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.

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

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.

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.

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

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.

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

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

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.

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.

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

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

NASA Technical Reports Server (NTRS)

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

2015-01-01

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

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.

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

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.

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.

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.

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.

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.

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

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.

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

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

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

NASA Technical Reports Server (NTRS)

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

1999-01-01

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

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.

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

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.

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.

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.

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.

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.

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.

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.

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.

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.

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 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.

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

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.

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.

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.

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.

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.

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.

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.

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 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.

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

PubMed

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

2007-02-01

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

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

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.

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.

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.

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.

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.

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

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.

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

PubMed

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

2017-10-13

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

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.

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

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

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.

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.

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…

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.

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.

SMI adaptive antenna arrays for weak interfering signals

NASA Technical Reports Server (NTRS)

Gupta, I. J.

1987-01-01

The performance of adaptive antenna arrays is studied when a sample matrix inversion (SMI) algorithm is used to control array weights. It is shown that conventional SMI adaptive antennas, like other adaptive antennas, are unable to suppress weak interfering signals (below thermal noise) encountered in broadcasting satellite communication systems. To overcome this problem, the SMI algorithm is modified. In the modified algorithm, the covariance matrix is modified such that the effect of thermal noise on the weights of the adaptive array is reduced. Thus, the weights are dictated by relatively weak coherent signals. It is shown that the modified algorithm provides the desired interference protection. The use of defocused feeds as auxiliary elements of an SMI adaptive array is also discussed.

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

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.

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.

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.

Probe Microphone Measurements: 20 Years of Progress

PubMed Central

Mueller, H. Gustav

2001-01-01

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

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.

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

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

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.

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

DOT National Transportation Integrated Search

2006-05-08

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

Optimizing Satellite Communications With Adaptive and Phased Array Antennas

NASA Technical Reports Server (NTRS)

Ingram, Mary Ann; Romanofsky, Robert; Lee, Richard Q.; Miranda, Felix; Popovic, Zoya; Langley, John; Barott, William C.; Ahmed, M. Usman; Mandl, Dan

2004-01-01

A new adaptive antenna array architecture for low-earth-orbiting satellite ground stations is being investigated. These ground stations are intended to have no moving parts and could potentially be operated in populated areas, where terrestrial interference is likely. The architecture includes multiple, moderately directive phased arrays. The phased arrays, each steered in the approximate direction of the satellite, are adaptively combined to enhance the Signal-to-Noise and Interference-Ratio (SNIR) of the desired satellite. The size of each phased array is to be traded-off with the number of phased arrays, to optimize cost, while meeting a bit-error-rate threshold. Also, two phased array architectures are being prototyped: a spacefed lens array and a reflect-array. If two co-channel satellites are in the field of view of the phased arrays, then multi-user detection techniques may enable simultaneous demodulation of the satellite signals, also known as Space Division Multiple Access (SDMA). We report on Phase I of the project, in which fixed directional elements are adaptively combined in a prototype to demodulate the S-band downlink of the EO-1 satellite, which is part of the New Millennium Program at NASA.

Adaptive arrays for satellite communications

NASA Technical Reports Server (NTRS)

Gupta, I. J.; Ksienski, A. A.

1984-01-01

The suppression of interfering signals in a satellite communication system was studied. Adaptive arrays are used to suppress interference at the reception site. It is required that the interference be suppressed to very low levels and a modified adaptive circuit is used which accomplishes the desired objective. Techniques for the modification of the transmit patterns to minimize interference with neighboring communication links are explored.

Removing Background Noise with Phased Array Signal Processing

NASA Technical Reports Server (NTRS)

Podboy, Gary; Stephens, David

2015-01-01

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

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

Adaptive array antenna for satellite cellular and direct broadcast communications

NASA Technical Reports Server (NTRS)

Horton, Charles R.; Abend, Kenneth

1993-01-01

Adaptive phased-array antennas provide cost-effective implementation of large, light weight apertures with high directivity and precise beamshape control. Adaptive self-calibration allows for relaxation of all mechanical tolerances across the aperture and electrical component tolerances, providing high performance with a low-cost, lightweight array, even in the presence of large physical distortions. Beam-shape is programmable and adaptable to changes in technical and operational requirements. Adaptive digital beam-forming eliminates uplink contention by allowing a single electronically steerable antenna to service a large number of receivers with beams which adaptively focus on one source while eliminating interference from others. A large, adaptively calibrated and fully programmable aperture can also provide precise beam shape control for power-efficient direct broadcast from space. Advanced adaptive digital beamforming technologies are described for: (1) electronic compensation of aperture distortion, (2) multiple receiver adaptive space-time processing, and (3) downlink beam-shape control. Cost considerations for space-based array applications are also discussed.

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.

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.

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.

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

Experimental Demonstration of Adaptive Infrared Multispectral Imaging Using Plasmonic Filter Array (Postprint)

DTIC Science & Technology

2016-10-10

AFRL-RX-WP-JA-2017-0189 EXPERIMENTAL DEMONSTRATION OF ADAPTIVE INFRARED MULTISPECTRAL IMAGING USING PLASMONIC FILTER ARRAY...March 2016 – 23 May 2016 4. TITLE AND SUBTITLE EXPERIMENTAL DEMONSTRATION OF ADAPTIVE INFRARED MULTISPECTRAL IMAGING USING PLASMONIC FILTER ARRAY...experimental demonstration of adaptive multispectral imagery using fabricated plasmonic spectral filter arrays and proposed target detection scenarios

The Semicircular Canal Microphonic

NASA Technical Reports Server (NTRS)

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

2002-01-01

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

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.

[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).

An experimental SMI adaptive antenna array simulator for weak interfering signals

NASA Technical Reports Server (NTRS)

Dilsavor, Ronald S.; Gupta, Inder J.

1991-01-01

An experimental sample matrix inversion (SMI) adaptive antenna array for suppressing weak interfering signals is described. The experimental adaptive array uses a modified SMI algorithm to increase the interference suppression. In the modified SMI algorithm, the sample covariance matrix is redefined to reduce the effect of thermal noise on the weights of an adaptive array. This is accomplished by subtracting a fraction of the smallest eigenvalue of the original covariance matrix from its diagonal entries. The test results obtained using the experimental system are compared with theoretical results. The two show a good agreement.

SMI adaptive antenna arrays for weak interfering signals. [Sample Matrix Inversion

NASA Technical Reports Server (NTRS)

Gupta, Inder J.

1986-01-01

The performance of adaptive antenna arrays in the presence of weak interfering signals (below thermal noise) is studied. It is shown that a conventional adaptive antenna array sample matrix inversion (SMI) algorithm is unable to suppress such interfering signals. To overcome this problem, the SMI algorithm is modified. In the modified algorithm, the covariance matrix is redefined such that the effect of thermal noise on the weights of adaptive arrays is reduced. Thus, the weights are dictated by relatively weak signals. It is shown that the modified algorithm provides the desired interference protection.

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

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.

Comparison of speech recognition with adaptive digital and FM remote microphone hearing assistance technology by listeners who use hearing aids.

PubMed

Thibodeau, Linda

2014-06-01

The purpose of this study was to compare the benefits of 3 types of remote microphone hearing assistance technology (HAT), adaptive digital broadband, adaptive frequency modulation (FM), and fixed FM, through objective and subjective measures of speech recognition in clinical and real-world settings. Participants included 11 adults, ages 16 to 78 years, with primarily moderate-to-severe bilateral hearing impairment (HI), who wore binaural behind-the-ear hearing aids; and 15 adults, ages 18 to 30 years, with normal hearing. Sentence recognition in quiet and in noise and subjective ratings were obtained in 3 conditions of wireless signal processing. Performance by the listeners with HI when using the adaptive digital technology was significantly better than that obtained with the FM technology, with the greatest benefits at the highest noise levels. The majority of listeners also preferred the digital technology when listening in a real-world noisy environment. The wireless technology allowed persons with HI to surpass persons with normal hearing in speech recognition in noise, with the greatest benefit occurring with adaptive digital technology. The use of adaptive digital technology combined with speechreading cues would allow persons with HI to engage in communication in environments that would have otherwise not been possible with traditional wireless technology.

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.

Effects of additional interfering signals on adaptive array performance

NASA Technical Reports Server (NTRS)

Moses, Randolph L.

1989-01-01

The effects of additional interference signals on the performance of a fully adaptive array are considered. The case where the number of interference signals exceeds the number of array degrees of freedom is addressed. It is shown how performance is affected as a function of the number of array elements, the number of interference signals, and the directivity of the array antennas. By using directive auxiliary elements, the performance of the array can be as good as the performance when the additional interference signals are not present.

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.

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.

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

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.

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

NASA Technical Reports Server (NTRS)

Cohn, R. B. (Inventor)

1983-01-01

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

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

Unstructured Adaptive Grid Computations on an Array of SMPs

NASA Technical Reports Server (NTRS)

Biswas, Rupak; Pramanick, Ira; Sohn, Andrew; Simon, Horst D.

1996-01-01

Dynamic load balancing is necessary for parallel adaptive methods to solve unsteady CFD problems on unstructured grids. We have presented such a dynamic load balancing framework called JOVE, in this paper. Results on a four-POWERnode POWER CHALLENGEarray demonstrated that load balancing gives significant performance improvements over no load balancing for such adaptive computations. The parallel speedup of JOVE, implemented using MPI on the POWER CHALLENCEarray, was significant, being as high as 31 for 32 processors. An implementation of JOVE that exploits 'an array of SMPS' architecture was also studied; this hybrid JOVE outperformed flat JOVE by up to 28% on the meshes and adaption models tested. With large, realistic meshes and actual flow-solver and adaption phases incorporated into JOVE, hybrid JOVE can be expected to yield significant advantage over flat JOVE, especially as the number of processors is increased, thus demonstrating the scalability of an array of SMPs architecture.

The Effect of a Pulsed Interference Signal on an Adaptive Array.

DTIC Science & Technology

1981-04-01

eigenvectors exist.) Using a spectral decomp- osition formula [10,11], we may write e-kM in the form -kM -k(T- )-kpo 3 -kg i (e -k = e a : .iZ e eie i , (28...N 0 (No/T b) In addition, for this analysis we shall assume the interference power at the array output has the same effect on detector performance... Sensitive Adaptive Array," to appear in IEEE Trans. Antennas and Propagation. 7. R.T. Compton, Jr., "The Tripole Antenna - An Adaptive Array with Full

Principles of Adaptive Array Processing

DTIC Science & Technology

2006-09-01

ACE with and without tapering (homogeneous case). These analytical results are less suited to predict the detection performance of a real system ...Nickel: Adaptive Beamforming for Phased Array Radars. Proc. Int. Radar Symposium IRS’98 (Munich, Sept. 1998), DGON and VDE /ITG, pp. 897-906.(Reprint also...strategies for airborne radar. Asilomar Conf. on Signals, Systems and Computers, Pacific Grove, CA, 1998, IEEE Cat.Nr. 0-7803-5148-7/98, pp. 1327-1331. [17

Micromirror Arrays for Adaptive Optics

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

Carr, E.J.

The long-range goal of this project is to develop the optical and mechanical design of a micromirror array for adaptive optics that will meet the following criteria: flat mirror surface ({lambda}/20), high fill factor (> 95%), large stroke (5-10 {micro}m), and pixel size {approx}-200 {micro}m. This will be accomplished by optimizing the mirror surface and actuators independently and then combining them using bonding technologies that are currently being developed.

Mission-Oriented Sensor Arrays and UAVs - a Case Study on Environmental Monitoring

NASA Astrophysics Data System (ADS)

Figueira, N. M.; Freire, I. L.; Trindade, O.; Simões, E.

2015-08-01

This paper presents a new concept of UAV mission design in geomatics, applied to the generation of thematic maps for a multitude of civilian and military applications. We discuss the architecture of Mission-Oriented Sensors Arrays (MOSA), proposed in Figueira et Al. (2013), aimed at splitting and decoupling the mission-oriented part of the system (non safety-critical hardware and software) from the aircraft control systems (safety-critical). As a case study, we present an environmental monitoring application for the automatic generation of thematic maps to track gunshot activity in conservation areas. The MOSA modeled for this application integrates information from a thermal camera and an on-the-ground microphone array. The use of microphone arrays technology is of particular interest in this paper. These arrays allow estimation of the direction-of-arrival (DOA) of the incoming sound waves. Information about events of interest is obtained by the fusion of the data provided by the microphone array, captured by the UAV, fused with information from the termal image processing. Preliminary results show the feasibility of the on-the-ground sound processing array and the simulation of the main processing module, to be embedded into an UAV in a future work. The main contributions of this paper are the proposed MOSA system, including concepts, models and architecture.

Implementation of LSCMA adaptive array terminal for mobile satellite communications

NASA Astrophysics Data System (ADS)

Zhou, Shun; Wang, Huali; Xu, Zhijun

2007-11-01

This paper considers the application of adaptive array antenna based on the least squares constant modulus algorithm (LSCMA) for interference rejection in mobile SATCOM terminals. A two-element adaptive array scheme is implemented with a combination of ADI TS201S DSP chips and Altera Stratix II FPGA device, which makes a cooperating computation for adaptive beamforming. Its interference suppressing performance is verified via Matlab simulations. Digital hardware system is implemented to execute the operations of LSCMA beamforming algorithm that is represented by an algorithm flowchart. The result of simulations and test indicate that this scheme can improve the anti-jamming performance of terminals.

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.

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…

Adaptive ground implemented phase array

NASA Technical Reports Server (NTRS)

Spearing, R. E.

1973-01-01

The simulation of an adaptive ground implemented phased array of five antenna elements is reported for a very high frequency system design that is tolerant to the radio frequency interference environment encountered by a tracking data relay satellite. Signals originating from satellites are received by the VHF ring array and both horizontal and vertical polarizations from each of the five elements are multiplexed and transmitted down to ground station. A panel on the transmitting end of the simulation chamber contains up to 10 S-band RFI sources along with the desired signal to simulate the dynamic relationship between user and TDRS. The 10 input channels are summed, and desired and interference signals are separated and corrected until the resultant sum signal-to-interference ratio is maximized. Testing performed with this simulation equipment demonstrates good correlation between predicted and actual results.

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.

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.

Adaptive antenna arrays for weak interfering signals

NASA Technical Reports Server (NTRS)

Gupta, I. J.

1985-01-01

The interference protection provided by adaptive antenna arrays to an Earth station or satellite receive antenna system is studied. The case where the interference is caused by the transmission from adjacent satellites or Earth stations whose signals inadverently enter the receiving system and interfere with the communication link is considered. Thus, the interfering signals are very weak. To increase the interference suppression, one can either decrease the thermal noise in the feedback loops or increase the gain of the auxiliary antennas in the interfering signal direction. Both methods are examined. It is shown that one may have to reduce the noise correlation to impractically low values and if directive auxiliary antennas are used, the auxiliary antenna size may have to be too large. One can, however, combine the two methods to achieve the specified interference suppression with reasonable requirements of noise decorrelation and auxiliary antenna size. Effects of the errors in the steering vector on the adaptive array performance are studied.

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.

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.

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.

LEO Download Capacity Analysis for a Network of Adaptive Array Ground Stations

NASA Technical Reports Server (NTRS)

Ingram, Mary Ann; Barott, William C.; Popovic, Zoya; Rondineau, Sebastien; Langley, John; Romanofsky, Robert; Lee, Richard Q.; Miranda, Felix; Steffes, Paul; Mandl, Dan

2005-01-01

To lower costs and reduce latency, a network of adaptive array ground stations, distributed across the United States, is considered for the downlink of a polar-orbiting low earth orbiting (LEO) satellite. Assuming the X-band 105 Mbps transmitter of NASA s Earth Observing 1 (EO-1) satellite with a simple line-of-sight propagation model, the average daily download capacity in bits for a network of adaptive array ground stations is compared to that of a single 11 m dish in Poker Flats, Alaska. Each adaptive array ground station is assumed to have multiple steerable antennas, either mechanically steered dishes or phased arrays that are mechanically steered in azimuth and electronically steered in elevation. Phased array technologies that are being developed for this application are the space-fed lens (SFL) and the reflectarray. Optimization of the different boresight directions of the phased arrays within a ground station is shown to significantly increase capacity; for example, this optimization quadruples the capacity for a ground station with eight SFLs. Several networks comprising only two to three ground stations are shown to meet or exceed the capacity of the big dish, Cutting the data rate by half, which saves modem costs and increases the coverage area of each ground station, is shown to increase the average daily capacity of the network for some configurations.

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.

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.

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.

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

PubMed

Posatskiy, A O; Chau, T

2012-04-01

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

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.

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.

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

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.

Development of adaptive liquid microlenses and microlens arrays

NASA Astrophysics Data System (ADS)

Berry, Shaun R.; Stewart, Jason B.; Thorsen, Todd A.; Guha, Ingrid

2013-03-01

We report on the development of sub-millimeter size adaptive liquid microlenses and microlens arrays using two immiscible liquids to form individual lenses. Microlenses and microlens arrays having aperture diameters as small as 50 microns were fabricated on a planar quartz substrate using patterned hydrophobic/hydrophilic regions. Liquid lenses were formed by a self-assembled oil dosing process that created well-defined lenses having a high fill factor. Variable focus was achieved by controlling the lens curvature through electrowetting. Greater than 70° of contact angle change was achieved with less than 20 volts, which results in a large optical power dynamic range.

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.

Adaptive Injection-locking Oscillator Array for RF Spectrum Analysis

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

Leung, Daniel

2011-04-19

A highly parallel radio frequency receiver using an array of injection-locking oscillators for on-chip, rapid estimation of signal amplitudes and frequencies is considered. The oscillators are tuned to different natural frequencies, and variable gain amplifiers are used to provide negative feedback to adapt the locking band-width with the input signal to yield a combined measure of input signal amplitude and frequency detuning. To further this effort, an array of 16 two-stage differential ring oscillators and 16 Gilbert-cell mixers is designed for 40-400 MHz operation. The injection-locking oscillator array is assembled on a custom printed-circuit board. Control and calibration is achievedmore » by on-board microcontroller.« less

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.

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.

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.

Optimized micromirror arrays for adaptive optics

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

Michalicek, M. Adrian

This paper describes the design, layout, fabrication, and surface characterization of highly optimized surface micromachined micromirror devices. Design considerations and fabrication capabilities are presented. These devices are fabricated in the state-of-the-art, four-level, planarized, ultra-low-stress polysilicon process available at Sandia National Laboratories known as the Sandia Ultra-planar Multi-level MEMS Technology (SUMMiT). This enabling process permits the development of micromirror devices with near-ideal characteristics that have previously been unrealizable in standard three-layer polysilicon processes. The reduced 1 {mu}m minimum feature sizes and 0.1 {mu}m mask resolution make it possible to produce dense wiring patterns and irregularly shaped flexures. Likewise, mirror surfaces canmore » be uniquely distributed and segmented in advanced patterns and often irregular shapes in order to minimize wavefront error across the pupil. The ultra-low-stress polysilicon and planarized upper layer allow designers to make larger and more complex micromirrors of varying shape and surface area within an array while maintaining uniform performance of optical surfaces. Powerful layout functions of the AutoCAD editor simplify the design of advanced micromirror arrays and make it possible to optimize devices according to the capabilities of the fabrication process. Micromirrors fabricated in this process have demonstrated a surface variance across the array from only 2{endash}3 nm to a worst case of roughly 25 nm while boasting active surface areas of 98{percent} or better. Combining the process planarization with a {open_quotes}planarized-by-design{close_quotes} approach will produce micromirror array surfaces that are limited in flatness only by the surface deposition roughness of the structural material. Ultimately, the combination of advanced process and layout capabilities have permitted the fabrication of highly optimized micromirror arrays for adaptive

Fiber Optic Microphone

NASA Technical Reports Server (NTRS)

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

1999-01-01

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

Adaptive antenna arrays for satellite communication

NASA Technical Reports Server (NTRS)

Gupta, Inder J.

1989-01-01

The feasibility of using adaptive antenna arrays to provide interference protection in satellite communications was studied. The feedback loops as well as the sample matric inversion (SMI) algorithm for weight control were studied. Appropriate modifications in the two were made to achieve the required interference suppression. An experimental system was built to test the modified feedback loops and the modified SMI algorithm. The performance of the experimental system was evaluated using bench generated signals and signals received from TVRO geosynchronous satellites. A summary of results is given. Some suggestions for future work are also presented.

Dynamic Adaptive Neural Network Arrays: A Neuromorphic Architecture

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

Disney, Adam; Reynolds, John

2015-01-01

Dynamic Adaptive Neural Network Array (DANNA) is a neuromorphic hardware implementation. It differs from most other neuromorphic projects in that it allows for programmability of structure, and it is trained or designed using evolutionary optimization. This paper describes the DANNA structure, how DANNA is trained using evolutionary optimization, and an application of DANNA to a very simple classification task.

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.

Study of large adaptive arrays for space technology applications

NASA Technical Reports Server (NTRS)

Berkowitz, R. S.; Steinberg, B.; Powers, E.; Lim, T.

1977-01-01

The research in large adaptive antenna arrays for space technology applications is reported. Specifically two tasks were considered. The first was a system design study for accurate determination of the positions and the frequencies of sources radiating from the earth's surface that could be used for the rapid location of people or vehicles in distress. This system design study led to a nonrigid array about 8 km in size with means for locating the array element positions, receiving signals from the earth and determining the source locations and frequencies of the transmitting sources. It is concluded that this system design is feasible, and satisfies the desired objectives. The second task was an experiment to determine the largest earthbound array which could simulate a spaceborne experiment. It was determined that an 800 ft array would perform indistinguishably in both locations and it is estimated that one several times larger also would serve satisfactorily. In addition the power density spectrum of the phase difference fluctuations across a large array was measured. It was found that the spectrum falls off approximately as f to the minus 5/2 power.

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.

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

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

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.

Adaptive antenna arrays for weak interfering signals. [in satellite communication

NASA Technical Reports Server (NTRS)

Gupta, I. J.; Ksienski, A. A.

1986-01-01

It is shown that conventional adaptive arrays are unable to suppress weak interfering signals. To overcome this problem, the feedback loops controlling the array weights were modified, reducing the noise level by reducing the correlation between the noise components of the two inputs to the loop correlator. Various techniques to decorrelate these noise components are discussed. An expression is derived for the amount of noise decorrelation required to achieve a specified interference suppression. The results are of interest in connection with satellite communications.

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.

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.

An adaptive array antenna for mobile satellite communications

NASA Technical Reports Server (NTRS)

Milne, Robert

1988-01-01

The adaptive array is linearly polarized and consists essentially of a driven lambda/4 monopole surrounded by an array of parasitic elements all mounted on a ground plane of finite size. The parasitic elements are all connected to ground via pin diodes. By applying suitable bias voltages, the desired parasitic elements can be activated and made highly reflective. The directivity and pointing of the antenna beam can be controlled in both the azimuth and elevation planes using high speed digital switching techniques. The antenna RF losses are neglible and the maximum gain is close to the theoretical value determined by the effective aperture size. The antenna is compact, has a low profile, is inexpensive to manufacture and can handle high transmitter power.

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.

Self-Adaptive System based on Field Programmable Gate Array for Extreme Temperature Electronics

NASA Technical Reports Server (NTRS)

Keymeulen, Didier; Zebulum, Ricardo; Rajeshuni, Ramesham; Stoica, Adrian; Katkoori, Srinivas; Graves, Sharon; Novak, Frank; Antill, Charles

2006-01-01

In this work, we report the implementation of a self-adaptive system using a field programmable gate array (FPGA) and data converters. The self-adaptive system can autonomously recover the lost functionality of a reconfigurable analog array (RAA) integrated circuit (IC) [3]. Both the RAA IC and the self-adaptive system are operating in extreme temperatures (from 120 C down to -180 C). The RAA IC consists of reconfigurable analog blocks interconnected by several switches and programmable by bias voltages. It implements filters/amplifiers with bandwidth up to 20 MHz. The self-adaptive system controls the RAA IC and is realized on Commercial-Off-The-Shelf (COTS) parts. It implements a basic compensation algorithm that corrects a RAA IC in less than a few milliseconds. Experimental results for the cold temperature environment (down to -180 C) demonstrate the feasibility of this approach.

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 Adaptive Beamforming to Signal Observations at the Mt. Meron Array, Israel

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

Harris, D. B.

2010-06-07

The Mt. Meron array consists of 16 stations spanning an aperture of 3-4 kilometers in northern Israel. The array is situated in a region of substantial topographic relief, and is surrounded by settlements at close range (Figure 1). Consequently the level of noise at the array is high, which requires efforts at mitigation if distant regional events of moderate magnitude are to be observed. This note describes an initial application of two classic adaptive beamforming algorithms to data from the array to observe P waves from 5 events east of the array ranging in distance from 1100- 2150 kilometers.

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.

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.

Optical microphone

DOEpatents

Veligdan, James T.

2000-01-11

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

Dynamic Pressure Microphones

NASA Astrophysics Data System (ADS)

Werner, E.

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

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.

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.

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.

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.

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.

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.

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.

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...

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

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.

Microphone Detects Boiler-Tube Leaks

NASA Technical Reports Server (NTRS)

Parthasarathy, S. P.

1985-01-01

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

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…

The capture and recreation of 3D auditory scenes

NASA Astrophysics Data System (ADS)

Li, Zhiyun

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

Adaptive multibeam phased array design for a Spacelab experiment

NASA Technical Reports Server (NTRS)

Noji, T. T.; Fass, S.; Fuoco, A. M.; Wang, C. D.

1977-01-01

The parametric tradeoff analyses and design for an Adaptive Multibeam Phased Array (AMPA) for a Spacelab experiment are described. This AMPA Experiment System was designed with particular emphasis to maximize channel capacity and minimize implementation and cost impacts for future austere maritime and aeronautical users, operating with a low gain hemispherical coverage antenna element, low effective radiated power, and low antenna gain-to-system noise temperature ratio.

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.

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 adaptive array antenna for mobile satellite communications

NASA Technical Reports Server (NTRS)

Milne, Robert

1990-01-01

The design of an adaptive array antenna for land vehicle operation and its performance in an operational satellite system is described. Linear and circularly polarized antenna designs are presented. The acquisition and tracking operation of a satellite is described and the effect on the communications signal is discussed. A number of system requirements are examined that have a major impact on the antenna design. The results of environmental, power handling, and RFI testing are presented and potential problems are identified.

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

Adaptive Detector Arrays for Optical Communications Receivers

NASA Technical Reports Server (NTRS)

Vilnrotter, V.; Srinivasan, M.

2000-01-01

The structure of an optimal adaptive array receiver for ground-based optical communications is described and its performance investigated. Kolmogorov phase screen simulations are used to model the sample functions of the focal-plane signal distribution due to turbulence and to generate realistic spatial distributions of the received optical field. This novel array detector concept reduces interference from background radiation by effectively assigning higher confidence levels at each instant of time to those detector elements that contain significant signal energy and suppressing those that do not. A simpler suboptimum structure that replaces the continuous weighting function of the optimal receiver by a hard decision on the selection of the signal detector elements also is described and evaluated. Approximations and bounds to the error probability are derived and compared with the exact calculations and receiver simulation results. It is shown that, for photon-counting receivers observing Poisson-distributed signals, performance improvements of approximately 5 dB can be obtained over conventional single-detector photon-counting receivers, when operating in high background environments.

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.

Total variation approach for adaptive nonuniformity correction in focal-plane arrays.

PubMed

Vera, Esteban; Meza, Pablo; Torres, Sergio

2011-01-15

In this Letter we propose an adaptive scene-based nonuniformity correction method for fixed-pattern noise removal in imaging arrays. It is based on the minimization of the total variation of the estimated irradiance, and the resulting function is optimized by an isotropic total variation approach making use of an alternating minimization strategy. The proposed method provides enhanced results when applied to a diverse set of real IR imagery, accurately estimating the nonunifomity parameters of each detector in the focal-plane array at a fast convergence rate, while also forming fewer ghosting artifacts.

Adaptive Arrays for Weak Interfering Signals: An Experimental System. M.S. Thesis

NASA Technical Reports Server (NTRS)

Ward, James

1987-01-01

An experimental adaptive antenna system was implemented to study the performance of adaptive arrays in the presence of weak interfering signals. It is a sidelobe canceler with two auxiliary elements. Modified feedback loops, which decorrelate the noise components of the two inputs to the loop correlators, control the array weights. Digital processing is used for algorithm implementation and performance evaluation. The results show that the system can suppress interfering signals which are 0 to 10 dB below the thermal noise level in the main channel by 20 to 30 dB. When the desired signal is strong in the auxiliary elements the amount of interference suppression decreases. The amount of degradation depends on the number of interfering signals incident on the communication system. A modified steering vector which overcomes this problem is proposed.

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.

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.

Lightweight fiber optic microphones and accelerometers

NASA Astrophysics Data System (ADS)

Bucaro, J. A.; Lagakos, N.

2001-06-01

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

Analysis of Modified SMI Method for Adaptive Array Weight Control. M.S. Thesis

NASA Technical Reports Server (NTRS)

Dilsavor, Ronald Louis

1989-01-01

An adaptive array is used to receive a desired signal in the presence of weak interference signals which need to be suppressed. A modified sample matrix inversion (SMI) algorithm controls the array weights. The modification leads to increased interference suppression by subtracting a fraction of the noise power from the diagonal elements of the covariance matrix. The modified algorithm maximizes an intuitive power ratio criterion. The expected values and variances of the array weights, output powers, and power ratios as functions of the fraction and the number of snapshots are found and compared to computer simulation and real experimental array performance. Reduced-rank covariance approximations and errors in the estimated covariance are also described.

Graphene electrostatic microphone and ultrasonic radio

PubMed Central

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

2015-01-01

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

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…

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.

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.

Experimental Demonstration of Adaptive Infrared Multispectral Imaging using Plasmonic Filter Array.

PubMed

Jang, Woo-Yong; Ku, Zahyun; Jeon, Jiyeon; Kim, Jun Oh; Lee, Sang Jun; Park, James; Noyola, Michael J; Urbas, Augustine

2016-10-10

In our previous theoretical study, we performed target detection using a plasmonic sensor array incorporating the data-processing technique termed "algorithmic spectrometry". We achieved the reconstruction of a target spectrum by extracting intensity at multiple wavelengths with high resolution from the image data obtained from the plasmonic array. The ultimate goal is to develop a full-scale focal plane array with a plasmonic opto-coupler in order to move towards the next generation of versatile infrared cameras. To this end, and as an intermediate step, this paper reports the experimental demonstration of adaptive multispectral imagery using fabricated plasmonic spectral filter arrays and proposed target detection scenarios. Each plasmonic filter was designed using periodic circular holes perforated through a gold layer, and an enhanced target detection strategy was proposed to refine the original spectrometry concept for spatial and spectral computation of the data measured from the plasmonic array. Both the spectrum of blackbody radiation and a metal ring object at multiple wavelengths were successfully reconstructed using the weighted superposition of plasmonic output images as specified in the proposed detection strategy. In addition, plasmonic filter arrays were theoretically tested on a target at extremely high temperature as a challenging scenario for the detection scheme.

Experimental Demonstration of Adaptive Infrared Multispectral Imaging using Plasmonic Filter Array

PubMed Central

Jang, Woo-Yong; Ku, Zahyun; Jeon, Jiyeon; Kim, Jun Oh; Lee, Sang Jun; Park, James; Noyola, Michael J.; Urbas, Augustine

2016-01-01

In our previous theoretical study, we performed target detection using a plasmonic sensor array incorporating the data-processing technique termed “algorithmic spectrometry”. We achieved the reconstruction of a target spectrum by extracting intensity at multiple wavelengths with high resolution from the image data obtained from the plasmonic array. The ultimate goal is to develop a full-scale focal plane array with a plasmonic opto-coupler in order to move towards the next generation of versatile infrared cameras. To this end, and as an intermediate step, this paper reports the experimental demonstration of adaptive multispectral imagery using fabricated plasmonic spectral filter arrays and proposed target detection scenarios. Each plasmonic filter was designed using periodic circular holes perforated through a gold layer, and an enhanced target detection strategy was proposed to refine the original spectrometry concept for spatial and spectral computation of the data measured from the plasmonic array. Both the spectrum of blackbody radiation and a metal ring object at multiple wavelengths were successfully reconstructed using the weighted superposition of plasmonic output images as specified in the proposed detection strategy. In addition, plasmonic filter arrays were theoretically tested on a target at extremely high temperature as a challenging scenario for the detection scheme. PMID:27721506

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.

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

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.

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

Local motion adaptation enhances the representation of spatial structure at EMD arrays

PubMed Central

Lindemann, Jens P.; Egelhaaf, Martin

2017-01-01

Neuronal representation and extraction of spatial information are essential for behavioral control. For flying insects, a plausible way to gain spatial information is to exploit distance-dependent optic flow that is generated during translational self-motion. Optic flow is computed by arrays of local motion detectors retinotopically arranged in the second neuropile layer of the insect visual system. These motion detectors have adaptive response characteristics, i.e. their responses to motion with a constant or only slowly changing velocity decrease, while their sensitivity to rapid velocity changes is maintained or even increases. We analyzed by a modeling approach how motion adaptation affects signal representation at the output of arrays of motion detectors during simulated flight in artificial and natural 3D environments. We focused on translational flight, because spatial information is only contained in the optic flow induced by translational locomotion. Indeed, flies, bees and other insects segregate their flight into relatively long intersaccadic translational flight sections interspersed with brief and rapid saccadic turns, presumably to maximize periods of translation (80% of the flight). With a novel adaptive model of the insect visual motion pathway we could show that the motion detector responses to background structures of cluttered environments are largely attenuated as a consequence of motion adaptation, while responses to foreground objects stay constant or even increase. This conclusion even holds under the dynamic flight conditions of insects. PMID:29281631

Laser microphone

DOEpatents

Veligdan, James T.

2000-11-14

A microphone for detecting sound pressure waves includes a laser resonator having a laser gain material aligned coaxially between a pair of first and second mirrors for producing a laser beam. A reference cell is disposed between the laser material and one of the mirrors for transmitting a reference portion of the laser beam between the mirrors. A sensing cell is disposed between the laser material and one of the mirrors, and is laterally displaced from the reference cell for transmitting a signal portion of the laser beam, with the sensing cell being open for receiving the sound waves. A photodetector is disposed in optical communication with the first mirror for receiving the laser beam, and produces an acoustic signal therefrom for the sound waves.

Optimization of multiple turbine arrays in a channel with tidally reversing flow by numerical modelling with adaptive mesh.

PubMed

Divett, T; Vennell, R; Stevens, C

2013-02-28

At tidal energy sites, large arrays of hundreds of turbines will be required to generate economically significant amounts of energy. Owing to wake effects within the array, the placement of turbines within will be vital to capturing the maximum energy from the resource. This study presents preliminary results using Gerris, an adaptive mesh flow solver, to investigate the flow through four different arrays of 15 turbines each. The goal is to optimize the position of turbines within an array in an idealized channel. The turbines are represented as areas of increased bottom friction in an adaptive mesh model so that the flow and power capture in tidally reversing flow through large arrays can be studied. The effect of oscillating tides is studied, with interesting dynamics generated as the tidal current reverses direction, forcing turbulent flow through the array. The energy removed from the flow by each of the four arrays is compared over a tidal cycle. A staggered array is found to extract 54 per cent more energy than a non-staggered array. Furthermore, an array positioned to one side of the channel is found to remove a similar amount of energy compared with an array in the centre of the channel.

77 FR 64446 - Wireless Microphones Proceeding

Federal Register 2010, 2011, 2012, 2013, 2014

2012-10-22

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

Experiment on a three-beam adaptive array for EHF frequency-hopped signals using a fast algorithm, phase-D

NASA Astrophysics Data System (ADS)

Yen, J. L.; Kremer, P.; Amin, N.; Fung, J.

1989-05-01

The Department of National Defence (Canada) has been conducting studies into multi-beam adaptive arrays for extremely high frequency (EHF) frequency hopped signals. A three-beam 43 GHz adaptive antenna and a beam control processor is under development. An interactive software package for the operation of the array, capable of applying different control algorithms is being written. A maximum signal to jammer plus noise ratio (SJNR) was found to provide superior performance in preventing degradation of user signals in the presence of nearby jammers. A new fast algorithm using a modified conjugate gradient approach was found to be a very efficient way to implement anti-jamming arrays based on maximum SJNR criterion. The present study was intended to refine and simplify this algorithm and to implement the algorithm on an experimental array for real-time evaluation of anti-jamming performance. A three-beam adaptive array was used. A simulation package was used in the evaluation of multi-beam systems using more than three beams and different user-jammer scenarios. An attempt to further reduce the computation burden through continued analysis of maximum SJNR met with limited success. A method to acquire and track an incoming laser beam is proposed.

Feasible pickup from intact ossicular chain with floating piezoelectric microphone.

PubMed

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

2012-02-22

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

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.

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

PubMed

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

2016-12-03

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

Rocket Motor Microphone Investigation

NASA Technical Reports Server (NTRS)

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

2010-01-01

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

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

PubMed

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

2017-11-01

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

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.

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)

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.

[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.

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.

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

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.

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.

Weight Vector Fluctuations in Adaptive Antenna Arrays Tuned Using the Least-Mean-Square Error Algorithm with Quadratic Constraint

NASA Astrophysics Data System (ADS)

Zimina, S. V.

2015-06-01

We present the results of statistical analysis of an adaptive antenna array tuned using the least-mean-square error algorithm with quadratic constraint on the useful-signal amplification with allowance for the weight-coefficient fluctuations. Using the perturbation theory, the expressions for the correlation function and power of the output signal of the adaptive antenna array, as well as the formula for the weight-vector covariance matrix are obtained in the first approximation. The fluctuations are shown to lead to the signal distortions at the antenna-array output. The weight-coefficient fluctuations result in the appearance of additional terms in the statistical characteristics of the antenna array. It is also shown that the weight-vector fluctuations are isotropic, i.e., identical in all directions of the weight-coefficient space.

Experiment on a three-beam adaptive array for EHF frequency-hopped signals using a fast algorithm, phase E

NASA Astrophysics Data System (ADS)

Yen, J. L.; Kremer, P.; Fung, J.

1990-05-01

The Department of National Defence (Canada) has been conducting studies into multi-beam adaptive arrays for extremely high frequency (EHF) frequency hopped signals. A three-beam 43 GHz adaptive antenna and a beam control processor is under development. An interactive software package for the operation of the array, capable of applying different control algorithms is being written. A maximum signal to jammer plus noise ratio (SJNR) has been found to provide superior performance in preventing degradation of user signals in the presence of nearby jammers. A new fast algorithm using a modified conjugate gradient approach has been found to be a very efficient way to implement anti-jamming arrays based on maximum SJNR criterion. The present study was intended to refine and simplify this algorithm and to implement the algorithm on an experimental array for real-time evaluation of anti-jamming performance. A three-beam adaptive array was used. A simulation package was used in the evaluation of multi-beam systems using more than three beams and different user-jammer scenarios. An attempt to further reduce the computation burden through further analysis of maximum SJNR met with limited success. The investigation of a new angle detector for spatial tracking in heterodyne laser space communications was completed.

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 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

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.

An SDR-Based Real-Time Testbed for GNSS Adaptive Array Anti-Jamming Algorithms Accelerated by GPU.

PubMed

Xu, Hailong; Cui, Xiaowei; Lu, Mingquan

2016-03-11

Nowadays, software-defined radio (SDR) has become a common approach to evaluate new algorithms. However, in the field of Global Navigation Satellite System (GNSS) adaptive array anti-jamming, previous work has been limited due to the high computational power demanded by adaptive algorithms, and often lack flexibility and configurability. In this paper, the design and implementation of an SDR-based real-time testbed for GNSS adaptive array anti-jamming accelerated by a Graphics Processing Unit (GPU) are documented. This testbed highlights itself as a feature-rich and extendible platform with great flexibility and configurability, as well as high computational performance. Both Space-Time Adaptive Processing (STAP) and Space-Frequency Adaptive Processing (SFAP) are implemented with a wide range of parameters. Raw data from as many as eight antenna elements can be processed in real-time in either an adaptive nulling or beamforming mode. To fully take advantage of the parallelism resource provided by the GPU, a batched method in programming is proposed. Tests and experiments are conducted to evaluate both the computational and anti-jamming performance. This platform can be used for research and prototyping, as well as a real product in certain applications.

An SDR-Based Real-Time Testbed for GNSS Adaptive Array Anti-Jamming Algorithms Accelerated by GPU

PubMed Central

Xu, Hailong; Cui, Xiaowei; Lu, Mingquan

2016-01-01

Nowadays, software-defined radio (SDR) has become a common approach to evaluate new algorithms. However, in the field of Global Navigation Satellite System (GNSS) adaptive array anti-jamming, previous work has been limited due to the high computational power demanded by adaptive algorithms, and often lack flexibility and configurability. In this paper, the design and implementation of an SDR-based real-time testbed for GNSS adaptive array anti-jamming accelerated by a Graphics Processing Unit (GPU) are documented. This testbed highlights itself as a feature-rich and extendible platform with great flexibility and configurability, as well as high computational performance. Both Space-Time Adaptive Processing (STAP) and Space-Frequency Adaptive Processing (SFAP) are implemented with a wide range of parameters. Raw data from as many as eight antenna elements can be processed in real-time in either an adaptive nulling or beamforming mode. To fully take advantage of the parallelism resource provided by the GPU, a batched method in programming is proposed. Tests and experiments are conducted to evaluate both the computational and anti-jamming performance. This platform can be used for research and prototyping, as well as a real product in certain applications. PMID:26978363

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.

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.

Target-in-the-loop high-power adaptive phase-locked fiber laser array using single-frequency dithering technique

NASA Astrophysics Data System (ADS)

Tao, R.; Ma, Y.; Si, L.; Dong, X.; Zhou, P.; Liu, Z.

2011-11-01

We present a theoretical and experimental study of a target-in-the-loop (TIL) high-power adaptive phase-locked fiber laser array. The system configuration of the TIL adaptive phase-locked fiber laser array is introduced, and the fundamental theory for TIL based on the single-dithering technique is deduced for the first time. Two 10-W-level high-power fiber amplifiers are set up and adaptive phase locking of the two fiber amplifiers is accomplished successfully by implementing a single-dithering algorithm on a signal processor. The experimental results demonstrate that the optical phase noise for each beam channel can be effectively compensated by the TIL adaptive optics system under high-power applications and the fringe contrast on a remotely located extended target is advanced from 12% to 74% for the two 10-W-level fiber amplifiers.

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.

Dynamic experiment design regularization approach to adaptive imaging with array radar/SAR sensor systems.

PubMed

Shkvarko, Yuriy; Tuxpan, José; Santos, Stewart

2011-01-01

We consider a problem of high-resolution array radar/SAR imaging formalized in terms of a nonlinear ill-posed inverse problem of nonparametric estimation of the power spatial spectrum pattern (SSP) of the random wavefield scattered from a remotely sensed scene observed through a kernel signal formation operator and contaminated with random Gaussian noise. First, the Sobolev-type solution space is constructed to specify the class of consistent kernel SSP estimators with the reproducing kernel structures adapted to the metrics in such the solution space. Next, the "model-free" variational analysis (VA)-based image enhancement approach and the "model-based" descriptive experiment design (DEED) regularization paradigm are unified into a new dynamic experiment design (DYED) regularization framework. Application of the proposed DYED framework to the adaptive array radar/SAR imaging problem leads to a class of two-level (DEED-VA) regularized SSP reconstruction techniques that aggregate the kernel adaptive anisotropic windowing with the projections onto convex sets to enforce the consistency and robustness of the overall iterative SSP estimators. We also show how the proposed DYED regularization method may be considered as a generalization of the MVDR, APES and other high-resolution nonparametric adaptive radar sensing techniques. A family of the DYED-related algorithms is constructed and their effectiveness is finally illustrated via numerical simulations.

Adaptive microlens array based on electrically charged polyvinyl chloride/dibutyl phthalate gel

NASA Astrophysics Data System (ADS)

Xu, Miao; Ren, Hongwen

2016-09-01

We prepared an adaptive microlens array (MLA) using a polyvinyl chloride/dibutyl phthalate gel and an indium-tin-oxide (ITO) glass substrate. The gel forms a membrane on the glass substrate and the ITO electrode has a ring array pattern. When the membrane is electrically charged by a DC voltage, the surface of the membrane above each circular electrode in the ring array can be deformed with a convex shape. As a result, the membrane functions as an MLA. By applying a voltage from 20 to ˜65 V to the electrode, the focal length of each microlens can be tuned from 300 to ˜160 μm. The dynamic response time can by reduced largely by changing the polarity of the DC voltage. Due to the advantages of optical isotropy, compact structure, and good stability, our MLA has potential applications in imaging, biometrics, and electronic displays.

An experimental adaptive array to suppress weak interfering signals

NASA Technical Reports Server (NTRS)

Walton, Eric K.; Gupta, Inder J.; Ksienski, Aharon A.; Ward, James

1988-01-01

An experimental adaptive antenna system to suppress weak interfering signals is described. It is a sidelobe canceller with two auxiliary elements. Modified feedback loops are used to control the array weights. The received signals are simulated in hardware for parameter control. Digital processing is used for algorithm implementation and performance evaluation. The experimental results are presented. They show that interfering signals as much as 10 dB below the thermal noise level in the main channel are suppressed by 20-30 dB. Such a system has potential application in suppressing the interference encountered in direct broadcast satellite communication systems.

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

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

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

NASA Technical Reports Server (NTRS)

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

1978-01-01

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

Analysis and experimental demonstration of conformal adaptive phase-locked fiber array for laser communications and beam projection applications

NASA Astrophysics Data System (ADS)

Liu, Ling

The primary goal of this research is the analysis, development, and experimental demonstration of an adaptive phase-locked fiber array system for free-space optical communications and laser beam projection applications. To our knowledge, the developed adaptive phase-locked system composed of three fiber collimators (subapertures) with tip-tilt wavefront phase control at each subaperture represents the first reported fiber array system that implements both phase-locking control and adaptive wavefront tip-tilt control capabilities. This research has also resulted in the following innovations: (a) The first experimental demonstration of a phase-locked fiber array with tip-tilt wave-front aberration compensation at each fiber collimator; (b) Development and demonstration of the fastest currently reported stochastic parallel gradient descent (SPGD) system capable of operation at 180,000 iterations per second; (c) The first experimental demonstration of a laser communication link based on a phase-locked fiber array; (d) The first successful experimental demonstration of turbulence and jitter-induced phase distortion compensation in a phase-locked fiber array optical system; (e) The first demonstration of laser beam projection onto an extended target with a randomly rough surface using a conformal adaptive fiber array system. Fiber array optical systems, the subject of this study, can overcome some of the draw-backs of conventional monolithic large-aperture transmitter/receiver optical systems that are usually heavy, bulky, and expensive. The primary experimental challenges in the development of the adaptive phased-locked fiber-array included precise (<5 microrad) alignment of the fiber collimators and development of fast (100kHz-class) phase-locking and wavefront tip-tilt control systems. The precise alignment of the fiber collimator array is achieved through a specially developed initial coarse alignment tool based on high precision piezoelectric picomotors and a

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.

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.

A Simple Laser Microphone for Classroom Demonstration

ERIC Educational Resources Information Center

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

2006-01-01

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

An Improved Adaptive Received Beamforming for Nested Frequency Offset and Nested Array FDA-MIMO Radar.

PubMed

Cheng, Sibei; Zhang, Qingjun; Bian, Mingming; Hao, Xinhong

2018-02-08

For the conventional FDA-MIMO (frequency diversity array multiple-input-multiple-output) Radar with uniform frequency offset and uniform linear array, the DOFs (degrees of freedom) of the adaptive beamformer are limited by the number of elements. A better performance-for example, a better suppression for strong interferences and a more desirable trade-off between the main lobe and side lobe-can be achieved with a greater number of DOFs. In order to obtain larger DOFs, this paper researches the signal model of the FDA-MIMO Radar with nested frequency offset and nested array, then proposes an improved adaptive beamforming method that uses the augmented matrix instead of the covariance matrix to calculate the optimum weight vectors and can be used to improve the output performances of FDA-MIMO Radar with the same element number or reduce the element number while maintain the approximate output performances such as the received beampattern, the main lobe width, side lobe depths and the output SINR (signal-to-interference-noise ratio). The effectiveness of the proposed scheme is verified by simulations.

An Improved Adaptive Received Beamforming for Nested Frequency Offset and Nested Array FDA-MIMO Radar

PubMed Central

Cheng, Sibei; Zhang, Qingjun; Bian, Mingming; Hao, Xinhong

2018-01-01

For the conventional FDA-MIMO (frequency diversity array multiple-input-multiple-output) Radar with uniform frequency offset and uniform linear array, the DOFs (degrees of freedom) of the adaptive beamformer are limited by the number of elements. A better performance—for example, a better suppression for strong interferences and a more desirable trade-off between the main lobe and side lobe—can be achieved with a greater number of DOFs. In order to obtain larger DOFs, this paper researches the signal model of the FDA-MIMO Radar with nested frequency offset and nested array, then proposes an improved adaptive beamforming method that uses the augmented matrix instead of the covariance matrix to calculate the optimum weight vectors and can be used to improve the output performances of FDA-MIMO Radar with the same element number or reduce the element number while maintain the approximate output performances such as the received beampattern, the main lobe width, side lobe depths and the output SINR (signal-to-interference-noise ratio). The effectiveness of the proposed scheme is verified by simulations. PMID:29419814

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.

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.

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].

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.

Analysis of modified SMI method for adaptive array weight control

NASA Technical Reports Server (NTRS)

Dilsavor, R. L.; Moses, R. L.

1989-01-01

An adaptive array is applied to the problem of receiving a desired signal in the presence of weak interference signals which need to be suppressed. A modification, suggested by Gupta, of the sample matrix inversion (SMI) algorithm controls the array weights. In the modified SMI algorithm, interference suppression is increased by subtracting a fraction F of the noise power from the diagonal elements of the estimated covariance matrix. Given the true covariance matrix and the desired signal direction, the modified algorithm is shown to maximize a well-defined, intuitive output power ratio criterion. Expressions are derived for the expected value and variance of the array weights and output powers as a function of the fraction F and the number of snapshots used in the covariance matrix estimate. These expressions are compared with computer simulation and good agreement is found. A trade-off is found to exist between the desired level of interference suppression and the number of snapshots required in order to achieve that level with some certainty. The removal of noise eigenvectors from the covariance matrix inverse is also discussed with respect to this application. Finally, the type and severity of errors which occur in the covariance matrix estimate are characterized through simulation.

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.

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.

Dynamic Experiment Design Regularization Approach to Adaptive Imaging with Array Radar/SAR Sensor Systems

PubMed Central

Shkvarko, Yuriy; Tuxpan, José; Santos, Stewart

2011-01-01

We consider a problem of high-resolution array radar/SAR imaging formalized in terms of a nonlinear ill-posed inverse problem of nonparametric estimation of the power spatial spectrum pattern (SSP) of the random wavefield scattered from a remotely sensed scene observed through a kernel signal formation operator and contaminated with random Gaussian noise. First, the Sobolev-type solution space is constructed to specify the class of consistent kernel SSP estimators with the reproducing kernel structures adapted to the metrics in such the solution space. Next, the “model-free” variational analysis (VA)-based image enhancement approach and the “model-based” descriptive experiment design (DEED) regularization paradigm are unified into a new dynamic experiment design (DYED) regularization framework. Application of the proposed DYED framework to the adaptive array radar/SAR imaging problem leads to a class of two-level (DEED-VA) regularized SSP reconstruction techniques that aggregate the kernel adaptive anisotropic windowing with the projections onto convex sets to enforce the consistency and robustness of the overall iterative SSP estimators. We also show how the proposed DYED regularization method may be considered as a generalization of the MVDR, APES and other high-resolution nonparametric adaptive radar sensing techniques. A family of the DYED-related algorithms is constructed and their effectiveness is finally illustrated via numerical simulations. PMID:22163859

Human Action Recognition Using Wireless Wearable In-Ear Microphone

NASA Astrophysics Data System (ADS)

Nishimura, Jun; Kuroda, Tadahiro

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

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.

Laser beam projection with adaptive array of fiber collimators. II. Analysis of atmospheric compensation efficiency.

PubMed

Lachinova, Svetlana L; Vorontsov, Mikhail A

2008-08-01

We analyze the potential efficiency of laser beam projection onto a remote object in atmosphere with incoherent and coherent phase-locked conformal-beam director systems composed of an adaptive array of fiber collimators. Adaptive optics compensation of turbulence-induced phase aberrations in these systems is performed at each fiber collimator. Our analysis is based on a derived expression for the atmospheric-averaged value of the mean square residual phase error as well as direct numerical simulations. Operation of both conformal-beam projection systems is compared for various adaptive system configurations characterized by the number of fiber collimators, the adaptive compensation resolution, and atmospheric turbulence conditions.

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.

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

System Measures Thermal Noise In A Microphone

NASA Technical Reports Server (NTRS)

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

1994-01-01

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

Adaptive optics for array telescopes using piston-and-tilt wave-front sensing

NASA Technical Reports Server (NTRS)

Wizinowich, P.; Mcleod, B.; Lloyd-Yhart, M.; Angel, J. R. P.; Colucci, D.; Dekany, R.; Mccarthy, D.; Wittman, D.; Scott-Fleming, I.

1992-01-01

A near-infrared adaptive optics system operating at about 50 Hz has been used to control phase errors adaptively between two mirrors of the Multiple Mirror Telescope by stabilizing the position of the interference fringe in the combined unresolved far-field image. The resultant integrated images have angular resolutions of better than 0.1 arcsec and fringe contrasts of more than 0.6. Measurements of wave-front tilt have confirmed the wavelength independence of image motion. These results show that interferometric sensing of phase errors, when combined with a system for sensing the wave-front tilt of the individual telescopes, will provide a means of achieving a stable diffraction-limited focus with segmented telescopes or arrays of telescopes.

Parametric Investigation of Laser Doppler Microphones

NASA Astrophysics Data System (ADS)

Daoud, M.; Naguib, A.

2002-11-01

The concept of a Laser Doppler Microphone (LDM) is based on utilizing the Doppler frequency shift of a focused laser beam to measure the unsteady velocity of the center point of a flexible polymer diaphragm that is mounted on top of a hole and subjected to the unsteady pressure. Time integration of the velocity signal yields a time series of the diaphragm displacement, which can be converted to pressure from knowledge of the sensor's deflection sensitivity. In our APS/DFD presentation last year, the stringent frequency resolution requirement of these new sensors and methods to meet this requirement were discussed. Here, the dependence of the sensor characteristics (sensitivity, bandwidth, and noise floor) on various significant parameters is investigated in detail by calibrating the sensor in a plane wave tube in the frequency range of 50 - 5000 Hz. Parameters investigated include sensor diaphragm material and thickness, sensor size, damping of the diaphragm motion and laser beam spot size. The results shed light on the operating limits of the new sensor and demonstrate its ability to conduct high-spatial-resolution measurements in typical high-Reynolds-number test facilities. Moreover, calibrated LDM sensors were used to conduct measurements in a separating/reattaching flow and the results are compared to classical electret-type microphones with a similar sensing diameter.

Analysis and design of a high power laser adaptive phased array transmitter

NASA Technical Reports Server (NTRS)

Mevers, G. E.; Soohoo, J. F.; Winocur, J.; Massie, N. A.; Southwell, W. H.; Brandewie, R. A.; Hayes, C. L.

1977-01-01

The feasibility of delivering substantial quantities of optical power to a satellite in low earth orbit from a ground based high energy laser (HEL) coupled to an adaptive antenna was investigated. Diffraction effects, atmospheric transmission efficiency, adaptive compensation for atmospheric turbulence effects, including the servo bandwidth requirements for this correction, and the adaptive compensation for thermal blooming were examined. To evaluate possible HEL sources, atmospheric investigations were performed for the CO2, (C-12)(O-18)2 isotope, CO and DF wavelengths using output antenna locations of both sea level and mountain top. Results indicate that both excellent atmospheric and adaption efficiency can be obtained for mountain top operation with a micron isotope laser operating at 9.1 um, or a CO laser operating single line (P10) at about 5.0 (C-12)(O-18)2um, which was a close second in the evaluation. Four adaptive power transmitter system concepts were generated and evaluated, based on overall system efficiency, reliability, size and weight, advanced technology requirements and potential cost. A multiple source phased array was selected for detailed conceptual design. The system uses a unique adaption technique of phase locking independent laser oscillators which allows it to be both relatively inexpensive and most reliable with a predicted overall power transfer efficiency of 53%.

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

Adaptive Wiener filter super-resolution of color filter array images.

PubMed

Karch, Barry K; Hardie, Russell C

2013-08-12

Digital color cameras using a single detector array with a Bayer color filter array (CFA) require interpolation or demosaicing to estimate missing color information and provide full-color images. However, demosaicing does not specifically address fundamental undersampling and aliasing inherent in typical camera designs. Fast non-uniform interpolation based super-resolution (SR) is an attractive approach to reduce or eliminate aliasing and its relatively low computational load is amenable to real-time applications. The adaptive Wiener filter (AWF) SR algorithm was initially developed for grayscale imaging and has not previously been applied to color SR demosaicing. Here, we develop a novel fast SR method for CFA cameras that is based on the AWF SR algorithm and uses global channel-to-channel statistical models. We apply this new method as a stand-alone algorithm and also as an initialization image for a variational SR algorithm. This paper presents the theoretical development of the color AWF SR approach and applies it in performance comparisons to other SR techniques for both simulated and real data.

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

A Phase Correction Technique Based on Spatial Movements of Antennas in Real-Time (S.M.A.R.T.) for Designing Self-Adapting Conformal Array Antennas

NASA Astrophysics Data System (ADS)

Roy, Sayan

This research presents a real-time adaptive phase correction technique for flexible phased array antennas on conformal surfaces of variable shapes. Previously reported pattern correctional methods for flexible phased array antennas require prior knowledge on the possible non-planar shapes in which the array may adapt for conformal applications. For the first time, this initial requirement of shape curvature knowledge is no longer needed and the instantaneous information on the relative location of array elements is used here for developing a geometrical model based on a set of Bezier curves. Specifically, by using an array of inclinometer sensors and an adaptive phase-correctional algorithm, it has been shown that the proposed geometrical model can successfully predict different conformal orientations of a 1-by-4 antenna array in real-time without the requirement of knowing the shape-changing characteristics of the surface the array is attached upon. Moreover, the phase correction technique is validated by determining the field patterns and broadside gain of the 1-by-4 antenna array on four different conformal surfaces with multiple points of curvatures. Throughout this work, measurements are shown to agree with the analytical solutions and full-wave simulations.

Performance benefits of adaptive, multimicrophone, interference-canceling systems in everyday environments

NASA Astrophysics Data System (ADS)

Desloge, Joseph G.; Zimmer, Martin J.; Zurek, Patrick M.

2004-05-01

Adaptive multimicrophone systems are currently used for a variety of noise-cancellation applications (such as hearing aids) to preserve signals arriving from a particular (target) direction while canceling other (jammer) signals in the environment. Although the performance of these systems is known to degrade with increasing reverberation, there are few measurements of adaptive performance in everyday reverberant environments. In this study, adaptive performance was compared to that of a simple, nonadaptive cardioid microphone to determine a measure of adaptive benefit. Both systems used recordings (at an Fs of 22050 Hz) from the same two omnidirectional microphones, which were separated by 1 cm. Four classes of environment were considered: outdoors, household, parking garage, and public establishment. Sources were either environmental noises (e.g., household appliances, restaurant noise) or a controlled noise source. In all situations, no target was present (i.e., all signals were jammers) to obtain maximal jammer cancellation. Adaptive processing was based upon the Griffiths-Jim generalized sidelobe canceller using filter lengths up to 400 points. Average intelligibility-weighted adaptive benefit levels at a source distance of 1 m were, at most, 1.5 dB for public establishments, 2 dB for household rooms and the parking garage, and 3 dB outdoors. [Work supported by NIOSH.

An Electromechanical Model for the Cochlear Microphonic

NASA Astrophysics Data System (ADS)

Teal, Paul D.; Lineton, Ben; Elliott, Stephen J.

2011-11-01

The first of the many electrical signals generated in the ear, nerves and brain as a response to a sound incident on the ear is the cochlear microphonic (CM). The CM is generated by the hair cells of the cochlea, primarily the outer hairs cells. The potentials of this signal are a nonlinear filtered version of the acoustic pressure at the tympanic membrane. The CM signal has been used very little in recent years for clinical audiology and audiological research. This is because of uncertainty in interpreting the CM signal as a diagnostic measure, and also because of the difficulty of obtaining the signal, which has usually required the use of a transtympanic electrode. There are however, several potential clinical and research applications for acquisition of the CM. To promote understanding of the CM, and potential clinical application, a model is presented which can account for the generation of the cochlear microphonic signal. The model incorporates micro-mechanical and macro-mechanical aspects of previously published models of the basilar membrane and reticular lamina, as well as cochlear fluid mechanics, piezoelectric activity and capacitance of the outer hair cells. It also models the electrical coupling of signals along the scalae.

Guidelines for Selecting Microphones for Human Voice Production Research

ERIC Educational Resources Information Center

Svec, Jan G.; Granqvist, Svante

2010-01-01

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

Wavefront sensing and adaptive control in phased array of fiber collimators

NASA Astrophysics Data System (ADS)

Lachinova, Svetlana L.; Vorontsov, Mikhail A.

2011-03-01

A new wavefront control approach for mitigation of atmospheric turbulence-induced wavefront phase aberrations in coherent fiber-array-based laser beam projection systems is introduced and analyzed. This approach is based on integration of wavefront sensing capabilities directly into the fiber-array transmitter aperture. In the coherent fiber array considered, we assume that each fiber collimator (subaperture) of the array is capable of precompensation of local (onsubaperture) wavefront phase tip and tilt aberrations using controllable rapid displacement of the tip of the delivery fiber at the collimating lens focal plane. In the technique proposed, this tip and tilt phase aberration control is based on maximization of the optical power received through the same fiber collimator using the stochastic parallel gradient descent (SPGD) technique. The coordinates of the fiber tip after the local tip and tilt aberrations are mitigated correspond to the coordinates of the focal-spot centroid of the optical wave backscattered off the target. Similar to a conventional Shack-Hartmann wavefront sensor, phase function over the entire fiber-array aperture can then be retrieved using the coordinates obtained. The piston phases that are required for coherent combining (phase locking) of the outgoing beams at the target plane can be further calculated from the reconstructed wavefront phase. Results of analysis and numerical simulations are presented. Performance of adaptive precompensation of phase aberrations in this laser beam projection system type is compared for various system configurations characterized by the number of fiber collimators and atmospheric turbulence conditions. The wavefront control concept presented can be effectively applied for long-range laser beam projection scenarios for which the time delay related with the double-pass laser beam propagation to the target and back is compared or even exceeds the characteristic time of the atmospheric turbulence change

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

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

NASA Technical Reports Server (NTRS)

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

1992-01-01

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

High temperature sensor/microphone development for active noise control

NASA Technical Reports Server (NTRS)

Shrout, Thomas R.

1993-01-01

1000 C. Concurrent with the materials study was an effort to define issues involved in the development of a microphone capable of operation at temperatures up to 1000 C; important since microphones capable of operation above 260 C are not generally available. The distinguishing feature of a microphone is its diaphragm which receives sound from the atmosphere: whereas, most other acoustic sensors receive sound through the solid structure on which they are installed. In order to gain an understanding of the potential problems involved in designing and testing a high temperature microphone, a prototype was constructed using a commercially available lithium niobate piezoelectric element in a stainless steel structure. The prototype showed excellent frequency response at room temperature, and responded to acoustic stimulation at 670 C, above which temperature the voltage output rapidly diminished because of decreased resistivity in the element. Samples of the PLS material were also evaluated in a simulated microphone configuration, but their voltage output was found to be a few mV compared to the 10 output of the prototype.

A Four-Phase Modulation System for Use with an Adaptive Array.

DTIC Science & Technology

1982-07-01

MODULATION SYSTEM FORIteia epr * ~USE WITH AN ADAPTIVE ARRAY *P RIGOG EOTM~E _____________________________________ ESL 711679-5 7s AUTHOeO~) 9 . CONTRACT r0...OUSOLE1T6 UNCLASSIFIED SECURITY CLASSIFICATION OF THIS P040E (when Doe I91 r2 UNCLASSIFIED 8ncumV CL"M,ICAnIo, o TP , ImS 8... ., 9 fte-H - LMS...nterval has a duration of : Tb seconds. a(t) is a pseudonotse code, i.e., a maximum length " lInear shift register sequence [ 9 ]. The code symbol interval

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.

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.

Reproducibility of Dual-Microphone Voice Range Profile Equipment

ERIC Educational Resources Information Center

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

2017-01-01

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

Adaptive antenna arrays for satellite communications: Design and testing

NASA Technical Reports Server (NTRS)

Gupta, I. J.; Swarner, W. G.; Walton, E. K.

1985-01-01

When two separate antennas are used with each feedback loop to decorrelate noise, the antennas should be located such that the phase of the interfering signal in the two antennas is the same while the noise in them is uncorrelated. Thus, the antenna patterns and spatial distribution of the auxiliary antennas are quite important and should be carefully selected. The selection and spatial distribution of auxiliary elements is discussed when the main antenna is a center fed reflector antenna. It is shown that offset feeds of the reflector antenna can be used as auxiliary elements of an adaptive array to suppress weak interfering signals. An experimental system is designed to verify the theoretical analysis. The details of the experimental systems are presented.

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

DOEpatents

Kupperman, David S.; Spevak, Lev

1994-01-01

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

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

NASA Technical Reports Server (NTRS)

Fields, Richard S., Jr.

1995-01-01

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

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

PubMed

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

2007-10-01

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

Improvement of resolution in full-view linear-array photoacoustic computed tomography using a novel adaptive weighting method

NASA Astrophysics Data System (ADS)

Omidi, Parsa; Diop, Mamadou; Carson, Jeffrey; Nasiriavanaki, Mohammadreza

2017-03-01

Linear-array-based photoacoustic computed tomography is a popular methodology for deep and high resolution imaging. However, issues such as phase aberration, side-lobe effects, and propagation limitations deteriorate the resolution. The effect of phase aberration due to acoustic attenuation and constant assumption of the speed of sound (SoS) can be reduced by applying an adaptive weighting method such as the coherence factor (CF). Utilizing an adaptive beamforming algorithm such as the minimum variance (MV) can improve the resolution at the focal point by eliminating the side-lobes. Moreover, invisibility of directional objects emitting parallel to the detection plane, such as vessels and other absorbing structures stretched in the direction perpendicular to the detection plane can degrade resolution. In this study, we propose a full-view array level weighting algorithm in which different weighs are assigned to different positions of the linear array based on an orientation algorithm which uses the histogram of oriented gradient (HOG). Simulation results obtained from a synthetic phantom show the superior performance of the proposed method over the existing reconstruction methods.

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

PubMed Central

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

2015-01-01

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

The effectiveness of the directional microphone in the Oticon Medical Ponto Pro in participants with unilateral sensorineural hearing loss.

PubMed

Oeding, Kristi; Valente, Michael

2013-09-01

Current bone anchored hearing solutions (BAHSs) have incorporated automatic adaptive multichannel directional microphones (DMs). Previous fixed single-channel hypercardioid DMs in BAHSs have provided benefit in a diffuse listening environment, but little data are available on the performance of adaptive multichannel DMs in BAHSs for persons with unilateral sensorineural hearing loss (USNHL). The primary goal was to determine if statistically significant differences existed in the mean Reception Threshold for Sentences (RTS in dB) in diffuse uncorrelated restaurant noise between unaided, an omnidirectional microphone (OM), split DM (SDM), and full DM (FDM) in the Oticon Medical Ponto Pro. A second goal was to assess subjective benefit using the Abbreviated Profile of Hearing Aid Benefit (APHAB) comparing the Ponto Pro to the participant's current BAHS, and the Ponto Pro and participant's own BAHS to unaided. The third goal was to compare RTS data of the Ponto Pro to data from an identical study examining Cochlear Americas' Divino. A randomized repeated measures, single blind design was used to measure an RTS for each participant for unaided, OM, SDM, and FDM. Fifteen BAHS users with USNHL were recruited from Washington University in St. Louis and the surrounding area. The Ponto Pro was fit by measuring in-situ bone conduction thresholds and was worn for 4 wk. An RTS was obtained utilizing Hearing in Noise Test (HINT) sentences in uncorrelated restaurant noise from an eight loudspeaker array, and subjective benefit was determined utilizing the APHAB. Analysis of variance (ANOVA) was used to analyze the results of the Ponto Pro HINT and APHAB data, and comparisons between the Ponto Pro and previous Divino data. No statistically significant differences existed in mean RTS between unaided, the Ponto Pro's OM, SDM, or FDM (p = 0.10). The Ponto Pro provided statistically significant benefit for the Background Noise (BN) (p < 0.01) and Reverberation (RV) (p < 0

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.

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.

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...

Performance comparisons on spatial lattice algorithm and direct matrix inverse method with application to adaptive arrays processing

NASA Technical Reports Server (NTRS)

An, S. H.; Yao, K.

1986-01-01

Lattice algorithm has been employed in numerous adaptive filtering applications such as speech analysis/synthesis, noise canceling, spectral analysis, and channel equalization. In this paper the application to adaptive-array processing is discussed. The advantages are fast convergence rate as well as computational accuracy independent of the noise and interference conditions. The results produced by this technique are compared to those obtained by the direct matrix inverse method.

Electrode surface profile and the performance of condenser microphones.

PubMed

Fletcher, N H; Thwaites, S

2002-12-01

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

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

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.

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.

Adaptive and mobile ground sensor array.

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

Holzrichter, Michael Warren; O'Rourke, William T.; Zenner, Jennifer

The goal of this LDRD was to demonstrate the use of robotic vehicles for deploying and autonomously reconfiguring seismic and acoustic sensor arrays with high (centimeter) accuracy to obtain enhancement of our capability to locate and characterize remote targets. The capability to accurately place sensors and then retrieve and reconfigure them allows sensors to be placed in phased arrays in an initial monitoring configuration and then to be reconfigured in an array tuned to the specific frequencies and directions of the selected target. This report reviews the findings and accomplishments achieved during this three-year project. This project successfully demonstrated autonomousmore » deployment and retrieval of a payload package with an accuracy of a few centimeters using differential global positioning system (GPS) signals. It developed an autonomous, multisensor, temporally aligned, radio-frequency communication and signal processing capability, and an array optimization algorithm, which was implemented on a digital signal processor (DSP). Additionally, the project converted the existing single-threaded, monolithic robotic vehicle control code into a multi-threaded, modular control architecture that enhances the reuse of control code in future projects.« less

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

PubMed

Tan, Chee Wee; Miao, Jianmin

2009-06-01

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

Measurement of Gravitational Acceleration Using a Computer Microphone Port

ERIC Educational Resources Information Center

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

2012-01-01

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

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

NASA Astrophysics Data System (ADS)

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

2009-05-01

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

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

PubMed

Chung, King

2012-06-01

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

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

NASA Technical Reports Server (NTRS)

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

1993-01-01

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

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.

Virtual microphone sensing through vibro-acoustic modelling and Kalman filtering

NASA Astrophysics Data System (ADS)

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

2018-05-01

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

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.

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

PubMed Central

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

2012-01-01

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

Miniaturized Nanocomposite Piezoelectric Microphones for UAS Applications

DTIC Science & Technology

2012-10-22

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

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.

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

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

Adaptation of the Biolog Phenotype MicroArrayTM Technology to Profile the Obligate Anaerobe Geobacter metallireducens

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

Joyner, Dominique; Fortney, Julian; Chakraborty, Romy

2010-05-17

The Biolog OmniLog? Phenotype MicroArray (PM) plate technology was successfully adapted to generate a select phenotypic profile of the strict anaerobe Geobacter metallireducens (G.m.). The profile generated for G.m. provides insight into the chemical sensitivity of the organism as well as some of its metabolic capabilities when grown with a basal medium containing acetate and Fe(III). The PM technology was developed for aerobic organisms. The reduction of a tetrazolium dye by the test organism represents metabolic activity on the array which is detected and measured by the OmniLog(R) system. We have previously adapted the technology for the anaerobic sulfate reducingmore » bacterium Desulfovibrio vulgaris. In this work, we have taken the technology a step further by adapting it for the iron reducing obligate anaerobe Geobacter metallireducens. In an osmotic stress microarray it was determined that the organism has higher sensitivity to impermeable solutes 3-6percent KCl and 2-5percent NaNO3 that result in osmotic stress by osmosis to the cell than to permeable non-ionic solutes represented by 5-20percent ethylene glycol and 2-3percent urea. The osmotic stress microarray also includes an array of osmoprotectants and precursor molecules that were screened to identify substrates that would provide osmotic protection to NaCl stress. None of the substrates tested conferred resistance to elevated concentrations of salt. Verification studies in which G.m. was grown in defined medium amended with 100mM NaCl (MIC) and the common osmoprotectants betaine, glycine and proline supported the PM findings. Further verification was done by analysis of transcriptomic profiles of G.m. grown under 100mM NaCl stress that revealed up-regulation of genes related to degradation rather than accumulation of the above-mentioned osmoprotectants. The phenotypic profile, supported by additional analysis indicates that the accumulation of these osmoprotectants as a response to salt stress does

A New Kind of Laser Microphone for Photoacoustic Applications

DTIC Science & Technology

2008-12-01

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

49 CFR 325.73 - Microphone distance correction factors. 1

Code of Federal Regulations, 2013 CFR

2013-10-01

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

49 CFR 325.73 - Microphone distance correction factors. 1

Code of Federal Regulations, 2012 CFR

2012-10-01

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

49 CFR 325.73 - Microphone distance correction factors. 1

Code of Federal Regulations, 2014 CFR

2014-10-01

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

Improved neural network based scene-adaptive nonuniformity correction method for infrared focal plane arrays.

PubMed

Lai, Rui; Yang, Yin-tang; Zhou, Duan; Li, Yue-jin

2008-08-20

An improved scene-adaptive nonuniformity correction (NUC) algorithm for infrared focal plane arrays (IRFPAs) is proposed. This method simultaneously estimates the infrared detectors' parameters and eliminates the nonuniformity causing fixed pattern noise (FPN) by using a neural network (NN) approach. In the learning process of neuron parameter estimation, the traditional LMS algorithm is substituted with the newly presented variable step size (VSS) normalized least-mean square (NLMS) based adaptive filtering algorithm, which yields faster convergence, smaller misadjustment, and lower computational cost. In addition, a new NN structure is designed to estimate the desired target value, which promotes the calibration precision considerably. The proposed NUC method reaches high correction performance, which is validated by the experimental results quantitatively tested with a simulative testing sequence and a real infrared image sequence.

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.

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

PubMed

Green, Joshua J; Eigsti, Inge-Marie

2017-09-01

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

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

PubMed Central

Ziegelwanger, Harald; Majdak, Piotr; Kreuzer, Wolfgang

2015-01-01

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

Effects of Adaptive Antenna Arrays on Broadband Signals.

DTIC Science & Technology

1980-06-01

dimensional array geometry. The signal impinging on the antenna array elements is assumed to have originated from a point source in the far field , or...tg9 (4) The assumptions used to identify the far field region of an array also lead to an approximation for ti(6) . It is ti (0 ) x i sin(e) (5) c...implementing the open form transfer function and coefficients of Eqs (16) 53 .. ... ... .. . . .. . . .. ... .. . ..... . .... . . .. through (21). For a

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

PubMed

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

2006-01-01

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

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

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.

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

NASA Astrophysics Data System (ADS)

Dumoulin, Romain

Despite the fact that noise-induced hearing loss remains the number one occupational disease in developed countries, individual noise exposure levels are still rarely known and infrequently tracked. Indeed, efforts to standardize noise exposure levels present disadvantages such as costly instrumentation and difficulties associated with on site implementation. Given their advanced technical capabilities and widespread daily usage, mobile phones could be used to measure noise levels and make noise monitoring more accessible. However, the use of mobile phones for measuring noise exposure is currently limited due to the lack of formal procedures for their calibration and challenges regarding the measurement procedure. Our research investigated the calibration of mobile phone-based solutions for measuring noise exposure using a mobile phone's built-in microphones and wearable external microphones. The proposed calibration approach integrated corrections that took into account microphone placement error. The corrections were of two types: frequency-dependent, using a digital filter and noise level-dependent, based on the difference between the C-weighted noise level minus A-weighted noise level of the noise measured by the phone. The electro-acoustical limitations and measurement calibration procedure of the mobile phone were investigated. The study also sought to quantify the effect of noise exposure characteristics on the accuracy of calibrated mobile phone measurements. Measurements were carried out in reverberant and semi-anechoic chambers with several mobiles phone units of the same model, two types of external devices (an earpiece and a headset with an in-line microphone) and an acoustical test fixture (ATF). The proposed calibration approach significantly improved the accuracy of the noise level measurements in diffuse and free fields, with better results in the diffuse field and with ATF positions causing little or no acoustic shadowing. Several sources of errors

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.

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

PubMed

Kim, Hannah; Ricketts, Todd A

2013-01-01

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

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

NASA Technical Reports Server (NTRS)

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

2008-01-01

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

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.

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

ERIC Educational Resources Information Center

Kyriakou, Kyriaki; Fisher, Helene R.

2013-01-01

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

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.

SNP-array reveals genome-wide patterns of geographical and potential adaptive divergence across the natural range of Atlantic salmon (Salmo salar).

PubMed

Bourret, Vincent; Kent, Matthew P; Primmer, Craig R; Vasemägi, Anti; Karlsson, Sten; Hindar, Kjetil; McGinnity, Philip; Verspoor, Eric; Bernatchez, Louis; Lien, Sigbjørn

2013-02-01

Atlantic salmon (Salmo salar) is one of the most extensively studied fish species in the world due to its significance in aquaculture, fisheries and ongoing conservation efforts to protect declining populations. Yet, limited genomic resources have hampered our understanding of genetic architecture in the species and the genetic basis of adaptation to the wide range of natural and artificial environments it occupies. In this study, we describe the development of a medium-density Atlantic salmon single nucleotide polymorphism (SNP) array based on expressed sequence tags (ESTs) and genomic sequencing. The array was used in the most extensive assessment of population genetic structure performed to date in this species. A total of 6176 informative SNPs were successfully genotyped in 38 anadromous and freshwater wild populations distributed across the species natural range. Principal component analysis clearly differentiated European and North American populations, and within Europe, three major regional genetic groups were identified for the first time in a single analysis. We assessed the potential for the array to disentangle neutral and putative adaptive divergence of SNP allele frequencies across populations and among regional groups. In Europe, secondary contact zones were identified between major clusters where endogenous and exogenous barriers could be associated, rendering the interpretation of environmental influence on potentially adaptive divergence equivocal. A small number of markers highly divergent in allele frequencies (outliers) were observed between (multiple) freshwater and anadromous populations, between northern and southern latitudes, and when comparing Baltic populations to all others. We also discuss the potential future applications of the SNP array for conservation, management and aquaculture. © 2012 Blackwell Publishing Ltd.

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

NASA Astrophysics Data System (ADS)

Toda, Minoru; Thompson, Mitchell

2005-09-01

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

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.

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

PubMed

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

2018-03-23

To determine if a newly-designed, forehead-mounted surface microphone would yield equivalent estimates of audibility when compared to audibility measured with a skull simulator for adult bone conduction users. Data was analyzed using a within subjects, repeated measures design. There were two different sensors (skull simulator and surface microphone) measuring the same hearing aid programmed to the same settings for all subjects. We were looking for equivalent results. Twenty-one adult percutaneous bone conduction users (12 females and 9 males) were recruited for this study. Mean age was 54.32 years with a standard deviation of 14.51 years. Nineteen of the subjects had conductive/mixed hearing loss and two had single-sided deafness. To define audibility, we needed to establish two things: (1) in situ-level thresholds at each audiometric frequency in force (skull simulator) and in sound pressure level (SPL; surface microphone). Next, we measured the responses of the preprogrammed test device in force on the skull simulator and in SPL on the surface mic in response to pink noise at three input levels: 55, 65, and 75 dB SPL. The skull simulator responses were converted to real head force responses by means of an individual real head to coupler difference transform. Subtracting the real head force level thresholds from the real head force output of the test aid yielded the audibility for each audiometric frequency for the skull simulator. Subtracting the SPL thresholds from the surface microphone from the SPL output of the test aid yielded the audibility for each audiometric frequency for the surface microphone. The surface microphone was removed and retested to establish the test-retest reliability of the tool. We ran a 2 (sensor) × 3 (input level) × 10 (frequency) mixed analysis of variance to determine if there were any significant main effects and interactions. There was a significant three-way interaction, so we proceeded to explore our planned comparisons

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

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.

Performance of a modified feedback loop adaptive array with TVRO satellite signals

NASA Technical Reports Server (NTRS)

Steadman, Karl N.; Gupta, Inder J.; Walton, Eric K.

1990-01-01

Performance of an experimental adaptive antenna array system is evaluated using television receive-only (TVRO) satellite signals. The experimental system is a sidelobe canceller with two auxiliary channels. Modified feedback loops are used to enhance the suppression of weak interfering signals. The modified feedback loops used two spatialy separated antennas, each with an individual amplifier for each auxiliary channel. Thus, the experimental system uses five antenna elements. Instead of using five separate antennas, a reflector antenna with multiple feeds is used to receive signals from various TVRO satellites. The details of the earth station are given. It is shown that the experimental system can null up to two signals originating from interfering TVRO satellites while receiving the signals from a desired TVRO satellite.

Performance of a modified feedback loop adaptive array with TVRO satellite signals

NASA Technical Reports Server (NTRS)

Steadman, K.; Gupta, I. J.; Walton, E. K.

1990-01-01

The performance of an experimental adaptive antenna array system is evaluated using television-receive-only (TVRO) satellite signals. The experimental system is a sidelobe canceler with two auxiliary channels. Modified feedback loops are used to enhance the suppression of weak interfering signals. The modified feedback loops use two spatially separate antennas, each with an individual amplifier for each auxiliary channel. Thus, the experimental system uses five antenna elements. Instead of using five separate antennas, a reflector antenna with multiple feeds is used to receive signals from various TVRO satellites. The details of the earth station are given. It is shown that the experimental system can null up to two signals originating from interfering TVRO satellites while receiving the signals from a desired TVRO satellite.

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

PubMed

Tan, Chee Wee; Miao, Jianmin

2009-11-01

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

Processing of fetal heart rate through non-invasive adaptive system based on recursive least squares algorithm

NASA Astrophysics Data System (ADS)

Fajkus, Marcel; Nedoma, Jan; Martinek, Radek; Vasinek, Vladimir

2017-10-01

In this article, we describe an innovative non-invasive method of Fetal Phonocardiography (fPCG) using fiber-optic sensors and adaptive algorithm for the measurement of fetal heart rate (fHR). Conventional PCG is based on a noninvasive scanning of acoustic signals by means of a microphone placed on the thorax. As for fPCG, the microphone is placed on the maternal abdomen. Our solution is based on patent pending non-invasive scanning of acoustic signals by means of a fiber-optic interferometer. Fiber-optic sensors are resistant to technical artifacts such as electromagnetic interferences (EMI), thus they can be used in situations where it is impossible to use conventional EFM methods, e.g. during Magnetic Resonance Imaging (MRI) examination or in case of delivery in water. The adaptive evaluation system is based on Recursive least squares (RLS) algorithm. Based on real measurements provided on five volunteers with their written consent, we created a simplified dynamic signal model of a distribution of heartbeat sounds (HS) through the human body. Our created model allows us to verification of the proposed adaptive system RLS algorithm. The functionality of the proposed non-invasive adaptive system was verified by objective parameters such as Sensitivity (S+) and Signal to Noise Ratio (SNR).

Evaluation of Adaptive Noise Management Technologies for School-Age Children with Hearing Loss.

PubMed

Wolfe, Jace; Duke, Mila; Schafer, Erin; Jones, Christine; Rakita, Lori

2017-05-01

Children with hearing loss experience significant difficulty understanding speech in noisy and reverberant situations. Adaptive noise management technologies, such as fully adaptive directional microphones and digital noise reduction, have the potential to improve communication in noise for children with hearing aids. However, there are no published studies evaluating the potential benefits children receive from the use of adaptive noise management technologies in simulated real-world environments as well as in daily situations. The objective of this study was to compare speech recognition, speech intelligibility ratings (SIRs), and sound preferences of children using hearing aids equipped with and without adaptive noise management technologies. A single-group, repeated measures design was used to evaluate performance differences obtained in four simulated environments. In each simulated environment, participants were tested in a basic listening program with minimal noise management features, a manual program designed for that scene, and the hearing instruments' adaptive operating system that steered hearing instrument parameterization based on the characteristics of the environment. Twelve children with mild to moderately severe sensorineural hearing loss. Speech recognition and SIRs were evaluated in three hearing aid programs with and without noise management technologies across two different test sessions and various listening environments. Also, the participants' perceptual hearing performance in daily real-world listening situations with two of the hearing aid programs was evaluated during a four- to six-week field trial that took place between the two laboratory sessions. On average, the use of adaptive noise management technology improved sentence recognition in noise for speech presented in front of the participant but resulted in a decrement in performance for signals arriving from behind when the participant was facing forward. However, the improvement

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

NASA Astrophysics Data System (ADS)

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

2018-05-01

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

Spatial acoustic radiation of respiratory sounds for sleep evaluation.

PubMed

Shabtai, Noam R; Zigel, Yaniv

2017-09-01

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

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

PubMed

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

2006-02-01

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

Transmission mode adaptive beamforming for planar phased arrays and its application to 3D ultrasonic transcranial imaging

NASA Astrophysics Data System (ADS)

Shapoori, Kiyanoosh; Sadler, Jeffrey; Wydra, Adrian; Malyarenko, Eugene; Sinclair, Anthony; Maev, Roman G.

2013-03-01

A new adaptive beamforming method for accurately focusing ultrasound behind highly scattering layers of human skull and its application to 3D transcranial imaging via small-aperture planar phased arrays are reported. Due to its undulating, inhomogeneous, porous, and highly attenuative structure, human skull bone severely distorts ultrasonic beams produced by conventional focusing methods in both imaging and therapeutic applications. Strong acoustical mismatch between the skull and brain tissues, in addition to the skull's undulating topology across the active area of a planar ultrasonic probe, could cause multiple reflections and unpredictable refraction during beamforming and imaging processes. Such effects could significantly deflect the probe's beam from the intended focal point. Presented here is a theoretical basis and simulation results of an adaptive beamforming method that compensates for the latter effects in transmission mode, accompanied by experimental verification. The probe is a custom-designed 2 MHz, 256-element matrix array with 0.45 mm element size and 0.1mm kerf. Through its small footprint, it is possible to accurately measure the profile of the skull segment in contact with the probe and feed the results into our ray tracing program. The latter calculates the new time delay patterns adapted to the geometrical and acoustical properties of the skull phantom segment in contact with the probe. The time delay patterns correct for the refraction at the skull-brain boundary and bring the distorted beam back to its intended focus. The algorithms were implemented on the ultrasound open-platform ULA-OP (developed at the University of Florence).

Micromachined electrode array

DOEpatents

Okandan, Murat; Wessendorf, Kurt O.

2007-12-11

An electrode array is disclosed which has applications for neural stimulation and sensing. The electrode array, in certain embodiments, can include a plurality of electrodes each of which is flexibly attached to a common substrate using a plurality of springs to allow the electrodes to move independently. In other embodiments of the electrode array, the electrodes can be fixed to the substrate. The electrode array can be formed from a combination of bulk and surface micromachining, and can include electrode tips having an electroplated metal (e.g. platinum, iridium, gold or titanium) or a metal oxide (e.g. iridium oxide) for biocompatibility. The electrode array can be used to form a part of a neural prosthesis, and is particularly well adapted for use in an implantable retinal prosthesis.

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.

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.

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

PubMed

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

2016-12-26

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

Power Inversion in a Tapped Delay-Line Array.

DTIC Science & Technology

1975-03-01

and identify by block number) This report discusses recent studies on adaptive arrays for theNavy ITACS system. The report considers the power inversion...this report we discuss recent studies on adaptive arrays for the Navy ITACS system. The goal of this research is to develop an adaptive antenna system...here is a continuation of earlier research on power inversion by Compton, Lee, and Schwegman [1,2,3,4]. This work differs from previous studies in that

Acoustic Wave Guiding by Reconfigurable Tessellated Arrays

NASA Astrophysics Data System (ADS)

Zou, Chengzhe; Lynd, Danielle T.; Harne, Ryan L.

2018-01-01

The reconfiguration of origami tessellations is a prime vehicle to harness for adapting system properties governed by a structural form. While the knowledge of mechanical property changes associated with origami tessellation folding has been extensively built up, the opportunities to integrate other physics into a framework of tessellated, adaptive structures remain to be fully exploited. Acoustics appears to be a prime domain to marry with origami science. Specifically, deep technical analogies are revealed between wave-guiding properties achieved via digital methods that virtually reposition array elements and the actual repositioning of facets by folding origami-inspired tessellations. Here we capitalize on this analogy to investigate acoustic arrays established upon facet layouts of origami-inspired tessellations. We show that a concept of reconfigurable tessellated arrays may guide waves more effectively than traditional digitally phased arrays using fewer transducer elements. Moreover, we show that the refinement of tessellated arrays trends to the ideal case of classical wave radiators or receivers grounded in principles of geometrical acoustics. By linear wave physics shared among myriad scientific disciplines and across orders of magnitude in length scale, these discoveries may cultivate numerous opportunities for wave-guiding adaptive structures inspired by low-dimensional origami tessellations.

A Readout Integrated Circuit (ROIC) employing self-adaptive background current compensation technique for Infrared Focal Plane Array (IRFPA)

NASA Astrophysics Data System (ADS)

Zhou, Tong; Zhao, Jian; He, Yong; Jiang, Bo; Su, Yan

2018-05-01

A novel self-adaptive background current compensation circuit applied to infrared focal plane array is proposed in this paper, which can compensate the background current generated in different conditions. Designed double-threshold detection strategy is to estimate and eliminate the background currents, which could significantly reduce the hardware overhead and improve the uniformity among different pixels. In addition, the circuit is well compatible to various categories of infrared thermo-sensitive materials. The testing results of a 4 × 4 experimental chip showed that the proposed circuit achieves high precision, wide application and high intelligence. Tape-out of the 320 × 240 readout circuit, as well as the bonding, encapsulation and imaging verification of uncooled infrared focal plane array, have also been completed.

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

PubMed

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

1998-06-01

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

Adaptive enhancement of learning protocol in hippocampal cultured networks grown on multielectrode arrays

PubMed Central

Pimashkin, Alexey; Gladkov, Arseniy; Mukhina, Irina; Kazantsev, Victor

2013-01-01

Learning in neuronal networks can be investigated using dissociated cultures on multielectrode arrays supplied with appropriate closed-loop stimulation. It was shown in previous studies that weakly respondent neurons on the electrodes can be trained to increase their evoked spiking rate within a predefined time window after the stimulus. Such neurons can be associated with weak synaptic connections in nearby culture network. The stimulation leads to the increase in the connectivity and in the response. However, it was not possible to perform the learning protocol for the neurons on electrodes with relatively strong synaptic inputs and responding at higher rates. We proposed an adaptive closed-loop stimulation protocol capable to achieve learning even for the highly respondent electrodes. It means that the culture network can reorganize appropriately its synaptic connectivity to generate a desired response. We introduced an adaptive reinforcement condition accounting for the response variability in control stimulation. It significantly enhanced the learning protocol to a large number of responding electrodes independently on its base response level. We also found that learning effect preserved after 4–6 h after training. PMID:23745105

Adaptive lesion formation using dual mode ultrasound array system

NASA Astrophysics Data System (ADS)

Liu, Dalong; Casper, Andrew; Haritonova, Alyona; Ebbini, Emad S.

2017-03-01

We present the results from an ultrasound-guided focused ultrasound platform designed to perform real-time monitoring and control of lesion formation. Real-time signal processing of echogenicity changes during lesion formation allows for identification of signature events indicative of tissue damage. The detection of these events triggers the cessation or the reduction of the exposure (intensity and/or time) to prevent overexposure. A dual mode ultrasound array (DMUA) is used for forming single- and multiple-focus patterns in a variety of tissues. The DMUA approach allows for inherent registration between the therapeutic and imaging coordinate systems providing instantaneous, spatially-accurate feedback on lesion formation dynamics. The beamformed RF data has been shown to have high sensitivity and specificity to tissue changes during lesion formation, including in vivo. In particular, the beamformed echo data from the DMUA is very sensitive to cavitation activity in response to HIFU in a variety of modes, e.g. boiling cavitation. This form of feedback is characterized by sudden increase in echogenicity that could occur within milliseconds of the application of HIFU (see http://youtu.be/No2wh-ceTLs for an example). The real-time beamforming and signal processing allowing the adaptive control of lesion formation is enabled by a high performance GPU platform (response time within 10 msec). We present results from a series of experiments in bovine cardiac tissue demonstrating the robustness and increased speed of volumetric lesion formation for a range of clinically-relevant exposures. Gross histology demonstrate clearly that adaptive lesion formation results in tissue damage consistent with the size of the focal spot and the raster scan in 3 dimensions. In contrast, uncontrolled volumetric lesions exhibit significant pre-focal buildup due to excessive exposure from multiple full-exposure HIFU shots. Stopping or reducing the HIFU exposure upon the detection of such an

Use of Adaptive Digital Signal Processing to Improve Speech Communication for Normally Hearing aand Hearing-Impaired Subjects.

ERIC Educational Resources Information Center

Harris, Richard W.; And Others

1988-01-01

A two-microphone adaptive digital noise cancellation technique improved word-recognition ability for 20 normal and 12 hearing-impaired adults by reducing multitalker speech babble and speech spectrum noise 18-22 dB. Word recognition improvements averaged 37-50 percent for normal and 27-40 percent for hearing-impaired subjects. Improvement was best…

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.

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.

Adapting Controlled-source Coherence Analysis to Dense Array Data in Earthquake Seismology

NASA Astrophysics Data System (ADS)

Schwarz, B.; Sigloch, K.; Nissen-Meyer, T.

2017-12-01

Exploration seismology deals with highly coherent wave fields generated by repeatable controlled sources and recorded by dense receiver arrays, whose geometry is tailored to back-scattered energy normally neglected in earthquake seismology. Owing to these favorable conditions, stacking and coherence analysis are routinely employed to suppress incoherent noise and regularize the data, thereby strongly contributing to the success of subsequent processing steps, including migration for the imaging of back-scattering interfaces or waveform tomography for the inversion of velocity structure. Attempts have been made to utilize wave field coherence on the length scales of passive-source seismology, e.g. for the imaging of transition-zone discontinuities or the core-mantle-boundary using reflected precursors. Results are however often deteriorated due to the sparse station coverage and interference of faint back-scattered with transmitted phases. USArray sampled wave fields generated by earthquake sources at an unprecedented density and similar array deployments are ongoing or planned in Alaska, the Alps and Canada. This makes the local coherence of earthquake data an increasingly valuable resource to exploit.Building on the experience in controlled-source surveys, we aim to extend the well-established concept of beam-forming to the richer toolbox that is nowadays used in seismic exploration. We suggest adapted strategies for local data coherence analysis, where summation is performed with operators that extract the local slope and curvature of wave fronts emerging at the receiver array. Besides estimating wave front properties, we demonstrate that the inherent data summation can also be used to generate virtual station responses at intermediate locations where no actual deployment was performed. Owing to the fact that stacking acts as a directional filter, interfering coherent wave fields can be efficiently separated from each other by means of coherent subtraction. We

Mutual coupling, channel model, and BER for curvilinear antenna arrays

NASA Astrophysics Data System (ADS)

Huang, Zhiyong

This dissertation introduces a wireless communications system with an adaptive beam-former and investigates its performance with different antenna arrays. Mutual coupling, real antenna elements and channel models are included to examine the system performance. In a beamforming system, mutual coupling (MC) among the elements can significantly degrade the system performance. However, MC effects can be compensated if an accurate model of mutual coupling is available. A mutual coupling matrix model is utilized to compensate mutual coupling in the beamforming of a uniform circular array (UCA). Its performance is compared with other models in uplink and downlink beamforming scenarios. In addition, the predictions are compared with measurements and verified with results from full-wave simulations. In order to accurately investigate the minimum mean-square-error (MSE) of an adaptive array in MC, two different noise models, the environmental and the receiver noise, are modeled. The minimum MSEs with and without data domain MC compensation are analytically compared. The influence of mutual coupling on the convergence is also examined. In addition, the weight compensation method is proposed to attain the desired array pattern. Adaptive arrays with different geometries are implemented with the minimum MSE algorithm in the wireless communications system to combat interference at the same frequency. The bit-error-rate (BER) of systems with UCA, uniform rectangular array (URA) and UCA with center element are investigated in additive white Gaussian noise plus well-separated signals or random direction signals scenarios. The output SINR of an adaptive array with multiple interferers is analytically examined. The influence of the adaptive algorithm convergence on the BER is investigated. The UCA is then investigated in a narrowband Rician fading channel. The channel model is built and the space correlations are examined. The influence of the number of signal paths, number of the

Adaptive Array for Weak Interfering Signals: Geostationary Satellite Experiments. M.S. Thesis

NASA Technical Reports Server (NTRS)

Steadman, Karl

1989-01-01

The performance of an experimental adaptive array is evaluated using signals from an existing geostationary satellite interference environment. To do this, an earth station antenna was built to receive signals from various geostationary satellites. In these experiments the received signals have a frequency of approximately 4 GHz (C-band) and have a bandwidth of over 35 MHz. These signals are downconverted to a 69 MHz intermediate frequency in the experimental system. Using the downconverted signals, the performance of the experimental system for various signal scenarios is evaluated. In this situation, due to the inherent thermal noise, qualitative instead of quantitative test results are presented. It is shown that the experimental system can null up to two interfering signals well below the noise level. However, to avoid the cancellation of the desired signal, the use a steering vector is needed. Various methods to obtain an estimate of the steering vector are proposed.

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

PubMed

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

2017-05-16

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

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

NASA Astrophysics Data System (ADS)

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

2017-05-01

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

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

PubMed Central

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

2017-01-01

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

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

PubMed Central

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

2016-01-01

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

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

Federal Register 2010, 2011, 2012, 2013, 2014

2013-07-26

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

Coherent Optical Adaptive Techniques (COAT)

DTIC Science & Technology

1975-01-01

8217 neceeemry and Identity by block number) Laser Phased Array Adaptive Optics Atmospheric-Turbulence and Thermal Blooming Compensation 20...characteristics of an experimental, visible wavelength, eighteen-element, self-adaptive optical phased array. Measurements on a well-characterized...V LOCAL PHASING ■ LOOP OPTICAL DETECTOR’ LOCAL LOCK / ROOF TOP "^/PROPAGATION’ ^ GLINT ■lm FOCAL LENGTH LENS DETECTOR DMWI rh

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

ERIC Educational Resources Information Center

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

2014-01-01

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

MSAT-X phased array antenna adaptions to airborne applications

NASA Technical Reports Server (NTRS)

Sparks, C.; Chung, H. H.; Peng, S. Y.

1988-01-01

The Mobile Satellite Experiment (MSAT-X) phased array antenna is being modified to meet future requirements. The proposed system consists of two high gain antennas mounted on each side of a fuselage, and a low gain antenna mounted on top of the fuselage. Each antenna is an electronically steered phased array based on the design of the MSAT-X antenna. A beamforming network is connected to the array elements via coaxial cables. It is essential that the proposed antenna system be able to provide an adequate communication link over the required space coverage, which is 360 degrees in azimuth and from 20 degrees below the horizon to the zenith in elevation. Alternative design concepts are suggested. Both open loop and closed loop backup capabilities are discussed. Typical antenna performance data are also included.

Opto-VLSI-based photonic true-time delay architecture for broadband adaptive nulling in phased array antennas.

PubMed

Juswardy, Budi; Xiao, Feng; Alameh, Kamal

2009-03-16

This paper proposes a novel Opto-VLSI-based tunable true-time delay generation unit for adaptively steering the nulls of microwave phased array antennas. Arbitrary single or multiple true-time delays can simultaneously be synthesized for each antenna element by slicing an RF-modulated broadband optical source and routing specific sliced wavebands through an Opto-VLSI processor to a high-dispersion fiber. Experimental results are presented, which demonstrate the principle of the true-time delay unit through the generation of 5 arbitrary true-time delays of up to 2.5 ns each. (c) 2009 Optical Society of America

The CHARA array adaptive optics I: common-path optical and mechanical design, and preliminary on-sky results

NASA Astrophysics Data System (ADS)

Che, Xiao; Sturmann, Laszlo; Monnier, John D.; ten Brummelaar, Theo A.; Sturmann, Judit; Ridgway, Stephen T.; Ireland, Michael J.; Turner, Nils H.; McAlister, Harold A.

2014-07-01

The CHARA array is an optical interferometer with six 1-meter diameter telescopes, providing baselines from 33 to 331 meters. With sub-milliarcsecond angular resolution, its versatile visible and near infrared combiners offer a unique angle of studying nearby stellar systems by spatially resolving their detailed structures. To improve the sensitivity and scientific throughput, the CHARA array was funded by NSF-ATI in 2011 to install adaptive optics (AO) systems on all six telescopes. The initial grant covers Phase I of the AO systems, which includes on-telescope Wavefront Sensors (WFS) and non-common-path (NCP) error correction. Meanwhile we are seeking funding for Phase II which will add large Deformable Mirrors on telescopes to close the full AO loop. The corrections of NCP error and static aberrations in the optical system beyond the WFS are described in the second paper of this series. This paper describes the design of the common-path optical system and the on-telescope WFS, and shows the on-sky commissioning results.

Performance analysis of structured gradient algorithm. [for adaptive beamforming linear arrays

NASA Technical Reports Server (NTRS)

Godara, Lal C.

1990-01-01

The structured gradient algorithm uses a structured estimate of the array correlation matrix (ACM) to estimate the gradient required for the constrained least-mean-square (LMS) algorithm. This structure reflects the structure of the exact array correlation matrix for an equispaced linear array and is obtained by spatial averaging of the elements of the noisy correlation matrix. In its standard form the LMS algorithm does not exploit the structure of the array correlation matrix. The gradient is estimated by multiplying the array output with the receiver outputs. An analysis of the two algorithms is presented to show that the covariance of the gradient estimated by the structured method is less sensitive to the look direction signal than that estimated by the standard method. The effect of the number of elements on the signal sensitivity of the two algorithms is studied.

Adaption of the Magnetometer Towed Array geophysical system to meet Department of Energy needs for hazardous waste site characterization

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

Cochran, J.R.; McDonald, J.R.; Russell, R.J.

1995-10-01

This report documents US Department of Energy (DOE)-funded activities that have adapted the US Navy`s Surface Towed Ordnance Locator System (STOLS) to meet DOE needs for a ``... better, faster, safer and cheaper ...`` system for characterizing inactive hazardous waste sites. These activities were undertaken by Sandia National Laboratories (Sandia), the Naval Research Laboratory, Geo-Centers Inc., New Mexico State University and others under the title of the Magnetometer Towed Array (MTA).

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

NASA Astrophysics Data System (ADS)

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

2017-09-01

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

Estimation of Temporal Gait Parameters Using a Wearable Microphone-Sensor-Based System

PubMed Central

Wang, Cheng; Wang, Xiangdong; Long, Zhou; Yuan, Jing; Qian, Yueliang; Li, Jintao

2016-01-01

Most existing wearable gait analysis methods focus on the analysis of data obtained from inertial sensors. This paper proposes a novel, low-cost, wireless and wearable gait analysis system which uses microphone sensors to collect footstep sound signals during walking. This is the first time a microphone sensor is used as a wearable gait analysis device as far as we know. Based on this system, a gait analysis algorithm for estimating the temporal parameters of gait is presented. The algorithm fully uses the fusion of two feet footstep sound signals and includes three stages: footstep detection, heel-strike event and toe-on event detection, and calculation of gait temporal parameters. Experimental results show that with a total of 240 data sequences and 1732 steps collected using three different gait data collection strategies from 15 healthy subjects, the proposed system achieves an average 0.955 F1-measure for footstep detection, an average 94.52% accuracy rate for heel-strike detection and 94.25% accuracy rate for toe-on detection. Using these detection results, nine temporal related gait parameters are calculated and these parameters are consistent with their corresponding normal gait temporal parameters and labeled data calculation results. The results verify the effectiveness of our proposed system and algorithm for temporal gait parameter estimation. PMID:27999321

Adaptive Identification by Systolic Arrays.

DTIC Science & Technology

1987-12-01

BIBLIOGRIAPHY Anton , Howard, Elementary Linear Algebra , John Wiley & Sons, 19S4. Cristi, Roberto, A Parallel Structure Jor Adaptive Pole Placement...10 11. SYSTEM IDENTIFICATION M*YETHODS ....................... 12 A. LINEAR SYSTEM MODELING ......................... 12 B. SOLUTION OF SYSTEMS OF... LINEAR EQUATIONS ......... 13 C. QR DECOMPOSITION ................................ 14 D. RECURSIVE LEAST SQUARES ......................... 16 E. BLOCK

Uncoordinated MAC for Adaptive Multi Beam Directional Networks: Analysis and Evaluation

DTIC Science & Technology

2016-08-01

control (MAC) policies for emerging systems that are equipped with fully digital antenna arrays which are capable of adaptive multi-beam directional...Adaptive Beam- forming, Multibeam, Directional Networking, Random Access, Smart Antennas I. INTRODUCTION Fully digital beamforming antenna arrays that...are capable of adaptive multi-beam communications are quickly becoming a reality. These antenna arrays allow users to form multiple simultaneous

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.

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.

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.

High-Gain Airborne Microphone Windscreen Characterization Method Using Modified Research Wind Tunnel

NASA Astrophysics Data System (ADS)

Banks, Joseph Andrew

In recent years, UAS (unmanned aerial systems) have gained improved functionality by integrating advanced cameras, sensors, and hardware systems; however, UAS still lack effective means to detect and record audio signals. This is partially due to the physical scale of hardware and complexity of that hardware's integration into UAS. The current study is part of a larger research effort to integrate a high-gain parabolic microphone into a UAV (unmanned aerial vehicle) for use in acoustic surveying. Due to the aerodynamic interaction between a flush mounted parabolic antenna and the free-stream grazing flow, it is necessary to fair the antenna into the aircraft using a windscreen. The current study develops a characterization method by which various windscreen designs and configurations can be optimized. This method measures a candidate windscreen's normal incidence sound transmission loss (STL) as well as the increase of hydrodynamic noise generated by its installation at a range of flow speeds. A test apparatus was designed and installed on the Low Speed Wind Tunnel at Oklahoma State University. The test apparatus utilizes a "quiet box" attached to the wind tunnel test section floor. A pass-through window between the wind tunnel test section and the quiet box allows candidate wind screens to be mounted between the two environments. Microphones mounted both in the wind tunnel test section, and within the quiet box record the acoustic spectrum at various flow speeds, ranging between 36 and 81 feet per second. A tensioned KevlarRTM wind screen validation specimen was fabricated to validate system performance. The STL spectrum is measured based on comparing the signal from microphones on either side of the KevlarRTM membrane. The results for normal incidence STL for the flow off scenario are compared to results presented in other studies for the same material under tension. Flow-on transmission loss spectral data along with the increase in flow noise caused by the

Environmental photobioreactor array (EPBRA) systems and apparatus related thereto

DOEpatents

Kramer, David; Zegarac, Robert; Lucker, Ben F.; Hall, Christopher; Abernathy, Casey; Carpenter, Joel; Cruz, Jeffrey

2017-11-14

A system is described herein that comprises one or more modular environmental photobioreactor arrays, each array containing two or more photobioreactors, wherein the system is adapted to monitor each of the photobioreactors and/or modulate the conditions with each of the photobioreactors. The photobioreactors are also adapted for measurement of multiple physiological parameters of a biomass contained therein. Various methods for selecting and characterizing biomass are also provided. In one embodiment, the biomass is algae.

Adaptive smart simulator for characterization and MPPT construction of PV array

NASA Astrophysics Data System (ADS)

Ouada, Mehdi; Meridjet, Mohamed Salah; Dib, Djalel

2016-07-01

Partial shading conditions are among the most important problems in large photovoltaic array. Many works of literature are interested in modeling, control and optimization of photovoltaic conversion of solar energy under partial shading conditions, The aim of this study is to build a software simulator similar to hard simulator and to produce a shading pattern of the proposed photovoltaic array in order to use the delivered information to obtain an optimal configuration of the PV array and construct MPPT algorithm. Graphical user interfaces (Matlab GUI) are built using a developed script, this tool is easy to use, simple, and has a rapid of responsiveness, the simulator supports large array simulations that can be interfaced with MPPT and power electronic converters.

Adaptive smart simulator for characterization and MPPT construction of PV array

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

Ouada, Mehdi, E-mail: mehdi.ouada@univ-annaba.org; Meridjet, Mohamed Salah; Dib, Djalel

2016-07-25

Partial shading conditions are among the most important problems in large photovoltaic array. Many works of literature are interested in modeling, control and optimization of photovoltaic conversion of solar energy under partial shading conditions, The aim of this study is to build a software simulator similar to hard simulator and to produce a shading pattern of the proposed photovoltaic array in order to use the delivered information to obtain an optimal configuration of the PV array and construct MPPT algorithm. Graphical user interfaces (Matlab GUI) are built using a developed script, this tool is easy to use, simple, and hasmore » a rapid of responsiveness, the simulator supports large array simulations that can be interfaced with MPPT and power electronic converters.« less

Flexible retinal electrode array

DOEpatents

Okandan, Murat [Albuquerque, NM; Wessendorf, Kurt O [Albuquerque, NM; Christenson, Todd R [Albuquerque, NM

2006-10-24

An electrode array which has applications for neural stimulation and sensing. The electrode array can include a large number of electrodes each of which is flexibly attached to a common substrate using a plurality of springs to allow the electrodes to move independently. The electrode array can be formed from a combination of bulk and surface micromachining, with electrode tips that can include an electroplated metal (e.g. platinum, iridium, gold or titanium) or a metal oxide (e.g. iridium oxide) for biocompatibility. The electrode array can be used to form a part of a neural prosthesis, and is particularly well adapted for use in an implantable retinal prosthesis where the electrodes can be tailored to provide a uniform gentle contact pressure with optional sensing of this contact pressure at one or more of the electrodes.

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

PubMed Central

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

2013-01-01

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

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.

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.

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.

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 fast adaptive convex hull algorithm on two-dimensional processor arrays with a reconfigurable BUS system

NASA Technical Reports Server (NTRS)

Olariu, S.; Schwing, J.; Zhang, J.

1991-01-01

A bus system that can change dynamically to suit computational needs is referred to as reconfigurable. We present a fast adaptive convex hull algorithm on a two-dimensional processor array with a reconfigurable bus system (2-D PARBS, for short). Specifically, we show that computing the convex hull of a planar set of n points taken O(log n/log m) time on a 2-D PARBS of size mn x n with 3 less than or equal to m less than or equal to n. Our result implies that the convex hull of n points in the plane can be computed in O(1) time in a 2-D PARBS of size n(exp 1.5) x n.

Adaptive Waveform Correlation Detectors for Arrays: Algorithms for Autonomous Calibration

DTIC Science & Technology

2007-09-01

March 17, 2005. The seismic signals from both master and detected events are followed by infrasound arrivals. Note the long duration of the...correlation coefficient traces with a significant array -gain. A detected event that is co-located with the master event will record the same time-difference...estimating the detection threshold reduction for a range of highly repeating seismic sources using arrays of different configurations and at different

Cold plasma decontamination using flexible jet arrays

NASA Astrophysics Data System (ADS)

Konesky, Gregory

2010-04-01

Arrays of atmospheric discharge cold plasma jets have been used to decontaminate surfaces of a wide range of microorganisms quickly, yet not damage that surface. Its effectiveness in decomposing simulated chemical warfare agents has also been demonstrated, and may also find use in assisting in the cleanup of radiological weapons. Large area jet arrays, with short dwell times, are necessary for practical applications. Realistic situations will also require jet arrays that are flexible to adapt to contoured or irregular surfaces. Various large area jet array prototypes, both planar and flexible, are described, as is the application to atmospheric decontamination.

Apparatus and method for imaging metallic objects using an array of giant magnetoresistive sensors

DOEpatents

Chaiken, Alison

2000-01-01

A portable, low-power, metallic object detector and method for providing an image of a detected metallic object. In one embodiment, the present portable low-power metallic object detector an array of giant magnetoresistive (GMR) sensors. The array of GMR sensors is adapted for detecting the presence of and compiling image data of a metallic object. In the embodiment, the array of GMR sensors is arranged in a checkerboard configuration such that axes of sensitivity of alternate GMR sensors are orthogonally oriented. An electronics portion is coupled to the array of GMR sensors. The electronics portion is adapted to receive and process the image data of the metallic object compiled by the array of GMR sensors. The embodiment also includes a display unit which is coupled to the electronics portion. The display unit is adapted to display a graphical representation of the metallic object detected by the array of GMR sensors. In so doing, a graphical representation of the detected metallic object is provided.

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

NASA Astrophysics Data System (ADS)

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

2014-05-01

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

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

ERIC Educational Resources Information Center

Ellison, Tisha Lewis; Kirkland, David E.

2014-01-01

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

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

NASA Astrophysics Data System (ADS)

Avison, Janine; Barham, Richard

2014-01-01

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

Performance Evaluation of Multichannel Adaptive Algorithms for Local Active Noise Control

NASA Astrophysics Data System (ADS)

DE DIEGO, M.; GONZALEZ, A.

2001-07-01

This paper deals with the development of a multichannel active noise control (ANC) system inside an enclosed space. The purpose is to design a real practical system which works well in local ANC applications. Moreover, the algorithm implemented in the adaptive controller should be robust, of low computational complexity and it should manage to generate a uniform useful-size zone of quite in order to allow the head motion of a person seated on a seat inside a car. Experiments were carried out under semi-anechoic and listening room conditions to verify the successful implementation of the multichannel system. The developed prototype consists of an array of up to four microphones used as error sensors mounted on the headrest of a seat place inside the enclosure. One loudspeaker was used as single primary source and two secondary sources were placed facing the seat. The aim of this multichannel system is to reduce the sound pressure levels in an area around the error sensors, following a local control strategy. When using this technique, the cancellation points are not only the error sensor positions but an area around them, which is measured by using a monitoring microphone. Different multichannel adaptive algorithms for ANC have been analyzed and their performance verified. Multiple error algorithms are used in order to cancel out different types of primary noise (engine noise and random noise) with several configurations (up to four channels system). As an alternative to the multiple error LMS algorithm (multichannel version of the filtered-X LMS algorithm, MELMS), the least maximum mean squares (LMMS) and the scanning error-LMS algorithm have been developed in this work in order to reduce computational complexity and achieve a more uniform residual field. The ANC algorithms were programmed on a digital signal processing board equipped with a TMS320C40 floating point DSP processor. Measurements concerning real-time experiments on local noise reduction in two

Steerable Space Fed Lens Array for Low-Cost Adaptive Ground Station Applications

NASA Technical Reports Server (NTRS)

Lee, Richard Q.; Popovic, Zoya; Rondineau, Sebastien; Miranda, Felix A.

2007-01-01

The Space Fed Lens Array (SFLA) is an alternative to a phased array antenna that replaces large numbers of expensive solid-state phase shifters with a single spatial feed network. SFLA can be used for multi-beam application where multiple independent beams can be generated simultaneously with a single antenna aperture. Unlike phased array antennas where feed loss increases with array size, feed loss in a lens array with more than 50 elements is nearly independent of the number of elements, a desirable feature for large apertures. In addition, SFLA has lower cost as compared to a phased array at the expense of total volume and complete beam continuity. For ground station applications, both of these tradeoff parameters are not important and can thus be exploited in order to lower the cost of the ground station. In this paper, we report the development and demonstration of a 952-element beam-steerable SFLA intended for use as a low cost ground station for communicating and tracking of a low Earth orbiting satellite. The dynamic beam steering is achieved through switching to different feed-positions of the SFLA via a beam controller.

Adaptive Noise Suppression Using Digital Signal Processing

NASA Technical Reports Server (NTRS)

Kozel, David; Nelson, Richard

1996-01-01

A signal to noise ratio dependent adaptive spectral subtraction algorithm is developed to eliminate noise from noise corrupted speech signals. The algorithm determines the signal to noise ratio and adjusts the spectral subtraction proportion appropriately. After spectra subtraction low amplitude signals are squelched. A single microphone is used to obtain both eh noise corrupted speech and the average noise estimate. This is done by determining if the frame of data being sampled is a voiced or unvoiced frame. During unvoice frames an estimate of the noise is obtained. A running average of the noise is used to approximate the expected value of the noise. Applications include the emergency egress vehicle and the crawler transporter.

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.

In situ Probe Microphone Measurement for Testing the Direct Acoustical Cochlear Stimulator.

PubMed

Stieger, Christof; Alnufaily, Yasser H; Candreia, Claudia; Caversaccio, Marco D; Arnold, Andreas M

2017-01-01

Hypothesis: Acoustical measurements can be used for functional control of a direct acoustic cochlear stimulator (DACS). Background: The DACS is a recently released active hearing implant that works on the principle of a conventional piston prosthesis driven by the rod of an electromagnetic actuator. An inherent part of the DACS actuator is a thin titanium diaphragm that allows for movement of the stimulation rod while hermetically sealing the housing. In addition to mechanical stimulation, the actuator emits sound into the mastoid cavity because of the motion of the diaphragm. Methods: We investigated the use of the sound emission of a DACS for intra-operative testing. We measured sound emission in the external auditory canal (P EAC ) and velocity of the actuators stimulation rod (V act ) in five implanted ears of whole-head specimens. We tested the influence various positions of the loudspeaker and a probe microphone on P EAC and simulated implant malfunction in one example. Results: Sound emission of the DACS with a signal-to-noise ratio >10 dB was observed between 0.5 and 5 kHz. Simulated implant misplacement or malfunction could be detected by the absence or shift in the characteristic resonance frequency of the actuator. P EAC changed by <6 dB for variations of the microphone and loudspeaker position. Conclusion: Our data support the feasibility of acoustical measurements for in situ testing of the DACS implant in the mastoid cavity as well as for post-operative monitoring of actuator function.

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.

Activity recognition of assembly tasks using body-worn microphones and accelerometers.

PubMed

Ward, Jamie A; Lukowicz, Paul; Tröster, Gerhard; Starner, Thad E

2006-10-01

In order to provide relevant information to mobile users, such as workers engaging in the manual tasks of maintenance and assembly, a wearable computer requires information about the user's specific activities. This work focuses on the recognition of activities that are characterized by a hand motion and an accompanying sound. Suitable activities can be found in assembly and maintenance work. Here, we provide an initial exploration into the problem domain of continuous activity recognition using on-body sensing. We use a mock "wood workshop" assembly task to ground our investigation. We describe a method for the continuous recognition of activities (sawing, hammering, filing, drilling, grinding, sanding, opening a drawer, tightening a vise, and turning a screwdriver) using microphones and three-axis accelerometers mounted at two positions on the user's arms. Potentially "interesting" activities are segmented from continuous streams of data using an analysis of the sound intensity detected at the two different locations. Activity classification is then performed on these detected segments using linear discriminant analysis (LDA) on the sound channel and hidden Markov models (HMMs) on the acceleration data. Four different methods at classifier fusion are compared for improving these classifications. Using user-dependent training, we obtain continuous average recall and precision rates (for positive activities) of 78 percent and 74 percent, respectively. Using user-independent training (leave-one-out across five users), we obtain recall rates of 66 percent and precision rates of 63 percent. In isolation, these activities were recognized with accuracies of 98 percent, 87 percent, and 95 percent for the user-dependent, user-independent, and user-adapted cases, respectively.

Dynamically Reconfigurable Systolic Array Accelerator

NASA Technical Reports Server (NTRS)

Dasu, Aravind; Barnes, Robert

2012-01-01

A polymorphic systolic array framework has been developed that works in conjunction with an embedded microprocessor on a field-programmable gate array (FPGA), which allows for dynamic and complimentary scaling of acceleration levels of two algorithms active concurrently on the FPGA. Use is made of systolic arrays and a hardware-software co-design to obtain an efficient multi-application acceleration system. The flexible and simple framework allows hosting of a broader range of algorithms, and is extendable to more complex applications in the area of aerospace embedded systems. FPGA chips can be responsive to realtime demands for changing applications needs, but only if the electronic fabric can respond fast enough. This systolic array framework allows for rapid partial and dynamic reconfiguration of the chip in response to the real-time needs of scalability, and adaptability of executables.

Analyzing acoustic phenomena with a smartphone microphone

NASA Astrophysics Data System (ADS)

Kuhn, Jochen; Vogt, Patrik

2013-02-01

This paper describes how different sound types can be explored using the microphone of a smartphone and a suitable app. Vibrating bodies, such as strings, membranes, or bars, generate air pressure fluctuations in their immediate vicinity, which propagate through the room in the form of sound waves. Depending on the triggering mechanism, it is possible to differentiate between four types of sound waves: tone, sound, noise, and bang. In everyday language, non-experts use the terms "tone" and "sound" synonymously; however, from a physics perspective there are very clear differences between the two terms. This paper presents experiments that enable learners to explore and understand these differences. Tuning forks and musical instruments (e.g., recorders and guitars) can be used as equipment for the experiments. The data are captured using a smartphone equipped with the appropriate app (in this paper we describe the app Audio Kit for iOS systems ). The values captured by the smartphone are displayed in a screen shot and then viewed directly on the smartphone or exported to a computer graphics program for printing.

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,

Spatio-Temporal Equalizer for a Receiving-Antenna Feed Array

NASA Technical Reports Server (NTRS)

Mukai, Ryan; Lee, Dennis; Vilnrotter, Victor

2010-01-01

A spatio-temporal equalizer has been conceived as an improved means of suppressing multipath effects in the reception of aeronautical telemetry signals, and may be adaptable to radar and aeronautical communication applications as well. This equalizer would be an integral part of a system that would also include a seven-element planar array of receiving feed horns centered at the focal point of a paraboloidal antenna that would be nominally aimed at or near the aircraft that would be the source of the signal that one seeks to receive (see Figure 1). This spatio-temporal equalizer would consist mostly of a bank of seven adaptive finite-impulse-response (FIR) filters one for each element in the array - and the outputs of the filters would be summed (see Figure 2). The combination of the spatial diversity of the feedhorn array and the temporal diversity of the filter bank would afford better multipath-suppression performance than is achievable by means of temporal equalization alone. The seven-element feed array would supplant the single feed horn used in a conventional paraboloidal ground telemetry-receiving antenna. The radio-frequency telemetry signals re ceiv ed by the seven elements of the array would be digitized, converted to complex baseband form, and sent to the FIR filter bank, which would adapt itself in real time to enable reception of telemetry at a low bit error rate, even in the presence of multipath of the type found at many flight test ranges.

Improvement of plastic optical fiber microphone based on moisture pattern sensing in devoiced breath

NASA Astrophysics Data System (ADS)

Taki, Tomohito; Honma, Satoshi; Morisawa, Masayuki; Muto, Shinzo

2008-03-01

Conversation is the most practical and common form in communication. However, people with a verbal handicap feel a difficulty to produce words due to variations in vocal chords. This research leads to develop a new devoiced microphone system based on distinguishes between the moisture patterns for each devoiced breaths, using a plastic optical fiber (POF) moisture sensor. In the experiment, five POF-type moisture sensors with fast response were fabricated by coating swell polymer with a slightly larger refractive index than that of fiber core and were set in front of mouth. When these sensors are exposed into humid air produced by devoiced breath, refractive index in cladding layer decreases by swelling and then the POF sensor heads change to guided type. Based on the above operation principle, the output light intensities from the five sensors set in front of mouth change each other. Using above mentioned output light intensity patterns, discernment of devoiced vowels in Japanese (a,i,u,e,o) was tried by means of DynamicProgramming-Matching (DP-matching) method. As the result, distinction rate over 90% was obtained to Japanese devoiced vowels. Therefore, using this system and a voice synthesizer, development of new microphone for the person with a functional disorder in the vocal chords seems to be possible.

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.

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

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.

Probe-tube microphone measures in hearing-impaired children and adults.

PubMed

Barlow, N L; Auslander, M C; Rines, D; Stelmachowicz, P G

1988-10-01

This study was designed to investigate the reliability of real-ear measurements of sound pressure level (SPL) and to compare these values with two coupler measures of SPL. A commercially available probe tube microphone system was used to measure real ear SPL in both children and adults. Test-retest reliability decreased as a function of frequency for both groups and, in general, was slightly poorer for the children. For both groups, coupler to real ear differences were larger for the 2 cm3 coupler than for the reduced volume coupler; however, no significant differences were observed between groups. In addition, a measure of ear canal volume was not found to be a good predictor of coupler to real ear discrepancies.

LSSA (Low-cost Silicon Solar Array) project

NASA Technical Reports Server (NTRS)

1976-01-01

Methods are explored for economically generating electrical power to meet future requirements. The Low-Cost Silicon Solar Array Project (LSSA) was established to reduce the price of solar arrays by improving manufacturing technology, adapting mass production techniques, and promoting user acceptance. The new manufacturing technology includes the consideration of new silicon refinement processes, silicon sheet growth techniques, encapsulants, and automated assembly production being developed under contract by industries and universities.

Adaptive array technique for differential-phase reflectometry in QUEST

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

Idei, H., E-mail: idei@triam.kyushu-u.ac.jp; Hanada, K.; Zushi, H.

2014-11-15

A Phased Array Antenna (PAA) was considered as launching and receiving antennae in reflectometry to attain good directivity in its applied microwave range. A well-focused beam was obtained in a launching antenna application, and differential-phase evolution was properly measured by using a metal reflector plate in the proof-of-principle experiment at low power test facilities. Differential-phase evolution was also evaluated by using the PAA in the Q-shu University Experiment with Steady State Spherical Tokamak (QUEST). A beam-forming technique was applied in receiving phased-array antenna measurements. In the QUEST device that should be considered as a large oversized cavity, standing wave effectmore » was significantly observed with perturbed phase evolution. A new approach using derivative of measured field on propagating wavenumber was proposed to eliminate the standing wave effect.« less

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 d

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.

Adaptive EAGLE dynamic solution adaptation and grid quality enhancement

NASA Technical Reports Server (NTRS)

Luong, Phu Vinh; Thompson, J. F.; Gatlin, B.; Mastin, C. W.; Kim, H. J.

1992-01-01

In the effort described here, the elliptic grid generation procedure in the EAGLE grid code was separated from the main code into a subroutine, and a new subroutine which evaluates several grid quality measures at each grid point was added. The elliptic grid routine can now be called, either by a computational fluid dynamics (CFD) code to generate a new adaptive grid based on flow variables and quality measures through multiple adaptation, or by the EAGLE main code to generate a grid based on quality measure variables through static adaptation. Arrays of flow variables can be read into the EAGLE grid code for use in static adaptation as well. These major changes in the EAGLE adaptive grid system make it easier to convert any CFD code that operates on a block-structured grid (or single-block grid) into a multiple adaptive code.

Biomimetic micromechanical adaptive flow-sensor arrays

NASA Astrophysics Data System (ADS)

Krijnen, Gijs; Floris, Arjan; Dijkstra, Marcel; Lammerink, Theo; Wiegerink, Remco

2007-05-01

We report current developments in biomimetic flow-sensors based on flow sensitive mechano-sensors of crickets. Crickets have one form of acoustic sensing evolved in the form of mechanoreceptive sensory hairs. These filiform hairs are highly perceptive to low-frequency sound with energy sensitivities close to thermal threshold. In this work we describe hair-sensors fabricated by a combination of sacrificial poly-silicon technology, to form silicon-nitride suspended membranes, and SU8 polymer processing for fabrication of hairs with diameters of about 50 μm and up to 1 mm length. The membranes have thin chromium electrodes on top forming variable capacitors with the substrate that allow for capacitive read-out. Previously these sensors have been shown to exhibit acoustic sensitivity. Like for the crickets, the MEMS hair-sensors are positioned on elongated structures, resembling the cercus of crickets. In this work we present optical measurements on acoustically and electrostatically excited hair-sensors. We present adaptive control of flow-sensitivity and resonance frequency by electrostatic spring stiffness softening. Experimental data and simple analytical models derived from transduction theory are shown to exhibit good correspondence, both confirming theory and the applicability of the presented approach towards adaptation.

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.

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.

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.

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

Geophysical Inversion with Adaptive Array Processing of Ambient Noise

NASA Astrophysics Data System (ADS)

Traer, James

2011-12-01

Land-based seismic observations of microseisms generated during Tropical Storms Ernesto and Florence are dominated by signals in the 0.15--0.5Hz band. Data from seafloor hydrophones in shallow water (70m depth, 130 km off the New Jersey coast) show dominant signals in the gravity-wave frequency band, 0.02--0.18Hz and low amplitudes from 0.18--0.3Hz, suggesting significant opposing wave components necessary for DF microseism generation were negligible at the site. Both storms produced similar spectra, despite differing sizes, suggesting near-coastal shallow water as the dominant region for observed microseism generation. A mathematical explanation for a sign-inversion induced to the passive fathometer response by minimum variance distortionless response (MVDR) beamforming is presented. This shows that, in the region containing the bottom reflection, the MVDR fathometer response is identical to that obtained with conventional processing multiplied by a negative factor. A model is presented for the complete passive fathometer response to ocean surface noise, interfering discrete noise sources, and locally uncorrelated noise in an ideal waveguide. The leading order term of the ocean surface noise produces the cross-correlation of vertical multipaths and yields the depth of sub-bottom reflectors. Discrete noise incident on the array via multipaths give multiple peaks in the fathometer response. These peaks may obscure the sub-bottom reflections but can be attenuated with use of Minimum Variance Distortionless Response (MVDR) steering vectors. A theory is presented for the Signal-to-Noise-Ratio (SNR) for the seabed reflection peak in the passive fathometer response as a function of seabed depth, seabed reflection coefficient, averaging time, bandwidth and spatial directivity of the noise field. The passive fathometer algorithm was applied to data from two drifting array experiments in the Mediterranean, Boundary 2003 and 2004, with 0.34s of averaging time. In the 2004

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.

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.

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

An experimental SMI adaptive antenna array for weak interfering signals

NASA Technical Reports Server (NTRS)

Dilsavor, R. L.; Gupta, I. J.

1989-01-01

A modified sample matrix inversion (SMI) algorithm designed to increase the suppression of weak interference is implemented on an existing experimental array system. The algorithm itself is fully described as are a number of issues concerning its implementation and evaluation, such as sample scaling, snapshot formation, weight normalization, power calculation, and system calibration. Several experiments show that the steady state performance (i.e., many snapshots are used to calculate the array weights) of the experimental system compares favorably with its theoretical performance. It is demonstrated that standard SMI does not yield adequate suppression of weak interference. Modified SMI is then used to experimentally increase this suppression by as much as 13dB.

Reconfigurable signal processor designs for advanced digital array radar systems

NASA Astrophysics Data System (ADS)

Suarez, Hernan; Zhang, Yan (Rockee); Yu, Xining

2017-05-01

The new challenges originated from Digital Array Radar (DAR) demands a new generation of reconfigurable backend processor in the system. The new FPGA devices can support much higher speed, more bandwidth and processing capabilities for the need of digital Line Replaceable Unit (LRU). This study focuses on using the latest Altera and Xilinx devices in an adaptive beamforming processor. The field reprogrammable RF devices from Analog Devices are used as analog front end transceivers. Different from other existing Software-Defined Radio transceivers on the market, this processor is designed for distributed adaptive beamforming in a networked environment. The following aspects of the novel radar processor will be presented: (1) A new system-on-chip architecture based on Altera's devices and adaptive processing module, especially for the adaptive beamforming and pulse compression, will be introduced, (2) Successful implementation of generation 2 serial RapidIO data links on FPGA, which supports VITA-49 radio packet format for large distributed DAR processing. (3) Demonstration of the feasibility and capabilities of the processor in a Micro-TCA based, SRIO switching backplane to support multichannel beamforming in real-time. (4) Application of this processor in ongoing radar system development projects, including OU's dual-polarized digital array radar, the planned new cylindrical array radars, and future airborne radars.

Low frequency wind noise contributions in measurement microphones.

PubMed

Raspet, Richard; Yu, Jiao; Webster, Jeremy

2008-03-01

In a previous paper [R. Raspet, et al., J. Acoust. Soc. Am. 119, 834-843 (2006)], a method was introduced to predict upper and lower bounds for wind noise measured in spherical wind-screens from the measured incident velocity spectra. That paper was restricted in that the predictions were only valid within the inertial range of the incident turbulence, and the data were from a measurement not specifically designed to test the predictions. This paper extends the previous predictions into the source region of the atmospheric wind turbulence, and compares the predictions to measurements made with a large range of wind-screen sizes. Predictions for the turbulence-turbulence interaction pressure spectrum as well as the stagnation pressure fluctuation spectrum are calculated from a form fit to the velocity fluctuation spectrum. While the predictions for turbulence-turbulence interaction agree well with measurements made within large (1.0 m) wind-screens, and the stagnation pressure predictions agree well with unscreened gridded microphone measurements, the mean shear-turbulence interaction spectra do not consistently appear in measurements.

Electrowetting-based adaptive vari-focal liquid lens array for 3D display

NASA Astrophysics Data System (ADS)

Won, Yong Hyub

2014-10-01

Electrowetting is a phenomenon that can control the surface tension of liquid when a voltage is applied. This paper introduces the fabrication method of liquid lens array by the electrowetting phenomenon. The fabricated 23 by 23 lens array has 1mm diameter size with 1.6 mm interval distance between adjacent lenses. The diopter of each lens was - 24~27 operated at 0V to 50V. The lens array chamber fabricated by Deep Reactive-Ion Etching (DRIE) is deposited with IZO and parylene C and tantalum oxide. To prevent water penetration and achieve high dielectric constant, parylene C and tantalum oxide (ɛ = 23 ~ 25) are used respectively. Hydrophobic surface which enables the range of contact angle from 60 to 160 degree is coated to maximize the effect of electrowetting causing wide band of dioptric power. Liquid is injected into each lens chamber by two different ways. First way was self water-oil dosing that uses cosolvent and diffusion effect, while the second way was micro-syringe by the hydrophobic surface properties. To complete the whole process of the lens array fabrication, underwater sealing was performed using UV adhesive that does not dissolve in water. The transient time for changing from concave to convex lens was measured <33ms (at frequency of 1kHz AC voltage.). The liquid lens array was tested unprecedentedly for integral imaging to achieve more advanced depth information of 3D image.

Multi-element array signal reconstruction with adaptive least-squares algorithms

NASA Technical Reports Server (NTRS)

Kumar, R.

1992-01-01

Two versions of the adaptive least-squares algorithm are presented for combining signals from multiple feeds placed in the focal plane of a mechanical antenna whose reflector surface is distorted due to various deformations. Coherent signal combining techniques based on the adaptive least-squares algorithm are examined for nearly optimally and adaptively combining the outputs of the feeds. The performance of the two versions is evaluated by simulations. It is demonstrated for the example considered that both of the adaptive least-squares algorithms are capable of offsetting most of the loss in the antenna gain incurred due to reflector surface deformations.

High density arrays of micromirrors

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

Folta, J. M.; Decker, J. Y.; Kolman, J.

We established and achieved our goal to (1) fabricate and evaluate test structures based on the micromirror design optimized for maskless lithography applications, (2) perform system analysis and code development for the maskless lithography concept, and (3) identify specifications for micromirror arrays (MMAs) for LLNL's adaptive optics (AO) applications and conceptualize new devices.

Active Hearing Mechanisms Inspire Adaptive Amplification in an Acoustic Sensor System.

PubMed

Guerreiro, Jose; Reid, Andrew; Jackson, Joseph C; Windmill, James F C

2018-06-01

Over many millions of years of evolution, nature has developed some of the most adaptable sensors and sensory systems possible, capable of sensing, conditioning and processing signals in a very power- and size-effective manner. By looking into biological sensors and systems as a source of inspiration, this paper presents the study of a bioinspired concept of signal processing at the sensor level. By exploiting a feedback control mechanism between a front-end acoustic receiver and back-end neuronal based computation, a nonlinear amplification with hysteretic behavior is created. Moreover, the transient response of the front-end acoustic receiver can also be controlled and enhanced. A theoretical model is proposed and the concept is prototyped experimentally through an embedded system setup that can provide dynamic adaptations of a sensory system comprising a MEMS microphone placed in a closed-loop feedback system. It faithfully mimics the mosquito's active hearing response as a function of the input sound intensity. This is an adaptive acoustic sensor system concept that can be exploited by sensor and system designers within acoustics and ultrasonic engineering fields.

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

High Altitude Infrasound Measurements using Balloon-Borne Arrays

NASA Astrophysics Data System (ADS)

Bowman, D. C.; Johnson, C. S.; Gupta, R. A.; Anderson, J.; Lees, J. M.; Drob, D. P.; Phillips, D.

2015-12-01

For the last fifty years, almost all infrasound sensors have been located on the Earth's surface. A few experiments consisting of microphones on poles and tethered aerostats comprise the remainder. Such surface and near-surface arrays likely do not capture the full diversity of acoustic signals in the atmosphere. Here, we describe results from a balloon mounted infrasound array that reached altitudes of up to 38 km (the middle stratosphere). The balloon drifted at the ambient wind speed, resulting in a near total reduction in wind noise. Signals consistent with tropospheric turbulence were detected. A spectral peak in the ocean microbarom range (0.12 - 0.35 Hz) was present on balloon-mounted sensors but not on static infrasound stations near the flight path. A strong 18 Hz signal, possibly related to building ventilation systems, was observed in the stratosphere. A wide variety of other narrow band acoustic signals of uncertain provenance were present throughout the flight, but were absent in simultaneous recordings from nearby ground stations. Similar phenomena were present in spectrograms from the last balloon infrasound campaign in the 1960s. Our results suggest that the infrasonic wave field in the stratosphere is very different from that which is readily detectable on surface stations. This has implications for modeling acoustic energy transfer between the lower and upper atmosphere as well as the detection of novel acoustic signals that never reach the ground. Our work provides valuable constraints on a proposed mission to detect earthquakes on Venus using balloon-borne infrasound sensors.

Memory device for two-dimensional radiant energy array computers

NASA Technical Reports Server (NTRS)

Schaefer, D. H.; Strong, J. P., III (Inventor)

1977-01-01

A memory device for two dimensional radiant energy array computers was developed, in which the memory device stores digital information in an input array of radiant energy digital signals that are characterized by ordered rows and columns. The memory device contains a radiant energy logic storing device having a pair of input surface locations for receiving a pair of separate radiant energy digital signal arrays and an output surface location adapted to transmit a radiant energy digital signal array. A regenerative feedback device that couples one of the input surface locations to the output surface location in a manner for causing regenerative feedback is also included

Adapter plate assembly for adjustable mounting of objects

DOEpatents

Blackburn, R.S.

1986-05-02

An adapter plate and two locking discs are together affixed to an optic table with machine screws or bolts threaded into a fixed array of internally threaded holes provided in the table surface. The adapter plate preferably has two, and preferably parallel, elongated locating slots each freely receiving a portion of one of the locking discs for secure affixation of the adapter plate to the optic table. A plurality of threaded apertures provided in the adapter plate are available to attach optical mounts or other devices onto the adapter plate in an orientation not limited by the disposition of the array of threaded holes in the table surface. An axially aligned but radially offset hole through each locking disc receives a screw that tightens onto the table, such that prior to tightening of the screw the locking disc may rotate and translate within each locating slot of the adapter plate for maximum flexibility of the orientation thereof.

Adapter plate assembly for adjustable mounting of objects

DOEpatents

Blackburn, Robert S.

1987-01-01

An adapter plate and two locking discs are together affixed to an optic table with machine screws or bolts threaded into a fixed array of internally threaded holes provided in the table surface. The adapter plate preferably has two, and preferably parallel, elongated locating slots each freely receiving a portion of one of the locking discs for secure affixation of the adapter plate to the optic table. A plurality of threaded apertures provided in the adapter plate are available to attach optical mounts or other devices onto the adapter plate in an orientation not limited by the disposition of the array of threaded holes in the table surface. An axially aligned but radially offset hole through each locking disc receives a screw that tightens onto the table, such that prior to tightening of the screw the locking disc may rotate and translate within each locating slot of the adapter plate for maximum flexibility of the orientation thereof.

Active Acoustic Array for Ultrasonic Biomedical Applications

DTIC Science & Technology

2001-07-30

of the human anatomy and means to 3 acoustically couple the acoustic array to the portion of the 4 human anatomy . The acoustic array is doubly...ultrasonic sound wave to be generated into the portion 14 of the 22 human anatomy . As previously mentioned, each of the acoustic 23 elements 28 acts as...human breast, it should be 3 recognized that the device can be adapted to detect cancer in 4 other portions of the human anatomy . 5 It is apparent

Adaptive Optics Communications Performance Analysis

NASA Technical Reports Server (NTRS)

Srinivasan, M.; Vilnrotter, V.; Troy, M.; Wilson, K.

2004-01-01

The performance improvement obtained through the use of adaptive optics for deep-space communications in the presence of atmospheric turbulence is analyzed. Using simulated focal-plane signal-intensity distributions, uncoded pulse-position modulation (PPM) bit-error probabilities are calculated assuming the use of an adaptive focal-plane detector array as well as an adaptively sized single detector. It is demonstrated that current practical adaptive optics systems can yield performance gains over an uncompensated system ranging from approximately 1 dB to 6 dB depending upon the PPM order and background radiation level.

Simulation of Thin-Film Damping and Thermal Mechanical Noise Spectra for Advanced Micromachined Microphone Structures.

PubMed

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

2008-06-01

In many micromachined sensors the thin (2-10 μm thick) air film between a compliant diaphragm and backplate electrode plays a dominant role in shaping both the dynamic and thermal noise characteristics of the device. Silicon microphone structures used in grating-based optical-interference microphones have recently been introduced that employ backplates with minimal area to achieve low damping and low thermal noise levels. Finite-element based modeling procedures based on 2-D discretization of the governing Reynolds equation are ideally suited for studying thin-film dynamics in such structures which utilize relatively complex backplate geometries. In this paper, the dynamic properties of both the diaphragm and thin air film are studied using a modal projection procedure in a commonly used finite element software and the results are used to simulate the dynamic frequency response of the coupled structure to internally generated electrostatic actuation pressure. The model is also extended to simulate thermal mechanical noise spectra of these advanced sensing structures. In all cases simulations are compared with measured data and show excellent agreement-demonstrating 0.8 pN/√Hz and 1.8 μPa/√Hz thermal force and thermal pressure noise levels, respectively, for the 1.5 mm diameter structures under study which have a fundamental diaphragm resonance-limited bandwidth near 20 kHz.

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

NASA Astrophysics Data System (ADS)

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

2018-05-01

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

Development and Technical Validation of the Mobile Based Assistive Listening System: A Smartphone-Based Remote Microphone.

PubMed

Lopez, Esteban Alejandro; Costa, Orozimbo Alves; Ferrari, Deborah Viviane

2016-10-01

The purpose of this research note is to describe the development and technical validation of the Mobile Based Assistive Listening System (MoBALS), a free-of-charge smartphone-based remote microphone application. MoBALS Version 1.0 was developed for Android (Version 2.1 or higher) and was coded with Java using Eclipse Indigo with the Android Software Development Kit. A Wi-Fi router with background traffic and 2 affordable smartphones were used for debugging and technical validation comprising, among other things, multicasting capability, data packet loss, and battery consumption. MoBALS requires at least 2 smartphones connected to the same Wi-Fi router for signal transmission and reception. Subscriber identity module cards or Internet connections are not needed. MoBALS can be used alone or connected to a hearing aid or cochlear implant via direct audio input. Maximum data packet loss was 99.28%, and minimum battery life was 5 hr. Other relevant design specifications and their implementation are described. MoBALS performed as a remote microphone with enhanced accessibility features and avoids overhead expenses by using already-available and affordable technology. The further development and technical revalidation of MoBALS will be followed by clinical evaluation with persons with hearing impairment.

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.

Using a signal cancellation technique to assess adaptive directivity of hearing aids.

PubMed

Wu, Yu-Hsiang; Bentler, Ruth A

2007-07-01

The directivity of an adaptive directional microphone hearing aid (DMHA) cannot be assessed by the method that calls for presenting a "probe" signal from a single loudspeaker to the DMHA that moves to different angles. This method is invalid because the probe signal itself changes the polar pattern. This paper proposes a method for assessing the adaptive DMHA using a "jammer" signal, presented from a second loudspeaker rotating with the DMHA, that simulates a noise source and freezes the polar pattern. Measurement at each angle is obtained by two sequential recordings from the DMHA, one using an input of a probe and a jammer, and the other with an input of the same probe and a phase-inverted jammer. After canceling out the jammer, the remaining response to the probe signal can be used to assess the directivity. In this paper, the new method is evaluated by comparing responses from five adaptive DMHAs to different jammer intensities and locations. This method was shown to be an accurate and reliable way to assess the directivity of the adaptive DMHA in a high-intensity-jammer condition.

Piezo-Phototronic Enhanced UV Sensing Based on a Nanowire Photodetector Array.

PubMed

Han, Xun; Du, Weiming; Yu, Ruomeng; Pan, Caofeng; Wang, Zhong Lin

2015-12-22

A large array of Schottky UV photodetectors (PDs) based on vertical aligned ZnO nanowires is achieved. By introducing the piezo-phototronic effect, the performance of the PD array is enhanced up to seven times in photoreponsivity, six times in sensitivity, and 2.8 times in detection limit. The UV PD array may have applications in optoelectronic systems, adaptive optical computing, and communication. © 2015 WILEY-VCH Verlag GmbH & Co. KGaA, Weinheim.

Digitally Controlled Slot Coupled Patch Array

NASA Technical Reports Server (NTRS)

D'Arista, Thomas; Pauly, Jerry

2010-01-01

A four-element array conformed to a singly curved conducting surface has been demonstrated to provide 2 dB axial ratio of 14 percent, while maintaining VSWR (voltage standing wave ratio) of 2:1 and gain of 13 dBiC. The array is digitally controlled and can be scanned with the LMS Adaptive Algorithm using the power spectrum as the objective, as well as the Direction of Arrival (DoA) of the beam to set the amplitude of the power spectrum. The total height of the array above the conducting surface is 1.5 inches (3.8 cm). A uniquely configured microstrip-coupled aperture over a conducting surface produced supergain characteristics, achieving 12.5 dBiC across the 2-to-2.13- GHz and 2.2-to-2.3-GHz frequency bands. This design is optimized to retain VSWR and axial ratio across the band as well. The four elements are uniquely configured with respect to one another for performance enhancement, and the appropriate phase excitation to each element for scan can be found either by analytical beam synthesis using the genetic algorithm with the measured or simulated far field radiation pattern, or an adaptive algorithm implemented with the digitized signal. The commercially available tuners and field-programmable gate array (FPGA) boards utilized required precise phase coherent configuration control, and with custom code developed by Nokomis, Inc., were shown to be fully functional in a two-channel configuration controlled by FPGA boards. A four-channel tuner configuration and oscilloscope configuration were also demonstrated although algorithm post-processing was required.

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

The Fuge Tube Diode Array Spectrophotometer

ERIC Educational Resources Information Center

Arneson, B. T.; Long, S. R.; Stewart, K. K.; Lagowski, J. J.

2008-01-01

We present the details for adapting a diode array UV-vis spectrophotometer to incorporate the use of polypropylene microcentrifuge tubes--fuge tubes--as cuvettes. Optical data are presented validating that the polyethylene fuge tubes are equivalent to the standard square cross section polystyrene or glass cuvettes generally used in…

CR-Calculus and adaptive array theory applied to MIMO random vibration control tests

NASA Astrophysics Data System (ADS)

Musella, U.; Manzato, S.; Peeters, B.; Guillaume, P.

2016-09-01

Performing Multiple-Input Multiple-Output (MIMO) tests to reproduce the vibration environment in a user-defined number of control points of a unit under test is necessary in applications where a realistic environment replication has to be achieved. MIMO tests require vibration control strategies to calculate the required drive signal vector that gives an acceptable replication of the target. This target is a (complex) vector with magnitude and phase information at the control points for MIMO Sine Control tests while in MIMO Random Control tests, in the most general case, the target is a complete spectral density matrix. The idea behind this work is to tailor a MIMO random vibration control approach that can be generalized to other MIMO tests, e.g. MIMO Sine and MIMO Time Waveform Replication. In this work the approach is to use gradient-based procedures over the complex space, applying the so called CR-Calculus and the adaptive array theory. With this approach it is possible to better control the process performances allowing the step-by-step Jacobian Matrix update. The theoretical bases behind the work are followed by an application of the developed method to a two-exciter two-axis system and by performance comparisons with standard methods.

An active drop counting device using condenser microphone for superheated emulsion detector

NASA Astrophysics Data System (ADS)

Das, Mala; Arya, A. S.; Marick, C.; Kanjilal, D.; Saha, S.