion of a suction device to modify the boundary layer upstream of the lined portion of the duct. The initial results achieved with the modified test section are provided in this report, and a comparison of these results with those achieved using the original test section is used to demonstrate that the data acquisition and analysis with the new test section can be confidently used for impedance eduction.

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.

Inconspicuous echolocation in hoary bats (Lasiurus cinereus).

PubMed

Corcoran, Aaron J; Weller, Theodore J

2018-05-16

Echolocation allows bats to occupy diverse nocturnal niches. Bats almost always use echolocation, even when other sensory stimuli are available to guide navigation. Here, using arrays of calibrated infrared cameras and ultrasonic microphones, we demonstrate that hoary bats ( Lasiurus cinereus ) use previously unknown echolocation behaviours that challenge our current understanding of echolocation. We describe a novel call type ('micro' calls) that has three orders of magnitude less sound energy than other bat calls used in open habitats. We also document bats flying close to microphones (less than 3 m) without producing detectable echolocation calls. Acoustic modelling indicates that bats are not producing calls that exceed 70-75 dB at 0.1 m, a level that would have little or no known use for a bat flying in the open at speeds exceeding 7 m s -1 This indicates that hoary bats sometimes fly without echolocation. We speculate that bats reduce echolocation output to avoid eavesdropping by conspecifics during the mating season. These findings might partly explain why tens of thousands of hoary bats are killed by wind turbines each year. They also challenge the long-standing assumption that bats-model organisms for sensory specialization-are reliant on sonar for nocturnal navigation. © 2018 The Author(s).

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.

Rye Canyon X-ray noise test: One-third octave-band data

NASA Technical Reports Server (NTRS)

Willshire, W. L., Jr.

1983-01-01

Acoustic data were obtained for the 25 ft. diameter X-wing rotor model during performance testing of the rotor system in hover. Data collected at the outdoor whirl tower test facility with a twelve microphone array were taken for approximately 150 test conditions comprised of various combinations of RPM, blade pressure ratio (BPR), and blade angle of attack (collective). The three test parameters had four values of RPM from 404 to 497, twelve values of BPR from 1.0 to 2.1, and six values of collective from 0.0 deg to 8.5 deg. Fifteen to twenty seconds of acoustic data were reduced to obtain an average 1/3 octave band spectrum for each microphone for each test condition. The complete, as measured, 1/3 octave band results for all the acoustic data are listed. Another part of the X-wing noise test was the acoustic calibration of the Rye Canyon whirl tower bowl. Corrections were computed which, when applied to as measured data, yield estimates of the free field X-wing noise. The free field estimates provide a more realistic measure of the rotor system noise levels. Trend analysis of the three test parameters on noise level were performed.

Acoustic Modifications of the Ames 40x80 Foot Wind Tunnel and Test Techniques for High-Speed Research Model Testing

NASA Technical Reports Server (NTRS)

Soderman, Paul T.; Olson, Larry (Technical Monitor)

1995-01-01

The NFAC 40- by 80- Foot Wind Tunnel at Ames is being refurbished with a new, deep acoustic lining in the test section which will make the facility nearly anechoic over a large frequency range. The modification history, key elements, and schedule will be discussed. Design features and expected performance gains will be described. Background noise reductions will be summarized. Improvements in aeroacoustic research techniques have been developed and used recently at NFAC on several wind tunnel tests of High Speed Research models. Research on quiet inflow microphones and struts will be described. The Acoustic Survey Apparatus in the 40x80 will be illustrated. A special intensity probe was tested for source localization. Multi-channel, high speed digital data acquisition is now used for acoustics. And most important, phased microphone arrays have been developed and tested which have proven to be very powerful for source identification and increased signal-to-noise ratio. Use of these tools for the HEAT model will be illustrated. In addition, an acoustically absorbent symmetry plane was built to satisfy the HEAT semispan aerodynamic and acoustic requirements. Acoustic performance of that symmetry plane will be shown.

Prey pursuit strategy of Japanese horseshoe bats during an in-flight target-selection task.

PubMed

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.

Sound vibration signal processing for detection and identification detonation (knock) to optimize performance Otto engine

NASA Astrophysics Data System (ADS)

Sujono, A.; Santoso, B.; Juwana, W. E.

2016-03-01

Problems of detonation (knock) on Otto engine (petrol engine) is completely unresolved problem until now, especially if want to improve the performance. This research did sound vibration signal processing engine with a microphone sensor, for the detection and identification of detonation. A microphone that can be mounted is not attached to the cylinder block, that's high temperature, so that its performance will be more stable, durable and inexpensive. However, the method of analysis is not very easy, because a lot of noise (interference). Therefore the use of new methods of pattern recognition, through filtration, and the regression function normalized envelope. The result is quite good, can achieve a success rate of about 95%.

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

PubMed

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

2009-08-01

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

Phased Array Noise Source Localization Measurements of an F404 Nozzle Plume at Both Full and Model Scale

NASA Technical Reports Server (NTRS)

Podboy, Gary G.; Bridges, James E.; Henderson, Brenda S.

2010-01-01

A 48-microphone planar phased array system was used to acquire jet noise source localization data on both a full-scale F404-GE-F400 engine and on a 1/4th scale model of a F400 series nozzle. The full-scale engine test data show the location of the dominant noise sources in the jet plume as a function of frequency for the engine in both baseline (no chevron) and chevron configurations. Data are presented for the engine operating both with and without afterburners. Based on lessons learned during this test, a set of recommendations are provided regarding how the phased array measurement system could be modified in order to obtain more useful acoustic source localization data on high-performance military engines in the future. The data obtained on the 1/4th scale F400 series nozzle provide useful insights regarding the full-scale engine jet noise source mechanisms, and document some of the differences associated with testing at model-scale versus fullscale.

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

Code of Federal Regulations, 2010 CFR

2010-10-01

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

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

Code of Federal Regulations, 2011 CFR

2011-10-01

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

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

NASA Technical Reports Server (NTRS)

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

1995-01-01

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

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

PubMed

Keidser, Gitte; Hartley, David; Carter, Lyndal

2008-12-01

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

Sub-Surface Windscreen for Outdoor Measurement of Infrasound

NASA Technical Reports Server (NTRS)

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

2014-01-01

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

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

NASA Astrophysics Data System (ADS)

Dehé, Alfons

2017-06-01

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

Experimental Study of Slat Noise from 30P30N Three-Element High-Lift Airfoil in JAXA Hard-Wall Low-Speed Wind Tunnel

NASA Technical Reports Server (NTRS)

Murayama, Mitsuhiro; Nakakita, Kazuyuki; Yamamoto, Kazuomi; Ura, Hiroki; Ito, Yasushi; Choudhari, Meelan M.

2014-01-01

Aeroacoustic measurements associated with noise radiation from the leading edge slat of the canonical, unswept 30P30N three-element high-lift airfoil configuration have been obtained in a 2 m x 2 m hard-wall wind tunnel at the Japan Aerospace Exploration Agency (JAXA). Performed as part of a collaborative effort on airframe noise between JAXA and the National Aeronautics and Space Administration (NASA), the model geometry and majority of instrumentation details are identical to a NASA model with the exception of a larger span. For an angle of attack up to 10 degrees, the mean surface Cp distributions agree well with free-air computational fluid dynamics predictions corresponding to a corrected angle of attack. After employing suitable acoustic treatment for the brackets and end-wall effects, an approximately 2D noise source map is obtained from microphone array measurements, thus supporting the feasibility of generating a measurement database that can be used for comparison with free-air numerical simulations. Both surface pressure spectra obtained via KuliteTM transducers and the acoustic spectra derived from microphone array measurements display a mixture of a broad band component and narrow-band peaks (NBPs), both of which are most intense at the lower angles of attack and become progressively weaker as the angle of attack is increased. The NBPs exhibit a substantially higher spanwise coherence in comparison to the broadband portion of the spectrum and, hence, confirm the trends observed in previous numerical simulations. Somewhat surprisingly, measurements show that the presence of trip dots between the stagnation point and slat cusp enhances the NBP levels rather than mitigating them as found in a previous experiment.

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

DTIC Science & Technology

2013-04-01

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

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

PubMed

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

2015-09-01

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

A Synchronized Sensor Array for Remote Monitoring of Avian and Bat Interactions with Offshore Renewable Energy Facilities

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

Suryan, Robert; Albertani, Roberto; Polagye, Brian

Wind energy production in the U.S. is projected to increase to 35% of our nation’s energy by 2050. This substantial increase in the U.S. is only a portion of the global wind industry growth, as many countries strive to reduce greenhouse gas emissions. A major environmental concern and potential market barrier for expansion of wind energy is bird and bat mortality from impacts with turbine blades, towers, and nacelles. Carcass surveys are the standard protocol for quantifying mortality at onshore sites. This method is imperfect, however, due to survey frequency at remote sites, removal of carcasses by scavengers between surveys,more » searcher efficiency, and other biases as well as delays of days to weeks or more in obtaining information on collision events. Furthermore, carcass surveys are not feasible at offshore wind energy sites. Near-real-time detection and quantification of interaction rates is possible at both onshore and offshore wind facilities using an onboard, integrated sensor package with data transmitted to central processing centers. We developed and experimentally tested an array of sensors that continuously monitors for interactions (including impacts) of birds and bats with wind turbines. The synchronized array includes three sensor nodes: (1) vibration (accelerometers and contact microphones), (2) optical (visual and infrared spectrum cameras), and (3) bioacoustics (acoustic and ultrasonic microphones). Accelerometers and contact acoustic microphones are placed at the root of each blade to detect impact vibrations and sound waves propagating through the structure. On-board data processing algorithms using wavelet analysis detect impact signals exceeding background vibration. Stereo-visual and infrared cameras were placed on the nacelle to allow target tracking, distance, and size calculations. On-board image processing and target detection algorithms identify moving targets within the camera field of view. Bioacoustic recorders monitor vocalizations and echolocations to aid in identifying organisms involved in interactions. Data from all sensors are temporarily stored in ring (i.e., circular) buffers with a duration varying by sensor type. Detection of target presence or impact by any of the sensors can trigger the archiving of data from all buffers for transmission to a central data processing center for evaluation and post-processing. This mitigates the risk of “data mortgages” posed by continual recording and minimizes personnel time required to manually review event data. We first conducted individual component tests at laboratories and field sites in Corvallis and Newport, Oregon, and Seattle and Sequim, Washington. We conducted additional component tests on research wind turbines at the North American Wind Research and Training Center, Mesalands Community College (MCC; General Electric 1.5 MW turbine), New Mexico, and the National Wind Technology Center, National Renewable Energy Laboratory (NREL; Controls Advanced Research Turbines 3 [CART 3] 600 kW Westinghouse turbine), Colorado. We conducted fully integrated system tests at NREL in October 2014 and April 2015. We used only research wind turbines so that we could conduct controlled, experimentally generated impacts using empty and water-filled tennis balls shot from a compressed air cannon on the ground. The approx. 57-140 g tennis balls (depending on water content) were at the upper mass range for bats, but lower mass range for marine birds. Therefore, the ability to detect collisions of most seabirds is likely greater than our experiments demonstrate, but possibly lower for some bats depending on the background signal of a given turbine. Vibration data demonstrated that background signals of operating turbines varied markedly among the CART 3 under normal operation (greatest), GE (moderate), and CART 3 during idle rotation (generator not engaged; least). In total, we measured 63 experimental blade impacts on the two research turbines. Impaction detection was dependent on background signals, position of impact on the blade (a tip strike resulted in the strongest impact signal), and impact kinetics (velocity of ball and whether the ball struck the surface of the blade or the leading edge of the blade struck the ball). Overall detection percentage ranged from 100% for the “quietest” conditions (CART 3 idle rotation), down to 35% for the noisiest (CART 3 normal operation). Impact signals were detected from sensors on more than one blade (i.e., blades other than the blade struck) 50% - 75% of the time. Stereo imaging provided valuable metrics, but increased data processing and equipment cost. Given the cost of cameras with sufficient resolution for target identification, we suggest mounting cameras directly on the blades to continuously view the entire rotor swept area with the fewest number of cameras. Bioacoustic microphones provide taxonomic identification, as well as information on ambient noise levels. They also assist in identifying environmental conditions such as hail storms, high winds, thunder, lightning, etc., that may contribute to a collision or a false positive detection. We demonstrated a proof of concept for an integrated sensor array to detect and identify bird and bat collisions with wind turbines. The next phase of research and development for this system will miniaturize and integrate sensors from all three nodes into a single wireless package that can be attached directly to the blade. This next generation system would use all “smart” sensors capable of onboard data processing to drastically reduce data streams and processing time on a central computer. A provisional patent for the blade mounted system was submitted by Oregon State University and recorded by the U.S. Patent and Trademark Office (application no. 62313028). Eventually, technology and industry advances will allow this low cost monitoring system to be designed into materials during manufacturing so that all turbines could be monitored with either a subset or full suite of sensors. As standard equipment on all commercial turbines, the sensor suite would allow the industry to effectively monitor whether individual turbines were causing mortalities or not and under what circumstances. It would also provide real-time evaluation of mechanical and structural integrity of a turbine via vibration, image, and acoustic data streams, thereby permitting modifications in operation to limit environmental or mechanical damage.« less

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

ERIC Educational Resources Information Center

American School Band Directors Association, Newark, OH.

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

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

PubMed

Chung, King; Mongeau, Luc; McKibben, Nicholas

2009-04-01

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

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

PubMed

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

2009-10-01

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

West Texas array experiment: Noise and source characterization of short-range infrasound and acoustic signals, along with lab and field evaluation of Intermountain Laboratories infrasound microphones

NASA Astrophysics Data System (ADS)

Fisher, Aileen

The term infrasound describes atmospheric sound waves with frequencies below 20 Hz, while acoustics are classified within the audible range of 20 Hz to 20 kHz. Infrasound and acoustic monitoring in the scientific community is hampered by low signal-to-noise ratios and a limited number of studies on regional and short-range noise and source characterization. The JASON Report (2005) suggests the infrasound community focus on more broad-frequency, observational studies within a tactical distance of 10 km. In keeping with that recommendation, this paper presents a study of regional and short-range atmospheric acoustic and infrasonic noise characterization, at a desert site in West Texas, covering a broad frequency range of 0.2 to 100 Hz. To spatially sample the band, a large number of infrasound gauges was needed. A laboratory instrument analysis is presented of the set of low-cost infrasound sensors used in this study, manufactured by Inter-Mountain Laboratories (IML). Analysis includes spectra, transfer functions and coherences to assess the stability and range of the gauges, and complements additional instrument testing by Sandia National Laboratories. The IMLs documented here have been found reliably coherent from 0.1 to 7 Hz without instrument correction. Corrections were built using corresponding time series from the commercially available and more expensive Chaparral infrasound gauge, so that the corrected IML outputs were able to closely mimic the Chaparral output. Arrays of gauges are needed for atmospheric sound signal processing. Our West Texas experiment consisted of a 1.5 km aperture, 23-gauge infrasound/acoustic array of IMLs, with a compact, 12 m diameter grid-array of rented IMLs at the center. To optimize signal recording, signal-to-noise ratio needs to be quantified with respect to both frequency band and coherence length. The higher-frequency grid array consisted of 25 microphones arranged in a five by five pattern with 3 meter spacing, without spatial wind noise filtering hoses or pipes. The grid was within the distance limits of a single gauge's normal hose array, and data were used to perform a spatial noise correlation study. The highest correlation values were not found in the lower frequencies as anticipated, owing to a lack of sources in the lower range and the uncorrelated nature of wind noise. The highest values, with cross-correlation averages between 0.4 and 0.7 from 3 to 17 m between gauges, were found at night from 10 and 20 Hz due to a continuous local noise source and low wind. Data from the larger array were used to identify continuous and impulsive signals in the area that comprise the ambient noise field. Ground truth infrasound and acoustic, time and location data were taken for a highway site, a wind farm, and a natural gas compressor. Close-range sound data were taken with a single IML "traveler" gauge. Spectrograms and spectrum peaks were used to identify their source signatures. Two regional location techniques were also tested with data from the large array by using a propane cannon as a controlled, impulsive source. A comparison is presented of the Multiple Signal Classification Algorithm (MUSIC) to a simple, quadratic, circular wavefront algorithm. MUSIC was unable to effectively separate noise and source eignenvalues and eigenvectors due to spatial aliasing of the propane cannon signal and a lack of incoherent noise. Only 33 out of 80 usable shots were located by MUSIC within 100 m. Future work with the algorithm should focus on location of impulsive and continuous signals with development of methods for accurate separation of signal and noise eigenvectors in the presence of coherent noise and possible spatial aliasing. The circular wavefront algorithm performed better with our specific dataset and successfully located 70 out of 80 propane cannon shots within 100 m of the original location, 66 of which were within 20 m. This method has low computation requirements, making it well suited for real-time automated processing and smaller computers. Future research could focus on development of the method for an automated system and statistical impulsive noise filtering for higher accuracy.

Sound Source Localization Using Non-Conformal Surface Sound Field Transformation Based on Spherical Harmonic Wave Decomposition

PubMed Central

Zhang, Lanyue; Ding, Dandan; Yang, Desen; Wang, Jia; Shi, Jie

2017-01-01

Spherical microphone arrays have been paid increasing attention for their ability to locate a sound source with arbitrary incident angle in three-dimensional space. Low-frequency sound sources are usually located by using spherical near-field acoustic holography. The reconstruction surface and holography surface are conformal surfaces in the conventional sound field transformation based on generalized Fourier transform. When the sound source is on the cylindrical surface, it is difficult to locate by using spherical surface conformal transform. The non-conformal sound field transformation by making a transfer matrix based on spherical harmonic wave decomposition is proposed in this paper, which can achieve the transformation of a spherical surface into a cylindrical surface by using spherical array data. The theoretical expressions of the proposed method are deduced, and the performance of the method is simulated. Moreover, the experiment of sound source localization by using a spherical array with randomly and uniformly distributed elements is carried out. Results show that the non-conformal surface sound field transformation from a spherical surface to a cylindrical surface is realized by using the proposed method. The localization deviation is around 0.01 m, and the resolution is around 0.3 m. The application of the spherical array is extended, and the localization ability of the spherical array is improved. PMID:28489065

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

PubMed

Kisenwether, Jessica Sofranko; Sataloff, Robert T

2015-09-01

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

ZnO thin film piezoelectric micromachined microphone with symmetric composite vibrating diaphragm

NASA Astrophysics Data System (ADS)

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

2017-05-01

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

Beampattern control of a microphone array to minimize secondary source contamination.

PubMed

Jordan, Peter; Fitzpatrick, John A; Meskell, Craig

2003-10-01

A null-steering technique is adapted and applied to a linear delay-and-sum beamformer in order to measure the noise generated by one of the propellers of a 1/8 scale twin propeller aircraft model. The technique involves shading the linear array using a set of weights, which are calculated according to the locations onto which the nulls need to be steered (in this case onto the second propeller). The technique is based on an established microwave antenna theory, and uses a plane-wave, or far field formulation in order to represent the response of the array by an nth-order polynomial, where n is the number of array elements. The roots of this polynomial correspond to the minima of the array response, and so by an appropriate choice of roots, a polynomial can be generated, the coefficients of which are the weights needed to achieve the prespecified set of null positions. It is shown that, for the technique to work with actual data, the cross-spectral matrix must be conditioned before array shading is implemented. This ensures that the shading function is not distorted by the intrinsic element weighting which can occur as a result of the directional nature of aeroacoustic systems. A difference of 6 dB between measurements before and after null steering shows the technique to have been effective in eliminating the contribution from one of the propellers, thus providing a quantitative measure of the acoustic energy from the other.

SCAMP: Rapid Focused Sonic Boom Waypoint Flight Planning Methods, Execution, and Results

NASA Technical Reports Server (NTRS)

Haering, Edward A., Jr.; Cliatt, Larry J., II; Delaney, Michael M., Jr.; Plotkin, Kenneth J.; Maglieri, Domenic J.; Brown, Jacob C.

2012-01-01

Successful execution of the flight phase of the Superboom Caustic Analysis and Measurement Project (SCAMP) required accurate placement of focused sonic booms on an array of prepositioned ground sensors. While the array was spread over a 10,000-ft-long area, this is a relatively small region when considering the speed of a supersonic aircraft and sonic boom ray path variability due to shifting atmospheric conditions and aircraft trajectories. Another requirement of the project was to determine the proper position for a microphone-equipped motorized glider to intercept the sonic boom caustic, adding critical timing to the constraints. Variability in several inputs to these calculations caused some shifts of the focus away from the optimal location. Reports of the sonic booms heard by persons positioned amongst the array were used to shift the focus closer to the optimal location for subsequent passes. This paper describes the methods and computations used to place the focused sonic boom on the SCAMP array and gives recommendations for their accurate placement by future quiet supersonic aircraft. For the SCAMP flights, 67% of the foci were placed on the ground array with measured positions within a few thousand feet of computed positions. Among those foci with large caustic elevation angles, 96% of foci were placed on the array, and measured positions were within a few hundred feet of computed positions. The motorized glider captured sonic booms on 59% of the passes when the instrumentation was operating properly.

FPGA implementation of adaptive beamforming in hearing aids.

PubMed

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.

Point source moving above a finite impedance reflecting plane - Experiment and theory

NASA Technical Reports Server (NTRS)

Norum, T. D.; Liu, C. H.

1978-01-01

A widely used experimental version of the acoustic monopole consists of an acoustic driver of restricted opening forced by a discrete frequency oscillator. To investigate the effects of forward motion on this source, it was mounted above an automobile and driven over an asphalt surface at constant speed past a microphone array. The shapes of the received signal were compared to results computed from an analysis of a fluctuating-mass-type point source moving above a finite impedance reflecting plane. Good agreement was found between experiment and theory when a complex normal impedance representative of a fairly hard acoustic surface was used in the analysis.

KSC-2009-2103

NASA Image and Video Library

2009-03-15

CAPE CANAVERAL, Fla. – In Firing Room 4 of the Launch Control Center at NASA's Kennedy Space Center in Florida, Center Director Bob Cabana (with microphone) congratulates the mission management team after the successful launch of space shuttle Discovery on the STS-119 mission. Launch was on time at 7:43 p.m. EDT. The STS-119 mission is the 28th to the space station and Discovery's 36th flight. Discovery will deliver the final pair of power-generating solar array wings and the S6 truss segment. Installation of S6 will signal the station's readiness to house a six-member crew for conducting increased science. Photo credit: NASA/Kim Shiflett

Static noise tests on modified augmentor wing jet STOL research aircraft

NASA Technical Reports Server (NTRS)

Cook, G. R.; Lilley, B. F.

1981-01-01

Noise measurements were made to determine if recent modifications made to the bifurcated jetpipe to increase engine thrust had at the same time reduced the noise level. The noise field was measured by a 6-microphone array positioned on a 30.5m (100 ft) sideline between 90 and 150 degrees from the left engine inlet. Noise levels were recorded at three flap angles over a range of engine thrust settings from flight idle to emergency power and plotted in one-third octave band spectra. Little attenuation was observed at maximum power, but significant attenuation was achieved at approach and cruise power levels.

Acoustic Network Localization and Interpretation of Infrasonic Pulses from Lightning

NASA Astrophysics Data System (ADS)

Arechiga, R. O.; Johnson, J. B.; Badillo, E.; Michnovicz, J. C.; Thomas, R. J.; Edens, H. E.; Rison, W.

2011-12-01

We improve on the localization accuracy of thunder sources and identify infrasonic pulses that are correlated across a network of acoustic arrays. We attribute these pulses to electrostatic charge relaxation (collapse of the electric field) and attempt to model their spatial extent and acoustic source strength. Toward this objective we have developed a single audio range (20-15,000 Hz) acoustic array and a 4-station network of broadband (0.01-500 Hz) microphone arrays with aperture of ~45 m. The network has an aperture of 1700 m and was installed during the summers of 2009-2011 in the Magdalena mountains of New Mexico, an area that is subject to frequent lightning activity. We are exploring a new technique based on inverse theory that integrates information from the audio range and the network of broadband acoustic arrays to locate thunder sources more accurately than can be achieved with a single array. We evaluate the performance of the technique by comparing the location of thunder sources with RF sources located by the lightning mapping array (LMA) of Langmuir Laboratory at New Mexico Tech. We will show results of this technique for lightning flashes that occurred in the vicinity of our network of acoustic arrays and over the LMA. We will use acoustic network detection of infrasonic pulses together with LMA data and electric field measurements to estimate the spatial distribution of the charge (within the cloud) that is used to produce a lightning flash, and will try to quantify volumetric charges (charge magnitude) within clouds.

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

PubMed

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

2016-07-01

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

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

NASA Astrophysics Data System (ADS)

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

2017-10-01

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

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

PubMed Central

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

2017-01-01

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

MEMS capacitive accelerometer-based middle ear microphone.

PubMed

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

2012-12-01

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

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

PubMed

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

2017-05-16

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

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

PubMed

Chung, King; Nelson, Lance; Teske, Melissa

2012-09-01

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

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

PubMed

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

2018-03-01

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

Cochlear microphonic broad tuning curves

NASA Astrophysics Data System (ADS)

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

2015-12-01

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

High-temperature fiber-optic lever microphone

NASA Technical Reports Server (NTRS)

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

1995-01-01

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

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

PubMed

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

2017-12-01

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

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

PubMed

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

2003-06-01

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

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

PubMed

Chung, King; McKibben, Nicholas

2011-10-01

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

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

PubMed

Ricketts, Todd A; Picou, Erin M

2013-09-01

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

The reduction of gunshot noise and auditory risk through the use of firearm suppressors and low-velocity ammunition.

PubMed

Murphy, William J; Flamme, Gregory A; Campbell, Adam R; Zechmann, Edward L; Tasko, Stephen M; Lankford, James E; Meinke, Deanna K; Finan, Donald S; Stewart, Michael

2018-02-01

This research assessed the reduction of peak levels, equivalent energy and sound power of firearm suppressors. The first study evaluated the effect of three suppressors at four microphone positions around four firearms. The second study assessed the suppressor-related reduction of sound power with a 3 m hemispherical microphone array for two firearms. The suppressors reduced exposures at the ear between 17 and 24 dB peak sound pressure level and reduced the 8 h equivalent A-weighted energy between 9 and 21 dB depending upon the firearm and ammunition. Noise reductions observed for the instructor's position about a metre behind the shooter were between 20 and 28 dB peak sound pressure level and between 11 and 26 dB L Aeq,8h . Firearm suppressors reduced the measured sound power levels between 2 and 23 dB. Sound power reductions were greater for the low-velocity ammunition than for the same firearms fired with high-velocity ammunition due to the effect of N-waves produced by a supersonic bullet. Firearm suppressors may reduce noise exposure, and the cumulative exposures of suppressed firearms can still present a significant hearing risk. Therefore, firearm users should always wear hearing protection whenever target shooting or hunting.

Measured noise of a scale model high speed propeller at simulated takeoff/approach conditions

NASA Technical Reports Server (NTRS)

Woodward, Richard P.

1987-01-01

A model high-speed advanced propeller, SR-7A, was tested in the NASA Lewis 9x15 foot anechoic wind tunnel at simulated takeoff/approach conditions of 0.2 Mach number. These tests were in support of the full-scale Propfan Text Assessment (PTA) flight program. Acoustic measurements were taken with fixed microphone arrays and with an axially translating microphone probe. Limited aerodynamic measurements were also taken to establish the propeller operating conditions. Tests were conducted with the propeller alone and with three down-stream wing configurations. The propeller was run over a range of blade setting angles from 32.0 deg. to 43.6 deg., tip speeds from 183 to 290 m/sec (600 to 950 ft/sec), and angles of attack from -10 deg. to +15 deg. The propeller alone BPF tone noise was found to increase 10 dB in the flyover plane at 15 deg. propeller axis angle of attack. The installation of the straight wing at minimum spacing of 0.54 wing chord increased the tone noise 5 dB under the wing of 10 deg. propeller axis angle of attack, while a similarly spaced inboard upswept wing only increased the tone noise 2 dB.

Trans Atlantic Infrasound Payload (TAIP) Operation Plan.

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

Bowman, Daniel; Lees, Jonathan M.

The Carolina Infrasound package, added as a piggyback to the 2016 ULDB ight, recorded unique acoustic signals such as the ocean microbarom and a large meteor. These data both yielded unique insights into the acoustic energy transfer from the lower to the upper atmosphere as well as highlighted the vast array of signals whose origins remain unknown. Now, the opportunity to y a payload across the north Atlantic offers an opportunity to sample one of the most active ocean microbarom sources on Earth. Improvements in payload capabilities should result in characterization of the higher frequency range of the stratospheric infrasoundmore » spectrum as well. Finally, numerous large mining and munitions disposal explosions in the region may provide \\ground truth" events for assessing the detection capability of infrasound microphones in the stratosphere. The flight will include three different types of infrasound sensors. One type is a pair of polarity reversed InfraBSU microphones (standard for high altitude flights since 2016), another is a highly sensitive Chaparral 60 modified for a very low corner period, and the final sensor is a lightweight, low power Gem infrasound package. By evaluating these configurations against each other on the same flight, we will be able to optimize future campaigns with different sensitivity and mass constraints.« less

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.

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

PubMed

Moatti, L; Busquet, D; Cotin, G

1983-01-01

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

A low-cost acoustic permeameter

NASA Astrophysics Data System (ADS)

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

2017-04-01

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

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

PubMed

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

2009-10-01

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

Wideband RELAX and wideband CLEAN for aeroacoustic imaging

NASA Astrophysics Data System (ADS)

Wang, Yanwei; Li, Jian; Stoica, Petre; Sheplak, Mark; Nishida, Toshikazu

2004-02-01

Microphone arrays can be used for acoustic source localization and characterization in wind tunnel testing. In this paper, the wideband RELAX (WB-RELAX) and the wideband CLEAN (WB-CLEAN) algorithms are presented for aeroacoustic imaging using an acoustic array. WB-RELAX is a parametric approach that can be used efficiently for point source imaging without the sidelobe problems suffered by the delay-and-sum beamforming approaches. WB-CLEAN does not have sidelobe problems either, but it behaves more like a nonparametric approach and can be used for both point source and distributed source imaging. Moreover, neither of the algorithms suffers from the severe performance degradations encountered by the adaptive beamforming methods when the number of snapshots is small and/or the sources are highly correlated or coherent with each other. A two-step optimization procedure is used to implement the WB-RELAX and WB-CLEAN algorithms efficiently. The performance of WB-RELAX and WB-CLEAN is demonstrated by applying them to measured data obtained at the NASA Langley Quiet Flow Facility using a small aperture directional array (SADA). Somewhat surprisingly, using these approaches, not only were the parameters of the dominant source accurately determined, but a highly correlated multipath of the dominant source was also discovered.

Spatial location influences vocal interactions in bullfrog choruses

PubMed Central

Bates, Mary E.; Cropp, Brett F.; Gonchar, Marina; Knowles, Jeffrey; Simmons, James A.; Simmons, Andrea Megela

2010-01-01

A multiple sensor array was employed to identify the spatial locations of all vocalizing male bullfrogs (Rana catesbeiana) in five natural choruses. Patterns of vocal activity collected with this array were compared with computer simulations of chorus activity. Bullfrogs were not randomly spaced within choruses, but tended to cluster into closely spaced groups of two to five vocalizing males. There were nonrandom, differing patterns of vocal interactions within clusters of closely spaced males and between different clusters. Bullfrogs located within the same cluster tended to overlap or alternate call notes with two or more other males in that cluster. These near-simultaneous calling bouts produced advertisement calls with more pronounced amplitude modulation than occurred in nonoverlapping notes or calls. Bullfrogs located in different clusters more often alternated entire calls or overlapped only small segments of their calls. They also tended to respond sequentially to calls of their farther neighbors compared to their nearer neighbors. Results of computational analyses showed that the observed patterns of vocal interactions were significantly different than expected based on random activity. The use of a multiple sensor array provides a richer view of the dynamics of choruses than available based on single microphone techniques. PMID:20370047

Wideband RELAX and wideband CLEAN for aeroacoustic imaging.

PubMed

Wang, Yanwei; Li, Jian; Stoica, Petre; Sheplak, Mark; Nishida, Toshikazu

2004-02-01

Microphone arrays can be used for acoustic source localization and characterization in wind tunnel testing. In this paper, the wideband RELAX (WB-RELAX) and the wideband CLEAN (WB-CLEAN) algorithms are presented for aeroacoustic imaging using an acoustic array. WB-RELAX is a parametric approach that can be used efficiently for point source imaging without the sidelobe problems suffered by the delay-and-sum beamforming approaches. WB-CLEAN does not have sidelobe problems either, but it behaves more like a nonparametric approach and can be used for both point source and distributed source imaging. Moreover, neither of the algorithms suffers from the severe performance degradations encountered by the adaptive beamforming methods when the number of snapshots is small and/or the sources are highly correlated or coherent with each other. A two-step optimization procedure is used to implement the WB-RELAX and WB-CLEAN algorithms efficiently. The performance of WB-RELAX and WB-CLEAN is demonstrated by applying them to measured data obtained at the NASA Langley Quiet Flow Facility using a small aperture directional array (SADA). Somewhat surprisingly, using these approaches, not only were the parameters of the dominant source accurately determined, but a highly correlated multipath of the dominant source was also discovered.

Microphone Phenomena Observed with EFL Students.

ERIC Educational Resources Information Center

Wilcox, Wilma B.

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

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

DTIC Science & Technology

2010-08-01

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

A high-temperature wideband pressure transducer

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J.

1975-01-01

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

Noise Robust Speech Recognition Applied to Voice-Driven Wheelchair

NASA Astrophysics Data System (ADS)

Sasou, Akira; Kojima, Hiroaki

2009-12-01

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

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

PubMed

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

2015-06-01

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

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

PubMed

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

2013-10-15

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

Aerodynamic Measurement Technology

NASA Technical Reports Server (NTRS)

Burner, Alpheus W.

2002-01-01

Ohio State University developed a new spectrally filtered light-scattering apparatus based on a diode laser injected-locked titanium: sapphire laser and rubidium vapor filter at 780.2 nm. When the device was combined with a stimulated Brillouin scattering phase conjugate mirror, the realizable peak attenuation of elastic scattering interferences exceeded 105. The potential of the system was demonstrated by performing Thomson scattering measurements. Under USAF-NASA funding, West Virginia University developed a Doppler global velocimetry system using inexpensive 8-bit charged coupled device cameras and digitizers and a CW argon ion laser. It has demonstrated a precision of +/- 2.5 m/sec in a swirling jet flow. Low-noise silicon-micromachined microphones developed and incorporated in a novel two-tier, hybrid packaging scheme at the University of Florida used printed circuit board technology to realize a MEMS-based directional acoustic array. The array demonstrated excellent performance relative to conventional sensor technologies and provides scaling technologies that can reduce cost and increase speed and mobility.

First images of thunder: Acoustic imaging of triggered lightning

NASA Astrophysics Data System (ADS)

Dayeh, M. A.; Evans, N. D.; Fuselier, S. A.; Trevino, J.; Ramaekers, J.; Dwyer, J. R.; Lucia, R.; Rassoul, H. K.; Kotovsky, D. A.; Jordan, D. M.; Uman, M. A.

2015-07-01

An acoustic camera comprising a linear microphone array is used to image the thunder signature of triggered lightning. Measurements were taken at the International Center for Lightning Research and Testing in Camp Blanding, FL, during the summer of 2014. The array was positioned in an end-fire orientation thus enabling the peak acoustic reception pattern to be steered vertically with a frequency-dependent spatial resolution. On 14 July 2014, a lightning event with nine return strokes was successfully triggered. We present the first acoustic images of individual return strokes at high frequencies (>1 kHz) and compare the acoustically inferred profile with optical images. We find (i) a strong correlation between the return stroke peak current and the radiated acoustic pressure and (ii) an acoustic signature from an M component current pulse with an unusual fast rise time. These results show that acoustic imaging enables clear identification and quantification of thunder sources as a function of lightning channel altitude.

Vector intensity reconstruction using the data completion method.

PubMed

Langrenne, Christophe; Garcia, Alexandre

2013-04-01

This paper presents an application of the data completion method (DCM) for vector intensity reconstructions. A mobile array of 36 pressure-pressure probes (72 microphones) is used to perform measurements near a planar surface. Nevertheless, since the proposed method is based on integral formulations, DCM can be applied with any kind of geometry. This method requires the knowledge of Cauchy data (pressure and velocity) on a part of the boundary of an empty domain in order to evaluate pressure and velocity on the remaining part of the boundary. Intensity vectors are calculated in the interior domain surrounded by the measurement array. This inverse acoustic problem requires the use of a regularization method to obtain a realistic solution. An experiment in a closed wooden car trunk mock-up excited by a shaker and two loudspeakers is presented. In this case, where the volume of the mock-up is small (0.61 m(3)), standing-waves and fluid structure interactions appear and show that DCM is a powerful tool to identify sources in a confined space.

STS-97 crew meets with the media at Launch Pad 39B

NASA Technical Reports Server (NTRS)

2000-01-01

From the slidewire landing zone at Launch Pad 39B, STS-97 Mission Specialist Joe Tanner (center, with microphone) speaks to the press about his extravehicular activity (EVA) during the mission. With him are the rest of the crew, Commander Brent Jett and Pilot Mike Bloomfield on the left and Mission Specialists Marc Garneau and Carlos Noriega on the right. The crew is at KSC to take part in Terminal Countdown Demonstration Test activities that include emergency egress training, familiarization with the payload, and a simulated launch countdown. Visible in the background are the solid rocket booster and external tank on Space Shuttle Endeavour. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:05 p.m. EST.

The STS-97 crew meets with the media at Launch Pad 39B

NASA Technical Reports Server (NTRS)

2000-01-01

From the slidewire landing zone at Launch Pad 39B, STS-97 Mission Specialist Joe Tanner (center, with microphone) speaks to the press about his extravehicular activity (EVA) during the mission. With him are the rest of the crew, Commander Brent Jett and Pilot Mike Bloomfield on the left and Mission Specialists Marc Garneau and Carlos Noriega on the right. The crew is at KSC to take part in Terminal Countdown Demonstration Test activities that include emergency egress training, familiarization with the payload, and a simulated launch countdown. Visible in the background are the solid rocket booster and external tank on Space Shuttle Endeavour. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:05 p.m. EST.

Digital Signal Processing in Acoustics--Part 2.

ERIC Educational Resources Information Center

Davies, H.; McNeill, D. J.

1986-01-01

Reviews the potential of a data acquisition system for illustrating the nature and significance of ideas in digital signal processing. Focuses on the fast Fourier transform and the utility of its two-channel format, emphasizing cross-correlation and its two-microphone technique of acoustic intensity measurement. Includes programing format. (ML)

The Benefit of Remote Microphones Using Four Wireless Protocols.

PubMed

Rodemerk, Krishna S; Galster, Jason A

2015-09-01

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

SonicBAT Testing

NASA Image and Video Library

2017-08-24

Teams from NASA's Armstrong Flight Research Center in California, and Langley Research Center in Virginia, are conducting supersonic flight tests to study the ways sonic booms travel. The Sonic Booms in Atmospheric Turbulence flight series, or SonicBAT, features a F/A-18 research aircraft to create sonic booms, flying at supersonic speeds just off the coast of Florida. In order to understand how atmospheric turbulence in a humic climate impacts how sonic booms travel, NASA is flying a TG-14 motorized glider to obtain data on sonic booms before they travel through atmospheric turbulence. That data is compared with similar data captured by two microphone arrays on the ground that hear sonic booms that have traveled through atmospheric turbulence.

49 CFR 325.33 - Site characteristics; highway operations.

Code of Federal Regulations, 2010 CFR

2010-10-01

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

Directional Microphone Hearing Aids in School Environments: Working toward Optimization

ERIC Educational Resources Information Center

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

2017-01-01

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

A high-temperature wideband pressure transducer

NASA Technical Reports Server (NTRS)

Zuckerwar, A. J.

1976-01-01

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

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.

A sophisticated, multi-channel data acquisition and processing system for high frequency noise research

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.

Improving the accuracy of smart devices to measure noise exposure.

PubMed

Roberts, Benjamin; Kardous, Chucri; Neitzel, Richard

2016-11-01

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

Background Noise Reduction Using Adaptive Noise Cancellation Determined by the Cross-Correlation

NASA Technical Reports Server (NTRS)

Spalt, Taylor B.; Brooks, Thomas F.; Fuller, Christopher R.

2012-01-01

Background noise due to flow in wind tunnels contaminates desired data by decreasing the Signal-to-Noise Ratio. The use of Adaptive Noise Cancellation to remove background noise at measurement microphones is compromised when the reference sensor measures both background and desired noise. The technique proposed modifies the classical processing configuration based on the cross-correlation between the reference and primary microphone. Background noise attenuation is achieved using a cross-correlation sample width that encompasses only the background noise and a matched delay for the adaptive processing. A present limitation of the method is that a minimum time delay between the background noise and desired signal must exist in order for the correlated parts of the desired signal to be separated from the background noise in the crosscorrelation. A simulation yields primary signal recovery which can be predicted from the coherence of the background noise between the channels. Results are compared with two existing methods.

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

NASA Astrophysics Data System (ADS)

Bicen, Baris

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

Limitations of Phased Array Beamforming in Open Rotor Noise Source Imaging

NASA Technical Reports Server (NTRS)

Horvath, Csaba; Envia, Edmane; Podboy, Gary G.

2013-01-01

Phased array beamforming results of the F31/A31 historical baseline counter-rotating open rotor blade set were investigated for measurement data taken on the NASA Counter-Rotating Open Rotor Propulsion Rig in the 9- by 15-Foot Low-Speed Wind Tunnel of NASA Glenn Research Center as well as data produced using the LINPROP open rotor tone noise code. The planar microphone array was positioned broadside and parallel to the axis of the open rotor, roughly 2.3 rotor diameters away. The results provide insight as to why the apparent noise sources of the blade passing frequency tones and interaction tones appear at their nominal Mach radii instead of at the actual noise sources, even if those locations are not on the blades. Contour maps corresponding to the sound fields produced by the radiating sound waves, taken from the simulations, are used to illustrate how the interaction patterns of circumferential spinning modes of rotating coherent noise sources interact with the phased array, often giving misleading results, as the apparent sources do not always show where the actual noise sources are located. This suggests that a more sophisticated source model would be required to accurately locate the sources of each tone. The results of this study also have implications with regard to the shielding of open rotor sources by airframe empennages.

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

PubMed

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

2016-01-01

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

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

Federal Register 2010, 2011, 2012, 2013, 2014

2010-05-07

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

Dereverberation and denoising based on generalized spectral subtraction by multi-channel LMS algorithm using a small-scale microphone array

NASA Astrophysics Data System (ADS)

Wang, Longbiao; Odani, Kyohei; Kai, Atsuhiko

2012-12-01

A blind dereverberation method based on power spectral subtraction (SS) using a multi-channel least mean squares algorithm was previously proposed to suppress the reverberant speech without additive noise. The results of isolated word speech recognition experiments showed that this method achieved significant improvements over conventional cepstral mean normalization (CMN) in a reverberant environment. In this paper, we propose a blind dereverberation method based on generalized spectral subtraction (GSS), which has been shown to be effective for noise reduction, instead of power SS. Furthermore, we extend the missing feature theory (MFT), which was initially proposed to enhance the robustness of additive noise, to dereverberation. A one-stage dereverberation and denoising method based on GSS is presented to simultaneously suppress both the additive noise and nonstationary multiplicative noise (reverberation). The proposed dereverberation method based on GSS with MFT is evaluated on a large vocabulary continuous speech recognition task. When the additive noise was absent, the dereverberation method based on GSS with MFT using only 2 microphones achieves a relative word error reduction rate of 11.4 and 32.6% compared to the dereverberation method based on power SS and the conventional CMN, respectively. For the reverberant and noisy speech, the dereverberation and denoising method based on GSS achieves a relative word error reduction rate of 12.8% compared to the conventional CMN with GSS-based additive noise reduction method. We also analyze the effective factors of the compensation parameter estimation for the dereverberation method based on SS, such as the number of channels (the number of microphones), the length of reverberation to be suppressed, and the length of the utterance used for parameter estimation. The experimental results showed that the SS-based method is robust in a variety of reverberant environments for both isolated and continuous speech recognition and under various parameter estimation conditions.

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

PubMed

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

2016-08-01

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

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

PubMed

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

2015-11-01

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

System for determining aerodynamic imbalance

NASA Technical Reports Server (NTRS)

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

1994-01-01

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

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.

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

Code of Federal Regulations, 2010 CFR

2010-10-01

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

Carbon granule probe microphone for leak detection. [recovery boilers

NASA Technical Reports Server (NTRS)

Parthasarathy, S. P. (Inventor)

1985-01-01

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

Sound-field measurement with moving microphones

PubMed Central

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

2017-01-01

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

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

PubMed

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

2013-06-01

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

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

NASA Astrophysics Data System (ADS)

Manz, Johannes; Dehe, Alfons; Schrag, Gabriele

2017-05-01

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

Photoacoustic and Photothermal Effects in Particulate Suspensions

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

Diebold, Gerald, J.

A summary of the research areas investigated by the author during the grant period is given. Experiments and theory have been carried out on the photoacoustic effect arising from a number of physical and chemical processes. A number of studies of the photoacoustic effect as it occurs in transient grating experiments have been completed. The research done with the Ludwig-Soret effect on the generation of shock waves is reported. Other research, such as that carried out on interferometric and beam deflection microphones, the use of microphones in vacuum as momentum flux detectors, and chemical generation of sonoluminescence is listed. Amore » list of published research including selected publications, a complete list of journal articles, books, review articles, and reviews are given.« less

Detection of the large meteoroid/NEO flux using infrasound: recent detection of the November 21, 1995, Colorado fireball

NASA Astrophysics Data System (ADS)

Revelle, Douglas O.; Whitaker, Rodney W.

1996-10-01

During the early morning of November 21, 1995, a fireball as bright as the full moon entered the atmosphere over southeastern Colorado and initially produced audible sonic boom reports from Texas to Wyoming. The event was detected locally by a security video camera which showed the reflection of the fireball event on the hood of a truck. The camera also recorded tree shadows cast by the light of the fireball. This recording includes the audio signal of a strong double boom as well. Subsequent investigation of the array near Los Alamos, New Mexico operated by the Los Alamos National Laboratory as part of its commitment to the Comprehensive Test Ban Treaty negotiations, showed the presence of an infrasonic signal from the proper direction at about the correct time for this fireball. The Los Alamos array is a four-element infrasonic system in near-continuous operation on the laboratory property. The nominal spacing between the array elements is 212 m. The basic sensor is a Globe Universal Sciences Model 100C microphone whose response is flat from about 0.1 to 300 Hz (which we filter at the high frequency end to be limited to 20 Hz). Each low frequency microphone is connected to a set of twelve porous hoses to reduce wind noise. The characteristics of the observed signal include the onset arrival time of 0939:20 UT (0239:20 MST), with a maximum timing uncertainty of plus or minus 2 minutes, the signal onset time delay from the appearance of the fireball of 21 minutes, 20 seconds, total signal duration of 2 minutes 10 seconds, the source location determined to be toward 31 degrees from true north, the horizontal trace velocity of 429 m/sec, the signal velocity of 0.29 plus or minus 0.03 km/sec, assuming a 375 km horizontal range to the fireball, the dominant signal frequency content of 0.25 to 0.84 Hz (analyzed in the frequency interval from 0.2 to 2.0 Hz), the maximum signal cross-correlation of 0.97 and the maximum signal amplitude of 2.0 plus or minus 0.1 microbars. Also, on the basis of the signal period at maximum amplitude, we estimate a probable source energy for this event of between 10 to 100 tons of TNT (53.0 tons nominal).

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

PubMed

Kochan, Kay; Sachau, Delf; Breitbach, Harald

2011-05-01

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

Ultra-low-noise preamplifier for condenser microphones.

PubMed

Starecki, Tomasz

2010-12-01

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

Delay-and-sum beamforming for direction of arrival estimation applied to gunshot acoustics

NASA Astrophysics Data System (ADS)

Ramos, António L. L.; Holm, Sverre; Gudvangen, Sigmund; Otterlei, Ragnvald

2011-06-01

Sniper positioning systems described in the literature use a two-step algorithm to estimate the sniper's location. First, the shockwave and the muzzle blast acoustic signatures must be detected and recognized, followed by an estimation of their respective direction-of-arrival (DOA). Second, the actual sniper's position is calculated based on the estimated DOA via an iterative algorithm that varies from system to system. The overall performance of such a system, however, is highly compromised when the first step is not carried out successfully. Currently available systems rely on a simple calculation of differences of time-of-arrival to estimate angles-of-arrival. This approach, however, lacks robustness by not taking full advantage of the array of sensors. This paper shows how the delay-and-sum beamforming technique can be applied to estimate the DOA for both the shockwave and the muzzle blast. The method has the twofold advantage of 1) adding an array gain of 10 logM, i.e., an increased SNR of 6 dB for a 4-microphone array, which is equivalent to doubling the detection range assuming free-field propagation; and 2) offering improved robustness in handling single- and multi-shots events as well as reflections by taking advantage of the spatial filtering capability.

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

NASA Technical Reports Server (NTRS)

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

2006-01-01

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

Contribution of crosstalk to the uncertainty of electrostatic actuator calibrations.

PubMed

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

2009-09-01

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

Unvoiced Speech Recognition Using Tissue-Conductive Acoustic Sensor

NASA Astrophysics Data System (ADS)

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

2006-12-01

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

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

PubMed

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

2010-04-01

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

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

PubMed Central

Homentcovschi, Dorel; Miles, Ronald N.

2011-01-01

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

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

PubMed

Homentcovschi, Dorel; Miles, Ronald N

2011-12-01

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

Optical multi-point measurements of the acoustic particle velocity with frequency modulated Doppler global velocimetry.

PubMed

Fischer, Andreas; König, Jörg; Haufe, Daniel; Schlüssler, Raimund; Büttner, Lars; Czarske, Jürgen

2013-08-01

To reduce the noise of machines such as aircraft engines, the development and propagation of sound has to be investigated. Since the applicability of microphones is limited due to their intrusiveness, contactless measurement techniques are required. For this reason, the present study describes an optical method based on the Doppler effect and its application for acoustic particle velocity (APV) measurements. While former APV measurements with Doppler techniques are point measurements, the applied system is capable of simultaneous measurements at multiple points. In its current state, the system provides linear array measurements of one component of the APV demonstrated by multi-tone experiments with tones up to 17 kHz for the first time.

Development of an Audio Microphone for the Mars Surveyor 98 Lander

NASA Astrophysics Data System (ADS)

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

1998-01-01

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

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

PubMed

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

2014-11-01

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

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

NASA Technical Reports Server (NTRS)

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

1992-01-01

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

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

PubMed

Ranjbar, Parivash; Stenström, Ingeborg

2013-01-01

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

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

PubMed

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

2006-01-01

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

Acoustic/infrasonic rocket engine signatures

NASA Astrophysics Data System (ADS)

Tenney, Stephen M.; Noble, John M.; Whitaker, Rodney W.; ReVelle, Douglas O.

2003-09-01

Infrasonics offers the potential of long-range acoustic detection of explosions, missiles and even sounds created by manufacturing plants. The atmosphere attenuates acoustic energy above 20 Hz quite rapidly, but signals below 10 Hz can propagate to long ranges. Space shuttle launches have been detected infrasonically from over 1000 km away and the Concorde airliner from over 400 km. This technology is based on microphones designed to respond to frequencies from .1 to 300 Hz that can be operated outdoors for extended periods of time with out degrading their performance. The US Army Research Laboratory and Los Alamos National Laboratory have collected acoustic and infrasonic signatures of static engine testing of two missiles. Signatures were collected of a SCUD missile engine at Huntsville, AL and a Minuteman engine at Edwards AFB. The engines were fixed vertically in a test stand during the burn. We will show the typical time waveform signals of these static tests and spectrograms for each type. High resolution, 24-bit data were collected at 512 Hz and 16-bit acoustic data at 10 kHz. Edwards data were recorded at 250 Hz and 50 Hz using a Geotech Instruments 24 bit digitizer. Ranges from the test stand varied from 1 km to 5 km. Low level and upper level meteorological data was collected to provide full details of atmospheric propagation during the engine test. Infrasonic measurements were made with the Chaparral Physics Model 2 microphone with porous garden hose attached for wind noise suppression. A B&K microphone was used for high frequency acoustic measurements. Results show primarily a broadband signal with distinct initiation and completion points. There appear to be features present in the signals that would allow identification of missile type. At 5 km the acoustic/infrasonic signal was clearly present. Detection ranges for the types of missile signatures measured will be predicted based on atmospheric modeling. As part of an experiment conducted by ARL, sounding rocket launches have been detected from over 150 km. A variety of rockets launched from NASA"s Wallops Island facility were detected over a two year span. Arrays of microphones were able to create a line of bearing to the source of the launches that took place during different times of the year. This same experiment has been able to detect the space shuttle from over 1000 km on a regular basis. These two sources represent opposite ends of the target size, but they do demonstrate the potential for the detection and location of rocket launches.

Effect of time-varying tropospheric models on near-regional and regional infrasound propagation as constrained by observational data

NASA Astrophysics Data System (ADS)

McKenna, Mihan H.; Stump, Brian W.; Hayward, Chris

2008-06-01

The Chulwon Seismo-Acoustic Array (CHNAR) is a regional seismo-acoustic array with co-located seismometers and infrasound microphones on the Korean peninsula. Data from forty-two days over the course of a year between October 1999 and August 2000 were analyzed; 2052 infrasound-only arrivals and 23 seismo-acoustic arrivals were observed over the six week study period. A majority of the signals occur during local working hours, hour 0 to hour 9 UT and appear to be the result of cultural activity located within a 250 km radius. Atmospheric modeling is presented for four sample days during the study period, one in each of November, February, April, and August. Local meteorological data sampled at six hour intervals is needed to accurately model the observed arrivals and this data produced highly temporally variable thermal ducts that propagated infrasound signals within 250 km, matching the temporal variation in the observed arrivals. These ducts change dramatically on the order of hours, and meteorological data from the appropriate sampled time frame was necessary to interpret the observed arrivals.

Measured Engine Installation Effects of Four Civil Transport Airplanes

NASA Technical Reports Server (NTRS)

Senzig, David A.; Fleming, Gregg G.; Shepherd, Kevin P.

2001-01-01

The Federal Aviation Administration's Integrated Noise Model (INM) is one of the primary tools for land use planning around airports. The INM currently calculates airplane noise lateral attenuation using the methods contained in the Society of Automotive Engineer's Aerospace Information Report No. 1751 (SAE AIR 1751). Researchers have noted that improved lateral attenuation algorithms may improve airplane noise prediction. The authors of SAE AIR 1751 based existing methods on empirical data collected from flight tests using 1960s-technology airplanes with tail-mounted engines. To determine whether the SAE AIR 1751 methods are applicable for predicting the engine installation component of lateral attenuation for airplanes with wing-mounted engines, the National Aeronautics and Space Administration (NASA) sponsored a series of flight tests during September 2000 at their Wallops Flight Facility. Four airplanes, a Boeing 767-400, a Douglas DC-9, a Dassault Falcon 2000, and a Beech KingAir, were flown through a 20 microphone array. The airplanes were flown through the array at various power settings, flap settings, and altitudes to simulate take-off and arrival configurations. This paper presents the preliminary findings of this study.

STS-97 crew meets with the media at Launch Pad 39B

NASA Technical Reports Server (NTRS)

2000-01-01

Standing in the slidewire landing zone at Launch Pad 39B, the STS-97 crew respond to questions from the media. Commander Brent Jett (on left, with microphone) introduces the rest of the crew (left to right) Pilot Mike Bloomfield and Mission Specialists Joe Tanner, Marc Garneau and Carlos Noriega. Garneau is with the Canadian Space Agency. The nets suspended behind them are a braking system catch net for the slidewire baskets that provide emergency exit from the orbiter and Fixed Service Structure. The crew is at KSC to take part in Terminal Countdown Demonstration Test activities that include emergency egress training, familiarization with the payload, and a simulated launch countdown. Mission STS-97is the sixth construction flight to the International Space Station. Its payload includes the P6 Integrated Truss Structure and a photovoltaic (PV) module, with giant solar arrays that will provide power to the Station. The mission includes two spacewalks to complete the solar array connections. STS-97 is scheduled to launch Nov. 30 at about 10:05 p.m. EST.

NASA Tech Briefs, February 2007

NASA Technical Reports Server (NTRS)

2007-01-01

Topics covered include: Calibration Test Set for a Phase-Comparison Digital Tracker; Wireless Acoustic Measurement System; Spiral Orbit Tribometer; Arrays of Miniature Microphones for Aeroacoustic Testing; Predicting Rocket or Jet Noise in Real Time; Computational Workbench for Multibody Dynamics; High-Power, High-Efficiency Ka-Band Space Traveling-Wave Tube; Gratings and Random Reflectors for Near-Infrared PIN Diodes; Optically Transparent Split-Ring Antennas for 1 to 10 GHz; Ice-Penetrating Robot for Scientific Exploration; Power-Amplifier Module for 145 to 165 GHz; Aerial Videography From Locally Launched Rockets; SiC Multi-Chip Power Modules as Power-System Building Blocks; Automated Design of Restraint Layer of an Inflatable Vessel; TMS for Instantiating a Knowledge Base With Incomplete Data; Simulating Flights of Future Launch Vehicles and Spacecraft; Control Code for Bearingless Switched- Reluctance Motor; Machine Aided Indexing and the NASA Thesaurus; Arbitrating Control of Control and Display Units; Web-Based Software for Managing Research; Driver Code for Adaptive Optics; Ceramic Paste for Patching High-Temperature Insulation; Fabrication of Polyimide-Matrix/Carbon and Boron-Fiber Tape; Protective Skins for Aerogel Monoliths; Code Assesses Risks Posed by Meteoroids and Orbital Debris; Asymmetric Bulkheads for Cylindrical Pressure Vessels; Self-Regulating Water-Separator System for Fuel Cells; Self-Advancing Step-Tap Drills; Array of Bolometers for Submillimeter- Wavelength Operation; Delta-Doped CCDs as Detector Arrays in Mass Spectrometers; Arrays of Bundles of Carbon Nanotubes as Field Emitters; Staggering Inflation To Stabilize Attitude of a Solar Sail; and Bare Conductive Tether for Decelerating a Spacecraft.

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.

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

PubMed

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

2018-01-01

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

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

PubMed Central

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

2018-01-01

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

Measurement of hearing aid internal noise1

PubMed Central

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

Evaluation of Speech Recognition of Cochlear Implant Recipients Using Adaptive, Digital Remote Microphone Technology and a Speech Enhancement Sound Processing Algorithm.

PubMed

Wolfe, Jace; Morais, Mila; Schafer, Erin; Agrawal, Smita; Koch, Dawn

2015-05-01

Cochlear implant recipients often experience difficulty with understanding speech in the presence of noise. Cochlear implant manufacturers have developed sound processing algorithms designed to improve speech recognition in noise, and research has shown these technologies to be effective. Remote microphone technology utilizing adaptive, digital wireless radio transmission has also been shown to provide significant improvement in speech recognition in noise. There are no studies examining the potential improvement in speech recognition in noise when these two technologies are used simultaneously. The goal of this study was to evaluate the potential benefits and limitations associated with the simultaneous use of a sound processing algorithm designed to improve performance in noise (Advanced Bionics ClearVoice) and a remote microphone system that incorporates adaptive, digital wireless radio transmission (Phonak Roger). A two-by-two way repeated measures design was used to examine performance differences obtained without these technologies compared to the use of each technology separately as well as the simultaneous use of both technologies. Eleven Advanced Bionics (AB) cochlear implant recipients, ages 11 to 68 yr. AzBio sentence recognition was measured in quiet and in the presence of classroom noise ranging in level from 50 to 80 dBA in 5-dB steps. Performance was evaluated in four conditions: (1) No ClearVoice and no Roger, (2) ClearVoice enabled without the use of Roger, (3) ClearVoice disabled with Roger enabled, and (4) simultaneous use of ClearVoice and Roger. Speech recognition in quiet was better than speech recognition in noise for all conditions. Use of ClearVoice and Roger each provided significant improvement in speech recognition in noise. The best performance in noise was obtained with the simultaneous use of ClearVoice and Roger. ClearVoice and Roger technology each improves speech recognition in noise, particularly when used at the same time. Because ClearVoice does not degrade performance in quiet settings, clinicians should consider recommending ClearVoice for routine, full-time use for AB implant recipients. Roger should be used in all instances in which remote microphone technology may assist the user in understanding speech in the presence of noise. American Academy of Audiology.

Speech Intelligibility in Various Noise Conditions with the Nucleus® 5 CP810 Sound Processor.

PubMed

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.

An Energy-Efficient and High-Quality Video Transmission Architecture in Wireless Video-Based Sensor Networks.

PubMed

Aghdasi, Hadi S; Abbaspour, Maghsoud; Moghadam, Mohsen Ebrahimi; Samei, Yasaman

2008-08-04

Technological progress in the fields of Micro Electro-Mechanical Systems (MEMS) and wireless communications and also the availability of CMOS cameras, microphones and small-scale array sensors, which may ubiquitously capture multimedia content from the field, have fostered the development of low-cost limited resources Wireless Video-based Sensor Networks (WVSN). With regards to the constraints of videobased sensor nodes and wireless sensor networks, a supporting video stream is not easy to implement with the present sensor network protocols. In this paper, a thorough architecture is presented for video transmission over WVSN called Energy-efficient and high-Quality Video transmission Architecture (EQV-Architecture). This architecture influences three layers of communication protocol stack and considers wireless video sensor nodes constraints like limited process and energy resources while video quality is preserved in the receiver side. Application, transport, and network layers are the layers in which the compression protocol, transport protocol, and routing protocol are proposed respectively, also a dropping scheme is presented in network layer. Simulation results over various environments with dissimilar conditions revealed the effectiveness of the architecture in improving the lifetime of the network as well as preserving the video quality.

Characterization of Unsteady Flow Structures Near Landing-Edge Slat. Part 2; 2D Computations

NASA Technical Reports Server (NTRS)

Khorrami, Mehdi; Choudhari, Meelan M.; Jenkins, Luther N.

2004-01-01

In our previous computational studies of a generic high-lift configuration, quasi-laminar (as opposed to fully turbulent) treatment of the slat cove region proved to be an effective approach for capturing the unsteady dynamics of the cove flow field. Combined with acoustic propagation via Ffowes Williams and Hawkings formulation, the quasi-laminar simulations captured some important features of the slat cove noise measured with microphone array techniques. However. a direct assessment of the computed cove flow field was not feasible due to the unavailability of off-surface flow measurements. To remedy this shortcoming, we have undertaken a combined experiment and computational study aimed at characterizing the flow structures and fluid mechanical processes within the slat cove region. Part I of this paper outlines the experimental aspects of this investigation focused on the 30P30N high-lift configuration; the present paper describes the accompanying computational results including a comparison between computation and experiment at various angles of attack. Even through predictions of the time-averaged flow field agree well with the measured data, the study indicates the need for further refinement of the zonal turbulence approach in order to capture the full dynamics of the cove's fluctuating flow field.

Real-time spectrum estimation–based dual-channel speech-enhancement algorithm for cochlear implant

PubMed Central

2012-01-01

Background Improvement of the cochlear implant (CI) front-end signal acquisition is needed to increase speech recognition in noisy environments. To suppress the directional noise, we introduce a speech-enhancement algorithm based on microphone array beamforming and spectral estimation. The experimental results indicate that this method is robust to directional mobile noise and strongly enhances the desired speech, thereby improving the performance of CI devices in a noisy environment. Methods The spectrum estimation and the array beamforming methods were combined to suppress the ambient noise. The directivity coefficient was estimated in the noise-only intervals, and was updated to fit for the mobile noise. Results The proposed algorithm was realized in the CI speech strategy. For actual parameters, we use Maxflat filter to obtain fractional sampling points and cepstrum method to differentiate the desired speech frame and the noise frame. The broadband adjustment coefficients were added to compensate the energy loss in the low frequency band. Discussions The approximation of the directivity coefficient is tested and the errors are discussed. We also analyze the algorithm constraint for noise estimation and distortion in CI processing. The performance of the proposed algorithm is analyzed and further be compared with other prevalent methods. Conclusions The hardware platform was constructed for the experiments. The speech-enhancement results showed that our algorithm can suppresses the non-stationary noise with high SNR. Excellent performance of the proposed algorithm was obtained in the speech enhancement experiments and mobile testing. And signal distortion results indicate that this algorithm is robust with high SNR improvement and low speech distortion. PMID:23006896

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

PubMed

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

2009-07-01

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

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.

Wireless acoustic modules for real-time data fusion using asynchronous sniper localization algorithms

NASA Astrophysics Data System (ADS)

Hengy, S.; De Mezzo, S.; Duffner, P.; Naz, P.

2012-11-01

The presence of snipers in modern conflicts leads to high insecurity for the soldiers. In order to improve the soldier's protection against this threat, the French German Research Institute of Saint-Louis (ISL) has been conducting studies in the domain of acoustic localization of shots. Mobile antennas mounted on the soldier's helmet were initially used for real-time detection, classification and localization of sniper shots. It showed good performances in land scenarios, but also in urban scenarios if the array was in the shot corridor, meaning that the microphones first detect the direct wave and then the reflections of the Mach and muzzle waves (15% distance estimation error compared to the actual shooter array distance). Fusing data sent by multiple sensor nodes distributed on the field showed some of the limitations of the technologies that have been implemented in ISL's demonstrators. Among others, the determination of the arrays' orientation was not accurate enough, thereby degrading the performance of data fusion. Some new solutions have been developed in the past year in order to obtain better performance for data fusion. Asynchronous localization algorithms have been developed and post-processed on data measured in both free-field and urban environments with acoustic modules on the line of sight of the shooter. These results are presented in the first part of the paper. The impact of GPS position estimation error is also discussed in the article in order to evaluate the possible use of those algorithms for real-time processing using mobile acoustic nodes. In the frame of ISL's transverse project IMOTEP (IMprovement Of optical and acoustical TEchnologies for the Protection), some demonstrators are developed that will allow real-time asynchronous localization of sniper shots. An embedded detection and classification algorithm is implemented on wireless acoustic modules that send the relevant information to a central PC. Data fusion is then processed and the estimated position of the shooter is sent back to the users. A SWIR active imaging system is used for localization refinement. A built-in DSP is related to the detection/classification tasks for each acoustic module. A GPS module is used for time difference of arrival and module's position estimation. Wireless communication is supported using ZigBee technology. These acoustic modules are described in the article and first results of real-time asynchronous sniper localization using those modules are discussed.

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.

Extreme Low Frequency Acoustic Measurement System

NASA Technical Reports Server (NTRS)

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

2017-01-01

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

Lumped-parameters equivalent circuit for condenser microphones modeling.

PubMed

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

2017-10-01

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

Extreme low frequency acoustic measurement system

NASA Technical Reports Server (NTRS)

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

2013-01-01

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

Robust snow avalanche detection using machine learning on infrasonic array data

NASA Astrophysics Data System (ADS)

Thüring, Thomas; Schoch, Marcel; van Herwijnen, Alec; Schweizer, Jürg

2014-05-01

Snow avalanches may threaten people and infrastructure in mountain areas. Automated detection of avalanche activity would be highly desirable, in particular during times of poor visibility, to improve hazard assessment, but also to monitor the effectiveness of avalanche control by explosives. In the past, a variety of remote sensing techniques and instruments for the automated detection of avalanche activity have been reported, which are based on radio waves (radar), seismic signals (geophone), optical signals (imaging sensor) or infrasonic signals (microphone). Optical imagery enables to assess avalanche activity with very high spatial resolution, however it is strongly weather dependent. Radar and geophone-based detection typically provide robust avalanche detection for all weather conditions, but are very limited in the size of the monitoring area. On the other hand, due to the long propagation distance of infrasound through air, the monitoring area of infrasonic sensors can cover a large territory using a single sensor (or an array). In addition, they are by far more cost effective than radars or optical imaging systems. Unfortunately, the reliability of infrasonic sensor systems has so far been rather low due to the strong variation of ambient noise (e.g. wind) causing a high false alarm rate. We analyzed the data collected by a low-cost infrasonic array system consisting of four sensors for the automated detection of avalanche activity at Lavin in the eastern Swiss Alps. A comparably large array aperture (~350m) allows highly accurate time delay estimations of signals which arrive at different times at the sensors, enabling precise source localization. An array of four sensors is sufficient for the time resolved source localization of signals in full 3D space, which is an excellent method to anticipate true avalanche activity. Robust avalanche detection is then achieved by using machine learning methods such as support vector machines. The system is initially trained by using characteristic data features from known avalanche and non-avalanche events. Data features are obtained from output signals of the source localization algorithm or from Fourier or time domain processing and support the learning phase of the system. A significantly improved detection rate as well as a reduction of the false alarm rate was achieved compared to previous approaches.

Generation of Higher Order Modes in a Rectangular Duct

NASA Technical Reports Server (NTRS)

Gerhold, Carl H.; Cabell, Randolph H.; Brown, Donald E.

2004-01-01

Advanced noise control methodologies to reduce sound emission from aircraft engines take advantage of the modal structure of the noise in the duct. This noise is caused by the interaction of rotor wakes with downstream obstructions such as exit guide vanes. Mode synthesis has been accomplished in circular ducts and current active noise control work has made use of this capability to cancel fan noise. The goal of the current effort is to examine the fundamental process of higher order mode propagation through an acoustically treated, curved duct. The duct cross-section is rectangular to permit greater flexibility in representation of a range of duct curvatures. The work presented is the development of a feedforward control system to generate a user-specified modal pattern in the duct. The multiple-error, filtered-x LMS algorithm is used to determine the magnitude and phase of signal input to the loudspeakers to produce a desired modal pattern at a set of error microphones. Implementation issues, including loudspeaker placement and error microphone placement, are discussed. Preliminary results from a 9-3/8 inch by 21 inch duct, using 12 loudspeakers and 24 microphones, are presented. These results demonstrate the ability of the control system to generate a user-specified mode while suppressing undesired modes.

Left-right and front-back spatial hearing with multiple directional microphone configurations in modern hearing aids.

PubMed

Carette, Evelyne; Van den Bogaert, Tim; Laureyns, Mark; Wouters, Jan

2014-10-01

Several studies have demonstrated negative effects of directional microphone configurations on left-right and front-back (FB) sound localization. New processing schemes, such as frequency-dependent directionality and front focus with wireless ear-to-ear communication in recent, commercial hearing aids may preserve the binaural cues necessary for left-right localization and may introduce useful spectral cues necessary for FB disambiguation. In this study, two hearing aids with different processing schemes, which were both designed to preserve the ability to localize sounds in the horizontal plane (left-right and FB), were compared. We compared horizontal (left-right and FB) sound localization performance of hearing aid users fitted with two types of behind-the-ear (BTE) devices. The first type of BTE device had four different programs that provided (1) no directionality, (2-3) symmetric frequency-dependent directionality, and (4) an asymmetric configuration. The second pair of BTE devices was evaluated in its omnidirectional setting. This setting automatically activates a soft forward-oriented directional scheme that mimics the pinna effect. Also, wireless communication between the hearing aids was present in this configuration (5). A broadband stimulus was used as a target signal. The directional hearing abilities of the listeners were also evaluated without hearing aids as a reference. A total of 12 listeners with moderate to severe hearing loss participated in this study. All were experienced hearing-aid users. As a reference, 11 listeners with normal hearing participated. The participants were positioned in a 13-speaker array (left-right, -90°/+90°) or 7-speaker array (FB, 0-180°) and were asked to report the number of the loudspeaker located the closest to where the sound was perceived. The root mean square error was calculated for the left-right experiment, and the percentage of FB errors was used as a FB performance measure. RESULTS were analyzed with repeated-measures analysis of variance. For the left-right localization task, no significant differences could be proven between the unaided condition and both partial directional schemes and the omnidirectional scheme. The soft forward-oriented system and the asymmetric system did show a detrimental effect compared with the unaided condition. On average, localization was worst when users used the asymmetric condition. Analysis of the results of the FB experiment showed good performance, similar to unaided, with both the partial directional systems and the asymmetric configuration. Significantly worse performance was found with the omnidirectional and the omnidirectional soft forward-oriented BTE systems compared with the other hearing-aid systems. Bilaterally fitted partial directional systems preserve (part of) the binaural cues necessary for left-right localization and introduce, preserve, or enhance useful spectral cues that allow FB disambiguation. Omnidirectional systems, although good for left-right localization, do not provide the user with enough spectral information for an optimal FB localization performance. American Academy of Audiology.