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Sample records for adaptive microphone array

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

    NASA Technical Reports Server (NTRS)

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

    2011-01-01

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

  2. Acoustic positioning using multiple microphone arrays.

    PubMed

    Liu, Hui; Milios, Evangelos

    2005-05-01

    Passive acoustic techniques are presented to solve the localization problem of a sound source in three-dimensional space using off-the-shelf hardware. Multiple microphone arrays are employed, which operate independently, in estimating the direction of arrival of sound, or, equivalently, a direction vector from the array's geometric center towards the source. Direction vectors and array centers are communicated to a central processor, where the source is localized by finding the intersection of the direction lines defined by the direction vectors and the associated array centers. The performance of the method in the air is demonstrated experimentally and compared with a state-of-the-art method that requires centralized digitization of the signals from the microphones of all the arrays. PMID:15957748

  3. Design theory of microphone arrays for teleconferencing

    NASA Astrophysics Data System (ADS)

    Macomber, Dwight Frank

    2001-07-01

    Room reverberation and interfering acoustical noise lower the quality of speech transmission in teleconferencing. Conventional solutions for speech capture that suppress pickup of reverberation and interference typically constrain the motion of the speaker, encumber the speaker physically, or make it difficult for many different speakers to easily participate. These impediments result from cables, radio and headset microphones, or so-called house microphones located at fixed positions. Microphone arrays and matched-filter processing have been proposed as solutions to this sound capture problem. In part because the behavior of arrays in reverberant acoustic spaces has not been well quantified, there have been no guidelines for designing effective teleconference arrays. The principal goal of this work is the development of general design rules for speech acquisition arrays. Expressions for the array performance measures of signal-to-reverberation ratio and signal-to-interference ratio are derived using statistical acoustics, then verified by computer simulation using the image method of geometric room acoustics. The analysis is facilitated by assuming the reverberant sound field to be diffuse. The assumption is valid above approximately 250 Hz. All work assumes omnidirectional sources and sensors mounted on walls with low absorption. Array sensors should be placed close to, and roughly equidistant to the source, yet as far from each other as possible. The aperture of a full-band array should extend into three dimensions. Wall and ceiling mounting of sensors is recommended. Relations specify the number of sensors required for various room volumes, room absorptions, and source-to-sensor distances. A planar ceiling array expands the range of motion for speakers by widening the focal region above 300 Hz. Off-line audition of a 32-sensor array in a 60-cubicmeter room with a reverberation time of .63 second indicates that subjectively good array performance may be obtained

  4. Compressive sensing with a spherical microphone array.

    PubMed

    Fernandez-Grande, Efren; Xenaki, Angeliki

    2016-02-01

    A wave expansion method is proposed in this work, based on measurements with a spherical microphone array, and formulated in the framework provided by Compressive Sensing. The method promotes sparse solutions via ℓ1-norm minimization, so that the measured data are represented by few basis functions. This results in fine spatial resolution and accuracy. This publication covers the theoretical background of the method, including experimental results that illustrate some of the fundamental differences with the "conventional" least-squares approach. The proposed methodology is relevant for source localization, sound field reconstruction, and sound field analysis.

  5. 50 years of progress in microphone arrays for speech processing

    NASA Astrophysics Data System (ADS)

    Elko, Gary W.; Frisk, George V.

    2004-10-01

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

  6. Research on optical fiber microphone array based on Sagnac interferometer

    NASA Astrophysics Data System (ADS)

    Wu, Hongyan; Wang, Jian

    2015-05-01

    Extensive attention has been paid to optical fiber microphone because of its especial merits, such as anti-electromagnetic interference, corrosion resistance, high sensitivity, safety and reliability. In the present study, a kind of optical fiber microphone array based on Sagnac interferometer using a broadband source is proposed. On the basis of the high sound quality and wide bandwidth of optical fiber microphones, the acoustic source localization theory is tested and verified in practice. The results prove the possibility of determine the location of acoustic source in a wide range of frequencies accurately. Besides its feasibility, the scientific value and application prospect, such as in battlefield and ultrasonic detection field, are great.

  7. Llamas: Large-area microphone arrays and sensing systems

    NASA Astrophysics Data System (ADS)

    Sanz-Robinson, Josue

    Large-area electronics (LAE) provides a platform to build sensing systems, based on distributing large numbers of densely spaced sensors over a physically-expansive space. Due to their flexible, "wallpaper-like" form factor, these systems can be seamlessly deployed in everyday spaces. They go beyond just supplying sensor readings, but rather they aim to transform the wealth of data from these sensors into actionable inferences about our physical environment. This requires vertically integrated systems that span the entirety of the signal processing chain, including transducers and devices, circuits, and signal processing algorithms. To this end we develop hybrid LAE / CMOS systems, which exploit the complementary strengths of LAE, enabling spatially distributed sensors, and CMOS ICs, providing computational capacity for signal processing. To explore the development of hybrid sensing systems, based on vertical integration across the signal processing chain, we focus on two main drivers: (1) thin-film diodes, and (2) microphone arrays for blind source separation: 1) Thin-film diodes are a key building block for many applications, such as RFID tags or power transfer over non-contact inductive links, which require rectifiers for AC-to-DC conversion. We developed hybrid amorphous / nanocrystalline silicon diodes, which are fabricated at low temperatures (<200 °C) to be compatible with processing on plastic, and have high current densities (5 A/cm2 at 1 V) and high frequency operation (cutoff frequency of 110 MHz). 2) We designed a system for separating the voices of multiple simultaneous speakers, which can ultimately be fed to a voice-command recognition engine for controlling electronic systems. On a device level, we developed flexible PVDF microphones, which were used to create a large-area microphone array. On a circuit level we developed localized a-Si TFT amplifiers, and a custom CMOS IC, for system control, sensor readout and digitization. On a signal processing

  8. Design and Use of Microphone Directional Arrays for Aeroacoustic Measurements

    NASA Technical Reports Server (NTRS)

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

    1998-01-01

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

  9. 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.

  10. Novel error sensing microphone arrays for active control of turbofan rotor/stator tones

    NASA Astrophysics Data System (ADS)

    Walker, Bruce E.; Hersh, Alan S.; Rice, Edward J.; Sutliff, Daniel L.

    2003-10-01

    Active control of turbofan rotor/stator interaction tones is complicated by the simultaneous presence of multiple duct propagation modes. In-duct error sensing microphone arrays that can adequately resolve these modes typically require duct lengths that are incompatible with modern compact engine design. Two alternative approaches have been investigated. For inlet noise, an external linear array of microphones was positioned in the near/far radiation field transition region and weighted to provide error signals resolved either by duct mode or by radiation angle. For the exhaust, radially spaced microphones have been placed on duct bifurcation panels to provide supplemental radial-mode resolution. The concepts were tested in combination with an adaptive segmented liner in a static duct and as part of an active stator-vane system in the ANCF research facility at NASA/Glenn Research Center. [Work sponsored by NASA/Langley Research Center.

  11. Parallel Processing of Large Scale Microphone Arrays for Sound Capture

    NASA Astrophysics Data System (ADS)

    Jan, Ea-Ee.

    1995-01-01

    Performance of microphone sound pick up is degraded by deleterious properties of the acoustic environment, such as multipath distortion (reverberation) and ambient noise. The degradation becomes more prominent in a teleconferencing environment in which the microphone is positioned far away from the speaker. Besides, the ideal teleconference should feel as easy and natural as face-to-face communication with another person. This suggests hands-free sound capture with no tether or encumbrance by hand-held or body-worn sound equipment. Microphone arrays for this application represent an appropriate approach. This research develops new microphone array and signal processing techniques for high quality hands-free sound capture in noisy, reverberant enclosures. The new techniques combine matched-filtering of individual sensors and parallel processing to provide acute spatial volume selectivity which is capable of mitigating the deleterious effects of noise interference and multipath distortion. The new method outperforms traditional delay-and-sum beamformers which provide only directional spatial selectivity. The research additionally explores truncated matched-filtering and random distribution of transducers to reduce complexity and improve sound capture quality. All designs are first established by computer simulation of array performance in reverberant enclosures. The simulation is achieved by a room model which can efficiently calculate the acoustic multipath in a rectangular enclosure up to a prescribed order of images. It also calculates the incident angle of the arriving signal. Experimental arrays were constructed and their performance was measured in real rooms. Real room data were collected in a hard-walled laboratory and a controllable variable acoustics enclosure of similar size, approximately 6 x 6 x 3 m. An extensive speech database was also collected in these two enclosures for future research on microphone arrays. The simulation results are shown to be

  12. Wake Vortex Detection: Phased Microphone vs. Linear Infrasonic Array

    NASA Technical Reports Server (NTRS)

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

    2014-01-01

    Sensor technologies can make a significant impact on the detection of aircraft-generated vortices in an air space of interest, typically in the approach or departure corridor. Current state-of-the art sensor technologies do not provide three-dimensional measurements needed for an operational system or even for wake vortex modeling to advance the understanding of vortex behavior. Most wake vortex sensor systems used today have been developed only for research applications and lack the reliability needed for continuous operation. The main challenges for the development of an operational sensor system are reliability, all-weather operation, and spatial coverage. Such a sensor has been sought for a period of last forty years. Acoustic sensors were first proposed and tested by National Oceanic and Atmospheric Administration (NOAA) early in 1970s for tracking wake vortices but these acoustic sensors suffered from high levels of ambient noise. Over a period of the last fifteen years, there has been renewed interest in studying noise generated by aircraft wake vortices, both numerically and experimentally. The German Aerospace Center (DLR) was the first to propose the application of a phased microphone array for the investigation of the noise sources of wake vortices. The concept was first demonstrated at Berlins Airport Schoenefeld in 2000. A second test was conducted in Tarbes, France, in 2002, where phased microphone arrays were applied to study the wake vortex noise of an Airbus 340. Similarly, microphone phased arrays and other opto-acoustic microphones were evaluated in a field test at the Denver International Airport in 2003. For the Tarbes and Denver tests, the wake trajectories of phased microphone arrays and lidar were compared as these were installed side by side. Due to a built-in pressure equalization vent these microphones were not suitable for capturing acoustic noise below 20 Hz. Our group at NASA Langley Research Center developed and installed an

  13. Factors affecting the performance of large-aperture microphone arrays.

    PubMed

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

    2002-05-01

    Large arrays of microphones have been proposed and studied as a possible means of acquiring data in offices, conference rooms, and auditoria without requiring close-talking microphones. When such an array essentially surrounds all possible sources, it is said to have a large aperture. Large-aperture arrays have attractive properties of spatial resolution and signal-to-noise enhancement. This paper presents a careful comparison of theoretical and measured performance for an array of 256 microphones using simple delay-and-sum beamforming. This is the largest currently functional, all digital-signal-processing array that we know of. The array is wall-mounted in the moderately adverse environment of a general-purpose laboratory (8 m x 8 m x 3 m). The room has a T60 reverberation time of 550 ms. Reverberation effects in this room severely impact the array's performance. However, the width of the main lobe remains comparable to that of a simplified prediction. Broadband spatial resolution shows a single central peak with 10 dB gain about 0.4 m in diameter at the -3 dB level. Away from that peak, the response is approximately flat over most of the room. Optimal weighting for signal-to-noise enhancement degrades the spatial resolution minimally. Experimentally, we verify that signal-to-noise gain is less than proportional to the square root of the number of microphones probably due to the partial correlation of the noise between channels, to variation of signal intensity with polar angle about the source, and to imperfect correlation of the signal over the array caused by reverberations. We show measurements of the relative importance of each effect in our environment. PMID:12051434

  14. Factors affecting the performance of large-aperture microphone arrays.

    PubMed

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

    2002-05-01

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

  15. Factors affecting the performance of large-aperture microphone arrays

    NASA Astrophysics Data System (ADS)

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

    2002-05-01

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

  16. 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.

  17. Noise Reduction with Microphone Arrays for Speaker Identification

    SciTech Connect

    Cohen, Z

    2011-12-22

    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 identification 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?

  18. Room geometry inference based on spherical microphone array eigenbeam processing.

    PubMed

    Mabande, Edwin; Kowalczyk, Konrad; Sun, Haohai; Kellermann, Walter

    2013-10-01

    The knowledge of parameters characterizing an acoustic environment, such as the geometric information about a room, can be used to enhance the performance of several audio applications. In this paper, a novel method for three-dimensional room geometry inference based on robust and high-resolution beamforming techniques for spherical microphone arrays is presented. Unlike other approaches that are based on the measurement and processing of multiple room impulse responses, here, microphone array signal processing techniques for uncontrolled broadband acoustic signals are applied. First, the directions of arrival (DOAs) and time differences of arrival (TDOAs) of the direct signal and room reflections are estimated using high-resolution robust broadband beamforming techniques and cross-correlation analysis. In this context, the main challenges include the low reflected-signal to background-noise power ratio, the low energy of reflected signals relative to the direct signal, and their strong correlation with the direct signal and among each other. Second, the DOA and TDOA information is combined to infer the room geometry using geometric relations. The high accuracy of the proposed room geometry inference technique is confirmed by experimental evaluations based on both simulated and measured data for moderately reverberant rooms. PMID:24116416

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

    NASA Technical Reports Server (NTRS)

    Mueller, Arnold W.; Wilson, Mark R.

    1997-01-01

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

  20. Theory and design of compact hybrid microphone arrays on two-dimensional planes for three-dimensional soundfield analysis.

    PubMed

    Chen, Hanchi; Abhayapala, Thushara D; Zhang, Wen

    2015-11-01

    Soundfield analysis based on spherical harmonic decomposition has been widely used in various applications; however, a drawback is the three-dimensional geometry of the microphone arrays. In this paper, a method to design two-dimensional planar microphone arrays that are capable of capturing three-dimensional (3D) spatial soundfields is proposed. Through the utilization of both omni-directional and first order microphones, the proposed microphone array is capable of measuring soundfield components that are undetectable to conventional planar omni-directional microphone arrays, thus providing the same functionality as 3D arrays designed for the same purpose. Simulations show that the accuracy of the planar microphone array is comparable to traditional spherical microphone arrays. Due to its compact shape, the proposed microphone array greatly increases the feasibility of 3D soundfield analysis techniques in real-world applications.

  1. Sound field reconstruction using a spherical microphone array.

    PubMed

    Fernandez-Grande, Efren

    2016-03-01

    A method is presented that makes it possible to reconstruct an arbitrary sound field based on measurements with a spherical microphone array. The proposed method (spherical equivalent source method) makes use of a point source expansion to describe the sound field on the rigid spherical array, from which it is possible to reconstruct the sound field over a three-dimensional domain, inferring all acoustic quantities: sound pressure, particle velocity, and sound intensity. The problem is formulated using a Neumann Green's function that accounts for the presence of the rigid sphere in the medium. One can reconstruct the total sound field, or only the incident part, i.e., the scattering introduced by the sphere can be removed, making the array virtually transparent. The method makes it possible to use sequential measurements: different measurement positions can be combined, providing an extended measurement area consisting of an array of spheres, and the sound field at any point of the source-free domain can be estimated, not being restricted to spherical surfaces. Because it is formulated as an elementary wave model, it allows for diverse solution strategies (least squares, ℓ1-norm minimization, etc.), revealing an interesting perspective for further work.

  2. Sound field reconstruction using a spherical microphone array.

    PubMed

    Fernandez-Grande, Efren

    2016-03-01

    A method is presented that makes it possible to reconstruct an arbitrary sound field based on measurements with a spherical microphone array. The proposed method (spherical equivalent source method) makes use of a point source expansion to describe the sound field on the rigid spherical array, from which it is possible to reconstruct the sound field over a three-dimensional domain, inferring all acoustic quantities: sound pressure, particle velocity, and sound intensity. The problem is formulated using a Neumann Green's function that accounts for the presence of the rigid sphere in the medium. One can reconstruct the total sound field, or only the incident part, i.e., the scattering introduced by the sphere can be removed, making the array virtually transparent. The method makes it possible to use sequential measurements: different measurement positions can be combined, providing an extended measurement area consisting of an array of spheres, and the sound field at any point of the source-free domain can be estimated, not being restricted to spherical surfaces. Because it is formulated as an elementary wave model, it allows for diverse solution strategies (least squares, ℓ1-norm minimization, etc.), revealing an interesting perspective for further work. PMID:27036253

  3. Plane-wave decomposition by spherical-convolution microphone array

    NASA Astrophysics Data System (ADS)

    Rafaely, Boaz; Park, Munhum

    2001-05-01

    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.

  4. MEMS Microphone Array Sensor for Air-Coupled Impact-Echo.

    PubMed

    Groschup, Robin; Grosse, Christian U

    2015-01-01

    Impact-Echo (IE) is a nondestructive testing technique for plate like concrete structures. We propose a new sensor concept for air-coupled IE measurements. By using an array of MEMS (micro-electro-mechanical system) microphones, instead of a single receiver, several operational advantages compared to conventional sensing strategies in IE are achieved. The MEMS microphone array sensor is cost effective, less sensitive to undesired effects like acoustic noise and has an optimized sensitivity for signals that need to be extracted for IE data interpretation. The proposed sensing strategy is justified with findings from numerical simulations, showing that the IE resonance in plate like structures causes coherent surface displacements on the specimen under test in an area around the impact location. Therefore, by placing several MEMS microphones on a sensor array board, the IE resonance is easier to be identified in the recorded spectra than with single point microphones or contact type transducers. A comparative measurement between the array sensor, a conventional accelerometer and a measurement microphone clearly shows the suitability of MEMS type microphones and the advantages of using these microphones in an array arrangement for IE. The MEMS microphone array will make air-coupled IE measurements faster and more reliable.

  5. MEMS Microphone Array Sensor for Air-Coupled Impact-Echo

    PubMed Central

    Groschup, Robin; Grosse, Christian U.

    2015-01-01

    Impact-Echo (IE) is a nondestructive testing technique for plate like concrete structures. We propose a new sensor concept for air-coupled IE measurements. By using an array of MEMS (micro-electro-mechanical system) microphones, instead of a single receiver, several operational advantages compared to conventional sensing strategies in IE are achieved. The MEMS microphone array sensor is cost effective, less sensitive to undesired effects like acoustic noise and has an optimized sensitivity for signals that need to be extracted for IE data interpretation. The proposed sensing strategy is justified with findings from numerical simulations, showing that the IE resonance in plate like structures causes coherent surface displacements on the specimen under test in an area around the impact location. Therefore, by placing several MEMS microphones on a sensor array board, the IE resonance is easier to be identified in the recorded spectra than with single point microphones or contact type transducers. A comparative measurement between the array sensor, a conventional accelerometer and a measurement microphone clearly shows the suitability of MEMS type microphones and the advantages of using these microphones in an array arrangement for IE. The MEMS microphone array will make air-coupled IE measurements faster and more reliable. PMID:26121610

  6. 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.

  7. 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.

  8. A directional microphone array for acoustic studies of wind tunnel models

    NASA Technical Reports Server (NTRS)

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

    1974-01-01

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

  9. Standoff photoacoustic detections with high-sensitivity microphones and acoustic arrays

    NASA Astrophysics Data System (ADS)

    Choa, Fow-Sen; Wang, Chen-Chia; Khurgin, Jacob; Samuels, Alan; Trivedi, Sudhir; Gupta, Deepa

    2016-05-01

    Standoff detection of dangerous chemicals like explosives, nerve gases, and harmful aerosols has continuously been an important subject due to the serious concern about terrorist threats to both overseas and homeland lives and facility. Compared with other currently available standoff optical detection techniques, like Raman, photo-thermal, laser induced breakdown spectroscopy,...etc., photoacoustic (PA) sensing has the advantages of background free and very high detection sensitivity, no need of back reflection surfaces, and 1/R instead of 1/R2 signal decay distance dependence. Furthermore, there is still a great room for PA sensitivity improvement by using different PA techniques, including lockin amplifier, employing new microphones, and microphone array techniques. Recently, we have demonstrated standoff PA detection of isopropanol vapor, solid phase TNT and RDX at a standoff distance. To further calibrate the detection sensitivity, we use nerve gas simulants that were generated and calibrated by a commercial vapor generator. For field operations, array of microphones and microphone-reflector pairs can be utilized to achieve noise rejection and signal enhancement. We have experimentally demonstrated signal enhancement and noise reduction using an array of 4 microphone/4 reflector system as well as an array of 16-microphone/1 reflector. In this work we will review and compare different standoff techniques and discuss the advantages of using different photoacoustic techniques. We will also discuss new advancement of using new types of microphone and the performance comparison of using different structure of microphone arrays and combining lock-in amplifier with acoustic arrays. Demonstration of out-door real-time operations with high power mid-IR laser and microphone array will be presented.

  10. Low-frequency wind noise correlation in microphone arrays.

    PubMed

    Shields, F Douglas

    2005-06-01

    A three-axis orthogonal microphone array with ten sensors in each arm has been used to study wind noise in the frequency range from 0.05 to 50 Hz. Simultaneous measurements were made of the three components of the varying wind velocity. Measurements have been made for wind speeds from 4 to 7 m/s at three different sites. The frequency-dependent correlation of the wind noise over a range of wind velocities and atmospheric and environmental conditions in the downwind direction varies as exp(-3.2X)cos(27piX). For the crosswind and vertical directions, the correlation decays approximately as exp(-7Y), where X is the separation in wavelengths in the downwind direction and Y is this separation in the crosswind or vertical direction. Over a limited range of wave numbers, the power density spectra of the varying wind velocity varied as the wave number to the -(5/3) power and the pressure spectra as the -(7/3) power.

  11. Pressure-sensitive paint as a distributed optical microphone array.

    PubMed

    Gregory, James W; Sullivan, John P; Wanis, Sameh S; Komerath, Narayanan M

    2006-01-01

    Pressure-sensitive paint is presented and evaluated in this article as a quantitative technique for measurement of acoustic pressure fluctuations. This work is the culmination of advances in paint technology which enable unsteady measurements of fluctuations over 10 kHz at pressure levels as low as 125 dB. Pressure-sensitive paint may be thought of as a nano-scale array of optical microphones with a spatial resolution limited primarily by the resolution of the imaging device. Thus, pressure-sensitive paint is a powerful tool for making high-amplitude sound pressure measurements. In this work, the paint was used to record ensemble-averaged, time-resolved, quantitative measurements of two-dimensional mode shapes in an acoustic resonance cavity. A wall-mounted speaker generated nonlinear, standing acoustic waves in a rigid enclosure measuring 216 mm wide, 169 mm high, and 102 mm deep. The paint recorded the acoustic surface pressures of the (1,1,0) mode shape at approximately 1.3 kHz and a sound pressure level of 145.4 dB. Results from the paint are compared with data from a Kulite pressure transducer, and with linear acoustic theory. The paint may be used as a diagnostic technique for ultrasonic tests where high spatial resolution is essential, or in nonlinear acoustic applications such as shock tubes.

  12. Ultrasensitive directional microphone arrays for military operations in urban terrain.

    SciTech Connect

    Hall, Neal A.; Peterson, Kenneth Allen; Parker, Eric Paul; Resnick, Paul James; Okandan, Murat; Serkland, Darwin Keith

    2007-11-01

    Acoustic sensing systems are critical elements in detection of sniper events. The microphones developed in this project enable unique sensing systems that benefit significantly from the enhanced sensitivity and extremely compact foot-print. Surface and bulk micromachining technologies developed at Sandia have allowed the design, fabrication and characterization of these unique sensors. We have demonstrated sensitivity that is only available in 1/2 inch to 1 inch studio reference microphones--with our devices that have only 1 to 2mm diameter membranes in a volume less than 1cm{sup 3}.

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

    NASA Technical Reports Server (NTRS)

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

    2003-01-01

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

  14. Blind location and separation of callers in a natural chorus using a microphone array.

    PubMed

    Jones, Douglas L; Ratnam, Rama

    2009-08-01

    Male frogs and toads call in dense choruses to attract females. Determining the vocal interactions and spatial distribution of the callers is important for understanding acoustic communication in such assemblies. It has so far proved difficult to simultaneously locate and recover the vocalizations of individual callers. Here a microphone-array technique is developed for blindly locating callers using arrival-time delays at the microphones, estimating their steering-vectors, and recovering the calls with a frequency-domain adaptive beamformer. The technique exploits the time-frequency sparseness of the signal space to recover sources even when there are more sources than sensors. The method is tested with data collected from a natural chorus of Gulf Coast toads (Bufo valliceps) and Northern cricket frogs (Acris crepitans). A spatial map of locations accurate to within a few centimeters is constructed, and the individual call waveforms are recovered for nine individual animals within a 9 x 9 m(2). These methods work well in low reverberation when there are no reflectors other than the ground. They will require modifications to incorporate multi-path propagation, particularly for the estimation of time-delays.

  15. Blind location and separation of callers in a natural chorus using a microphone array

    PubMed Central

    Jones, Douglas L.; Ratnam, Rama

    2009-01-01

    Male frogs and toads call in dense choruses to attract females. Determining the vocal interactions and spatial distribution of the callers is important for understanding acoustic communication in such assemblies. It has so far proved difficult to simultaneously locate and recover the vocalizations of individual callers. Here a microphone-array technique is developed for blindly locating callers using arrival-time delays at the microphones, estimating their steering-vectors, and recovering the calls with a frequency-domain adaptive beamformer. The technique exploits the time-frequency sparseness of the signal space to recover sources even when there are more sources than sensors. The method is tested with data collected from a natural chorus of Gulf Coast toads (Bufo valliceps) and Northern cricket frogs (Acris crepitans). A spatial map of locations accurate to within a few centimeters is constructed, and the individual call waveforms are recovered for nine individual animals within a 9×9 m2. These methods work well in low reverberation when there are no reflectors other than the ground. They will require modifications to incorporate multi-path propagation, particularly for the estimation of time-delays. PMID:19640054

  16. A biomimetic coupled circuit based microphone array for sound source localization.

    PubMed

    Xu, Huping; Xu, Xiangyuan; Jia, Han; Guan, Luyang; Bao, Ming

    2015-09-01

    An equivalent analog circuit is designed to mimic the coupled ears of the fly Ormia ochracea for sound source localization. This coupled circuit receives two signals with tiny phase difference from a space closed two-microphone array, and produces two signals with obvious intensity difference. The response sensitivity can be adjusted through the coupled circuit parameters. The directional characteristics of the coupled circuit have been demonstrated in the experiment. The miniature microphone array can localize the sound source with low computational burden by using the intensity difference. This system has significant advantages in various applications where the array size is limited. PMID:26428825

  17. A biomimetic coupled circuit based microphone array for sound source localization.

    PubMed

    Xu, Huping; Xu, Xiangyuan; Jia, Han; Guan, Luyang; Bao, Ming

    2015-09-01

    An equivalent analog circuit is designed to mimic the coupled ears of the fly Ormia ochracea for sound source localization. This coupled circuit receives two signals with tiny phase difference from a space closed two-microphone array, and produces two signals with obvious intensity difference. The response sensitivity can be adjusted through the coupled circuit parameters. The directional characteristics of the coupled circuit have been demonstrated in the experiment. The miniature microphone array can localize the sound source with low computational burden by using the intensity difference. This system has significant advantages in various applications where the array size is limited.

  18. Indirect calibration of a large microphone array for in-duct acoustic measurements

    NASA Astrophysics Data System (ADS)

    Leclère, Q.; Pereira, A.; Finez, A.; Souchotte, P.

    2016-08-01

    This paper addresses the problem of in situ calibration of a pin hole-mounted microphone array for in-duct acoustic measurements. One approach is to individually measure the frequency response of each microphone, by submitting the probe to be calibrated and a reference microphone to the same pressure field. Although simple, this task may be very time consuming for large microphone arrays and eventually suffer from lack of access to microphones once they are installed on the test bench. An alternative global calibration procedure is thus proposed in this paper. The approach is based on the fact that the acoustic pressure can be expanded onto an analytically known spatial basis. A projection operator is defined allowing the projection of measurements onto the duct modal basis. The main assumption of the method is that the residual resulting from the difference between actual and projected measurements is mainly dominated by calibration errors. An iterative procedure to estimate the calibration factors of each microphone is proposed and validated through an experimental set-up. In addition, it is shown that the proposed scheme allows an optimization of physical parameters such as the sound speed and parameters associated to the test bench itself, such as the duct radius or the termination reflection coefficient.

  19. Deconvolution for the localization of sound sources using a circular microphone array.

    PubMed

    Tiana-Roig, Elisabet; Jacobsen, Finn

    2013-09-01

    During the last decade, the aeroacoustic community has examined various methods based on deconvolution to improve the visualization of acoustic fields scanned with planar sparse arrays of microphones. These methods assume that the beamforming map in an observation plane can be approximated by a convolution of the distribution of the actual sources and the beamformer's point-spread function, defined as the beamformer's response to a point source. By deconvolving the resulting map, the resolution is improved, and the side-lobes effect is reduced or even eliminated compared to conventional beamforming. Even though these methods were originally designed for planar sparse arrays, in the present study, they are adapted to uniform circular arrays for mapping the sound over 360°. This geometry has the advantage that the beamforming output is practically independent of the focusing direction, meaning that the beamformer's point-spread function is shift-invariant. This makes it possible to apply computationally efficient deconvolution algorithms that consist of spectral procedures in the entire region of interest, such as the deconvolution approach for the mapping of the acoustic sources 2, the Fourier-based non-negative least squares, and the Richardson-Lucy. This investigation examines the matter with computer simulations and measurements. PMID:23967939

  20. 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.

  1. 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)

    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.

  2. Multi-microphone adaptive noise reduction strategies for coordinated stimulation in bilateral cochlear implant devices.

    PubMed

    Kokkinakis, Kostas; Loizou, Philipos C

    2010-05-01

    Bilateral cochlear implant (BI-CI) recipients achieve high word recognition scores in quiet listening conditions. Still, there is a substantial drop in speech recognition performance when there is reverberation and more than one interferers. BI-CI users utilize information from just two directional microphones placed on opposite sides of the head in a so-called independent stimulation mode. To enhance the ability of BI-CI users to communicate in noise, the use of two computationally inexpensive multi-microphone adaptive noise reduction strategies exploiting information simultaneously collected by the microphones associated with two behind-the-ear (BTE) processors (one per ear) is proposed. To this end, as many as four microphones are employed (two omni-directional and two directional) in each of the two BTE processors (one per ear). In the proposed two-microphone binaural strategies, all four microphones (two behind each ear) are being used in a coordinated stimulation mode. The hypothesis is that such strategies combine spatial information from all microphones to form a better representation of the target than that made available with only a single input. Speech intelligibility is assessed in BI-CI listeners using IEEE sentences corrupted by up to three steady speech-shaped noise sources. Results indicate that multi-microphone strategies improve speech understanding in single- and multi-noise source scenarios.

  3. 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.

  4. 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.

  5. Spatial perception of sound fields recorded by spherical microphone arrays with varying spatial resolution.

    PubMed

    Avni, Amir; Ahrens, Jens; Geier, Matthias; Spors, Sascha; Wierstorf, Hagen; Rafaely, Boaz

    2013-05-01

    The area of sound field synthesis has significantly advanced in the past decade, facilitated by the development of high-quality sound-field capturing and re-synthesis systems. Spherical microphone arrays are among the most recently developed systems for sound field capturing, enabling processing and analysis of three-dimensional sound fields in the spherical harmonics domain. In spite of these developments, a clear relation between sound fields recorded by spherical microphone arrays and their perception with a re-synthesis system has not yet been established, although some relation to scalar measures of spatial perception was recently presented. This paper presents an experimental study of spatial sound perception with the use of a spherical microphone array for sound recording and headphone-based binaural sound synthesis. Sound field analysis and processing is performed in the spherical harmonics domain with the use of head-related transfer functions and simulated enclosed sound fields. The effect of several factors, such as spherical harmonics order, frequency bandwidth, and spatial sampling, are investigated by applying the repertory grid technique to the results of the experiment, forming a clearer relation between sound-field capture with a spherical microphone array and its perception using binaural synthesis regarding space, frequency, and additional artifacts. The experimental study clearly shows that a source will be perceived more spatially sharp and more externalized when represented by a binaural stimuli reconstructed with a higher spherical harmonics order. This effect is apparent from low spherical harmonics orders. Spatial aliasing, as a result of sound field capturing with a finite number of microphones, introduces unpleasant artifacts which increased with the degree of aliasing error.

  6. Low frequency sound spatial encoding within an enclosure using spherical microphone arrays.

    PubMed

    Wang, Yan; Chen, Kean

    2016-07-01

    In a spherical coordinate system, interior sound field can be expressed in terms of a series of Fourier-Bessel expansions. The process that obtains the expansion coefficients by use of a microphone array (e.g., a spherical microphone array) is called spatial encoding. Until now spatial encoding has mainly been examined in a free field or a diffuse field which can be modeled as a sum of plane waves. For spatial encoding within an enclosure at low frequencies, special challenges would be encountered in two aspects. First, the expansions are influenced by array configurations. Second, an acoustic mode based model instead of a plane wave based one should be considered. This study focuses on these challenges. Different kinds of array configurations were compared specifically at low frequencies, and the spatial encoding for the cylindrical cavity modes was investigated. It was found that the spherical array with cardioid microphones was optimal when kr<1, the cavity modes can be effectively represented by only a sparse subset of expansion coefficients and a good reproduction can be achieved even outside the spherical valid region, which demonstrates an effective alternative way to describe the cylindrical cavity modes and can be implemented efficiently in practice.

  7. Low frequency sound spatial encoding within an enclosure using spherical microphone arrays.

    PubMed

    Wang, Yan; Chen, Kean

    2016-07-01

    In a spherical coordinate system, interior sound field can be expressed in terms of a series of Fourier-Bessel expansions. The process that obtains the expansion coefficients by use of a microphone array (e.g., a spherical microphone array) is called spatial encoding. Until now spatial encoding has mainly been examined in a free field or a diffuse field which can be modeled as a sum of plane waves. For spatial encoding within an enclosure at low frequencies, special challenges would be encountered in two aspects. First, the expansions are influenced by array configurations. Second, an acoustic mode based model instead of a plane wave based one should be considered. This study focuses on these challenges. Different kinds of array configurations were compared specifically at low frequencies, and the spatial encoding for the cylindrical cavity modes was investigated. It was found that the spherical array with cardioid microphones was optimal when kr<1, the cavity modes can be effectively represented by only a sparse subset of expansion coefficients and a good reproduction can be achieved even outside the spherical valid region, which demonstrates an effective alternative way to describe the cylindrical cavity modes and can be implemented efficiently in practice. PMID:27475162

  8. 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.

  9. 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.

  10. 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

  11. Evaluation of a portable two-microphone adaptive beamforming speech processor with cochlear implant patients.

    PubMed

    van Hoesel, R J; Clark, G M

    1995-04-01

    A two-microphone noise reduction technique was tested with four cochlear implant patients. The noise reduction technique, known as adaptive beamforming (ABF), used signals from only two microphones--one behind each ear--to attenuate sounds not arriving from the direction directly in front of the patient. The algorithm was implemented in a portable digital signal processor, and was compared with a strategy in which the two microphone signals were simply added together (two-microphone broadside strategy). Tests with the four patients were conducted in a soundproof booth with target speech arriving from in front of the patient and multitalker babble noise arriving at 90 deg to the left. Results at 0-dB signal-to-noise level (S/N) showed large improvements in speech intelligibility for all patients, when compared to the two-microphone broadside strategy. Precautions were taken to avoid cancellation of the target speech, and, accordingly, subjective tests showed no deterioration in performance for the adaptive beamformer in quiet. Physical measurement of the directional characteristics of the ABF was made with the microphones placed behind the ears of a KEMAR manikin and in the same acoustic environment as used with the patients. Results showed directional gain of approximately 10 dB when the angle of incidence for interfering noise was shifted more than 20 to 30 deg from directly in front of or behind the manikin.(ABSTRACT TRUNCATED AT 250 WORDS)

  12. Speech enhancement using an equivalent source inverse filtering-based microphone array.

    PubMed

    Bai, Mingsian R; Hur, Kur-Nan; Liu, Ying-Ting

    2010-03-01

    This paper presents a microphone array technique aimed at enhancing speech quality in a reverberant environment. This technique is based on the central idea of single-input-multiple-output equivalent source inverse filtering (SIMO-ESIF). The inverse filters required by the time-domain processing in the technique serve two purposes: de-reverberation and noise reduction. The proposed approach could be useful in telecommunication applications such as automotive hands-free systems, where noise-corrupted speech signal generally needs to be enhanced. SIMO-ESIF can be further enhanced against uncertainties and perturbations by including an adaptive generalized side-lobe canceller. The system is implemented and validated experimentally in a car. As indicated by numerous performance measures, the proposed system proved effective in reducing noise in human speech without significantly compromising the speech quality. In addition, listening tests were conducted to assess the subjective performance of the proposed system, with results processed by using the analysis of variance and a post hoc Fisher's least significant difference (LSD) test to assess the pairwise difference between the noise reduction (NR) algorithms.

  13. 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.

  14. 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.

  15. A measurement method of the flow rate in a pipe using a microphone array

    NASA Astrophysics Data System (ADS)

    Kim, Yong-Beum; Kim, Yang-Hann

    2002-09-01

    A method of measuring the flow rate in a pipe is proposed. The method utilizes one-dimensional acoustic pressure signals that are generated by a loud speaker. A microphone array mounted flush with the inner pipe wall is used to measure the signals. A formula for the flow rate, which is a function of the change of wave number, is derived from a simple mathematical model of sound field in the pipe conveying a viscous fluid. The change of the wave number, which is one of the results caused by flow, is estimated from the recursive relation among the measured microphone array signals. Since measurement errors, due to extraneous measurement noise and mismatch of response characteristics between microphones, exist in the estimated flow rate, a method of compensating the errors is proposed. By using this measurement method, the flow rate can be obtained more accurately than that of our previous method. To verify applicability of the measurement method, numerical simulation and experiments are performed. The estimated flow rates are within 5% error bound. copyright 2002 Acoustical Society of America.

  16. 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

  17. From manikin to microphone arrays development and application of binaural measurement devices

    NASA Astrophysics Data System (ADS)

    Mellert, Volker

    2004-10-01

    Ever since stereophonic systems were designed it was obvious to make the recording by two microphone channels which mimic the aural investigation of the acoustic environment by our two ears. Head-related stereophony aims at a subjectively optimal reproduction of a sound field, in particular with headphones, whereas the multi-channel synthesis of a sound field provides a listening condition which is more independent of the individual listeners acoustic properties (e.g., head size, near-field refraction pattern). Binaural measurement devices are comparatively less complex and costly and therefore in use for about 30 years for sound field investigations, as in concert hall acoustics or in the assessment of environmental and technical sounds. A review is given on the development of head-related stereophony for investigating (mainly) room acoustics. Concepts of future devices for the assessment of technical sound are presented. The classical head-shaped recording system (dummy head) is substituted by beam-forming microphone arrays.

  18. 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

    2014-01-01

    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.

  19. 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

  20. Directional microphone arrays: Reducing wind noise without killing your signal or filling up your disk

    NASA Astrophysics Data System (ADS)

    Zumberge, M. A.; Walker, K. T.; Dewolf, S.; Hedlin, M. A.; Shearer, P. M.; Berger, J.

    2008-12-01

    The bane of infrasound signal detection is the noise generated by the wind. While the physics of the noise is still a subject of investigation, it is clear that sampling pressure at many points over a length scale larger than the spatial coherence length of wind turbulence attenuates the noise. A dense array of microphones can exploit this approach, but this presents different challenges. Two mechanical wind filters using this approach are commonly employed by the nuclear monitoring community (rosette pipe and porous-hoses networks) and attach to a central microphone. To get large wind noise reduction and a low signal detection threshold in the frequency band of interest, these filters require large apertures. However, these wind filters with such large apertures have a poor omnidirectional instrument response for typical infrasound signals because the pressure signal propagates at the speed of sound through the pipes/hoses to the central microphone. A simple, but improved averaging approach would be to instantaneously sample a long length of the infrasound signal wavefront. Optical fiber infrasound sensors (OFIS) are an implementation of this idea. These sensors are compliant sealed tubes wrapped with two optical fibers that integrate pressure change instantaneously along the length of the tube with laser interferometery. Infrasound arrays typically consist of several microbarometers with wind filters separated by distances that provide predictable signal time separations, forming the basis for processing techniques that estimate the phase velocity direction. An analogous approach is to form an array of OFIS arms. The OFIS instrument response is a predictable function of the orientation of the arm with respect to the signal wavefront. An array of many OFIS arms with different azimuths permits at least one OFIS to record any signal without the signal attenuation inherent in equivalently-sized onmi-directional mechanical filters. OFIS arms that are wavefront

  1. 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.

  2. Case study using arrays of infrasonic microphones to detect and locate meteors and meteorites

    NASA Astrophysics Data System (ADS)

    Bedard, A. J., Jr.; Greene, G. E.

    Infrasonic data, in conjunction with surface and aircraft observations, are used to investigate the acoustic signals related to a fireball sighting that were detected on Apr. 22, 1975 by two infrasonic observatories in Colorado. It is deduced that the acoustic energy originated from an explosive interaction of the object with the atmosphere, at an altitude of about 25 km and at a distance of roughly 250 km from the observatories. It is concluded that the infrasonic microphone arrays involved successfully identified the region of fireball termination.

  3. Wavenumber-frequency deconvolution of aeroacoustic microphone phased array data of arbitrary coherence

    NASA Astrophysics Data System (ADS)

    Bahr, Christopher J.; Cattafesta, Louis N.

    2016-11-01

    Deconvolution of aeroacoustic data acquired with microphone phased arrays is a computationally challenging task for distributed sources with arbitrary coherence. A new technique for performing such deconvolution is proposed. This technique relies on analysis of the array data in the wavenumber-frequency domain, allowing for fast convolution and reduced storage requirements when compared to traditional coherent deconvolution. A positive semidefinite constraint for the iterative deconvolution procedure is implemented and shows improved behavior in terms of quantifiable convergence metrics when compared to a standalone covariance inequality constraint. A series of simulations validates the method's ability to resolve coherence and phase angle relationships between partially coherent sources, as well as determines convergence criteria for deconvolution analysis. Simulations for point sources near the microphone phased array show potential for handling such data in the wavenumber-frequency domain. In particular, a physics-based integration boundary calculation is described, and can successfully isolate sources and track the appropriate integration bounds with and without the presence of flow. Magnitude and phase relationships between multiple sources are successfully extracted. Limitations of the deconvolution technique are determined from the simulations, particularly in the context of a simulated acoustic field in a closed test section wind tunnel with strong boundary layer contamination. A final application to a trailing edge noise experiment conducted in an open-jet wind tunnel matches best estimates of acoustic levels from traditional calculation methods and qualitatively assesses the coherence characteristics of the trailing edge noise source.

  4. Effects of a near-field rigid sphere scatterer on the performance of linear microphone array beamformers.

    PubMed

    Hu, Yuxiang; Zhou, Haoran; Lu, Jing; Qiu, Xiaojun

    2016-08-01

    Beamformers enable a microphone array to capture acoustic signals from a sound source with high signal to noise ratio in a noisy environment, and the linear microphone array is of particular importance, in practice, due to its simplicity and easy implementation. A linear microphone array sometimes is used near some scattering objects, which affect its beamforming performance. This paper develops a numerical model with a linear microphone array near a rigid sphere for both far-field plane wave and near-field sources. The effects of the scatterer on two typical beamformers, i.e., the delay-and-sum beamformer and the superdirective beamformer, are investigated by both simulations and experiments. It is found that the directivity factor of both beamformers improves due to the increased equivalent array aperture when the size of the array is no larger than that of the scatter. With the increase of the array size, the directivity factor tends to deteriorate at high frequencies because of the rising side-lobes. When the array size is significantly larger than that of the scatterer, the scattering has hardly any influence on the beamforming performance. PMID:27586725

  5. 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.

  6. Compressive sensing based spinning mode detections by in-duct microphone arrays

    NASA Astrophysics Data System (ADS)

    Yu, Wenjun; Huang, Xun

    2016-05-01

    This paper presents a compressive sensing based experimental method for detecting spinning modes of sound waves propagating inside a cylindrical duct system. This method requires fewer dynamic pressure sensors than the number required by the Shannon-Nyquist sampling theorem so long as the incident waves are sparse in spinning modes. In this work, the proposed new method is firstly validated by preparing some of the numerical simulations with representative set-ups. Then, a duct acoustic testing rig with a spinning mode synthesiser and an in-duct microphone array is built to experimentally demonstrate the new approach. Both the numerical simulations and the experiment results are satisfactory, even when the practical issue of the background noise pollution is taken into account. The approach is beneficial for sensory array tests of silent aeroengines in particular and some other engineering systems with duct acoustics in general.

  7. Compressive sensing based spinning mode detections by in-duct microphone arrays

    NASA Astrophysics Data System (ADS)

    Yu, Wenjun; Huang, Xun

    2016-05-01

    This paper presents a compressive sensing based experimental method for detecting spinning modes of sound waves propagating inside a cylindrical duct system. This method requires fewer dynamic pressure sensors than the number required by the Shannon–Nyquist sampling theorem so long as the incident waves are sparse in spinning modes. In this work, the proposed new method is firstly validated by preparing some of the numerical simulations with representative set-ups. Then, a duct acoustic testing rig with a spinning mode synthesiser and an in-duct microphone array is built to experimentally demonstrate the new approach. Both the numerical simulations and the experiment results are satisfactory, even when the practical issue of the background noise pollution is taken into account. The approach is beneficial for sensory array tests of silent aeroengines in particular and some other engineering systems with duct acoustics in general.

  8. 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.

  9. 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

  10. SoundCompass: a distributed MEMS microphone array-based sensor for sound source localization.

    PubMed

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

    2014-01-23

    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.

  11. 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

  12. Three-dimensional acoustic imaging with planar microphone arrays and compressive sensing

    NASA Astrophysics Data System (ADS)

    Ning, Fangli; Wei, Jingang; Qiu, Lianfang; Shi, Hongbing; Li, Xiaofan

    2016-10-01

    For obtaining super-resolution source maps, we extend compressive sensing (CS) to three-dimensional acoustic imaging. Source maps are simulated with a planar microphone array and a CS algorithm. Comparing the source maps of the CS algorithm with those of the conventional beamformer (CBF) and Tikhonov Regularization (TIKR), we find that the CS algorithm is computationally more effective and can obtain much higher resolution source maps than the CBF and TIKR. The effectiveness of the CS algorithm is analyzed. The CS algorithm can locate the sound sources exactly when the frequency is above 4000 Hz and the signal-to-noise ratio (SNR) is above 12 dB. The location error of the CS algorithm increases as the frequency drops below the threshold, and the errors in location and power increase as SNR decreases. The further from the array the source is, the larger the location error is. The lateral resolution of the CS algorithm is much better than the range resolution. Finally, experimental measurements are conducted in a semi-anechoic room. Two mobile phones are served as sound sources. The results show that the CS algorithm can reconstruct two sound sources near the bottom of the two mobile phones where the speakers are located. The feasibility of the CS algorithm is also validated with the experiment.

  13. 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.

  14. 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.

  15. 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. PMID:14587592

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

    NASA Astrophysics Data System (ADS)

    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 18 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.

  17. 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.

  18. Musical-Noise Analysis in Methods of Integrating Microphone Array and Spectral Subtraction Based on Higher-Order Statistics

    NASA Astrophysics Data System (ADS)

    Takahashi, Yu; Saruwatari, Hiroshi; Shikano, Kiyohiro; Kondo, Kazunobu

    2010-12-01

    We conduct an objective analysis on musical noise generated by two methods of integrating microphone array signal processing and spectral subtraction. To obtain better noise reduction, methods of integrating microphone array signal processing and nonlinear signal processing have been researched. However, nonlinear signal processing often generates musical noise. Since such musical noise causes discomfort to users, it is desirable that musical noise is mitigated. Moreover, it has been recently reported that higher-order statistics are strongly related to the amount of musical noise generated. This implies that it is possible to optimize the integration method from the viewpoint of not only noise reduction performance but also the amount of musical noise generated. Thus, we analyze the simplest methods of integration, that is, the delay-and-sum beamformer and spectral subtraction, and fully clarify the features of musical noise generated by each method. As a result, it is clarified that a specific structure of integration is preferable from the viewpoint of the amount of generated musical noise. The validity of the analysis is shown via a computer simulation and a subjective evaluation.

  19. Investigating Separated Shear Layer Development over an Airfoil with an Imbedded Microphone Array

    NASA Astrophysics Data System (ADS)

    Yarusevych, Serhiy; Gerakopulos, Ryan

    2010-11-01

    At low Reynolds numbers, laminar boundary layer separation on an airfoil often leads to deterioration in airfoil performance and noise emissions. The development of a separated shear layer is governed by laminar to turbulent transition, involving formation of coherent structures. This study highlights the design of a time-resolved surface pressure measurement system capable of estimating salient flow characteristics based on the analysis of surface pressure fluctuations. Wind tunnel experiments were performed for a symmetric NACA 0018 aluminum airfoil model equipped with a total of 95 static pressure taps and 24 microphones. Tests were performed for a range of angles of attack and Reynolds numbers to investigate two flow regimes common to airfoils operating at low Reynolds numbers, namely, flow separation without subsequent reattachment and separation bubble. Experimental results show that the microphones can be utilized to estimate the extent of the separation region and study the development of flow disturbances in the separated shear layer. Using hot wire measurements for validation, it is demonstrated that the microphones can detect the frequency signature of disturbances amplified in the separated shear layer. Further statistical analysis is employed to estimate such important characteristics of the disturbances and coherent structures as spanwise correlation, propagation speed, and phase.

  20. Design and implementation of a space domain spherical microphone array with application to source localization and separation.

    PubMed

    Bai, Mingsian R; Yao, Yueh Hua; Lai, Chang-Sheng; Lo, Yi-Yang

    2016-03-01

    In this paper, four delay-and-sum (DAS) beamformers formulated in the modal domain and the space domain for open and solid spherical apertures are examined through numerical simulations. The resulting beampatterns reveal that the mainlobe of the solid spherical DAS array is only slightly narrower than that of the open array, whereas the sidelobes of the modal domain array are more significant than those of the space domain array due to the discrete approximation of continuous spherical Fourier transformation. To verify the theory experimentally, a three-dimensionally printed spherical array on which 32 micro-electro-mechanical system microphones are mounted is utilized for localization and separation of sound sources. To overcome the basis mismatch problem in signal separation, source localization is first carried out using minimum variance distortionless response beamformer. Next, Tikhonov regularization (TIKR) and compressive sensing (CS) are employed to extract the source signal amplitudes. Simulations and experiments are conducted to validate the proposed spherical array system. Objective perceptual evaluation of speech quality test and a subjective listening test are undertaken in performance evaluation. The experimental results demonstrate better separation quality achieved by the CS approach than by the TIKR approach at the cost of computational complexity. PMID:27036243

  1. Design and implementation of a space domain spherical microphone array with application to source localization and separation.

    PubMed

    Bai, Mingsian R; Yao, Yueh Hua; Lai, Chang-Sheng; Lo, Yi-Yang

    2016-03-01

    In this paper, four delay-and-sum (DAS) beamformers formulated in the modal domain and the space domain for open and solid spherical apertures are examined through numerical simulations. The resulting beampatterns reveal that the mainlobe of the solid spherical DAS array is only slightly narrower than that of the open array, whereas the sidelobes of the modal domain array are more significant than those of the space domain array due to the discrete approximation of continuous spherical Fourier transformation. To verify the theory experimentally, a three-dimensionally printed spherical array on which 32 micro-electro-mechanical system microphones are mounted is utilized for localization and separation of sound sources. To overcome the basis mismatch problem in signal separation, source localization is first carried out using minimum variance distortionless response beamformer. Next, Tikhonov regularization (TIKR) and compressive sensing (CS) are employed to extract the source signal amplitudes. Simulations and experiments are conducted to validate the proposed spherical array system. Objective perceptual evaluation of speech quality test and a subjective listening test are undertaken in performance evaluation. The experimental results demonstrate better separation quality achieved by the CS approach than by the TIKR approach at the cost of computational complexity.

  2. 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.

  3. 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.

  4. 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., Jr.

    2004-01-01

    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.

  5. 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.

  6. Adaptive beamforming for array signal processing in aeroacoustic measurements.

    PubMed

    Huang, Xun; Bai, Long; Vinogradov, Igor; Peers, Edward

    2012-03-01

    Phased microphone arrays have become an important tool in the localization of noise sources for aeroacoustic applications. In most practical aerospace cases the conventional beamforming algorithm of the delay-and-sum type has been adopted. Conventional beamforming cannot take advantage of knowledge of the noise field, and thus has poorer resolution in the presence of noise and interference. Adaptive beamforming has been used for more than three decades to address these issues and has already achieved various degrees of success in areas of communication and sonar. In this work an adaptive beamforming algorithm designed specifically for aeroacoustic applications is discussed and applied to practical experimental data. It shows that the adaptive beamforming method could save significant amounts of post-processing time for a deconvolution method. For example, the adaptive beamforming method is able to reduce the DAMAS computation time by at least 60% for the practical case considered in this work. Therefore, adaptive beamforming can be considered as a promising signal processing method for aeroacoustic measurements.

  7. 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.

  8. A unified systolic array for adaptive beamforming

    SciTech Connect

    Bojanczyk, A.W.; Luk, F.T. )

    1990-04-01

    The authors present a new algorithm and systolic array for adaptive beamforming. The authors algorithm uses only orthogonal transformations and thus should have better numerical properties. The algorithm can be implemented on one single p {times} p triangular array of programmable processors that offers a throughput of one residual element per cycle.

  9. 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…

  10. Research on algorithms for adaptive antenna arrays

    NASA Astrophysics Data System (ADS)

    Widrow, B.; Newman, W.; Gooch, R.; Duvall, K.; Shur, D.

    1981-08-01

    The fundamental efficiency of adaptive algorithms is analyzed. It is found that noise in the adaptive weights increases with convergence speed. This causes loss in mean-square-error performance. Efficiency is considered from the point of view of misadjustment versus speed of convergence. A new version of the LMS algorithm based on Newton's method is analyzed and shown to make maximally efficient use of real-time input data. The performance of this algorithm is not affected by eigenvalue disparity. Practical algorithms can be devised that closely approximate Newton's method. In certain cases, the steepest descent version of LMS performs as well as Newton's method. The efficiency of adaptive algorithms with nonstationary input environments is analyzed where signals, jammers, and background noises can be of a transient and nonstationary nature. A new adaptive filtering method for broadband adaptive beamforming is described which uses both poles and zeros in the adaptive signal filtering paths from the antenna elements to the final array output.

  11. Fiber-optic microphones for battlefield acoustics.

    PubMed

    Wooler, John P F; Crickmore, Roger I

    2007-05-01

    We describe recent work to evaluate the potential performance of an interferometric fiber-optic microphone for battlefield acoustics. The microphone design has high sensitivity and flat response at low frequency and is readily multiplexed. The design is simple and could be manufactured at low cost, which is desirable since in operation the sensors may need to be disposable. Field trial results from an array of microphones are discussed. PMID:17429460

  12. Fiber-optic microphones for battlefield acoustics

    NASA Astrophysics Data System (ADS)

    Wooler, John P. F.; Crickmore, Roger I.

    2007-05-01

    We describe recent work to evaluate the potential performance of an interferometric fiber-optic microphone for battlefield acoustics. The microphone design has high sensitivity and flat response at low frequency and is readily multiplexed. The design is simple and could be manufactured at low cost, which is desirable since in operation the sensors may need to be disposable. Field trial results from an array of microphones are discussed.

  13. 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.

  14. Photorefractive processing for large adaptive phased arrays.

    PubMed

    Weverka, R T; Wagner, K; Sarto, A

    1996-03-10

    An adaptive null-steering phased-array optical processor that utilizes a photorefractive crystal to time integrate the adaptive weights and null out correlated jammers is described. This is a beam-steering processor in which the temporal waveform of the desired signal is known but the look direction is not. The processor computes the angle(s) of arrival of the desired signal and steers the array to look in that direction while rotating the nulls of the antenna pattern toward any narrow-band jammers that may be present. We have experimentally demonstrated a simplified version of this adaptive phased-array-radar processor that nulls out the narrow-band jammers by using feedback-correlation detection. In this processor it is assumed that we know a priori only that the signal is broadband and the jammers are narrow band. These are examples of a class of optical processors that use the angular selectivity of volume holograms to form the nulls and look directions in an adaptive phased-array-radar pattern and thereby to harness the computational abilities of three-dimensional parallelism in the volume of photorefractive crystals. The development of this processing in volume holographic system has led to a new algorithm for phased-array-radar processing that uses fewer tapped-delay lines than does the classic time-domain beam former. The optical implementation of the new algorithm has the further advantage of utilization of a single photorefractive crystal to implement as many as a million adaptive weights, allowing the radar system to scale to large size with no increase in processing hardware.

  15. 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.

  16. 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.

  17. 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

  18. 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.

  19. 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.

  20. Optimized micromirror arrays for adaptive optics

    NASA Astrophysics Data System (ADS)

    Michalicek, M. Adrian; Comtois, John H.; Hetherington, Dale L.

    1999-01-01

    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 μm minimum feature sizes and 0.1 μm mask resolution make it possible to produce dense wiring patterns and irregularly shaped flexures. Likewise, mirror surfaces can 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 2-3 nm to a worst case of roughly 25 nm while boasting active surface areas of 98% or better. Combining the process planarization with a ``planarized-by-design'' 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 optics.

  1. The CHARA Array Adaptive Optics Program

    NASA Astrophysics Data System (ADS)

    Ten Brummelaar, Theo; Che, Xiao; McAlister, Harold A.; Ireland, Michael; Monnier, John D.; Mourard, Denis; Ridgway, Stephen T.; sturmann, judit; Sturmann, Laszlo; Turner, Nils H.; Tuthill, Peter

    2016-01-01

    The CHARA array is an optical/near infrared interferometer consisting of six 1-meter diameter telescopes the longest baseline of which is 331 meters. With sub-millisecond angular resolution, the CHARA array is able to spatially resolve nearby stellar systems to reveal the detailed structures. To improve the sensitivity and scientific throughput, the CHARA array was funded by NSF-ATI in 2011, and by NSF-MRI in 2015, for an upgrade of adaptive optics (AO) systems to all six telescopes. The initial grant covers Phase I of the adaptive optics system, which includes an on-telescope Wavefront Sensor and non-common-path (NCP) error correction. The WFS use a fairly standard Shack-Hartman design and will initially close the tip tilt servo and log wavefront errors for use in data reduction and calibration. The second grant provides the funding for deformable mirrors for each telescope which will be used closed loop to remove atmospheric aberrations from the beams. There are then over twenty reflections after the WFS at the telescopes that bring the light several hundred meters into the beam combining laboratory. Some of these, including the delay line and beam reducing optics, are powered elements, and some of them, in particular the delay lines and telescope Coude optics, are continually moving. This means that the NCP problems in an interferometer are much greater than those found in more standard telescope systems. A second, slow AO system is required in the laboratory to correct for these NCP errors. We will breifly describe the AO system, and it's current status, as well as discuss the new science enabled by the system with a focus on our YSO program.

  2. Microphones for Oral History.

    ERIC Educational Resources Information Center

    Mould, David H.

    1987-01-01

    Discusses factors, such as frequency response and impedance, that need to be considered when purchasing a microphone for interviewing purposes. Examines the various applications and placement of microphones and provides a list of U.S. addresses for the major U.S., European, and Japanese microphone manufacturers. (GEA)

  3. 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.

  4. Graphene Electrostatic Microphone

    NASA Astrophysics Data System (ADS)

    Zhou, Qin; Onishi, Seita; Zettl, A.

    2015-03-01

    We demonstrate a wideband electrostatic graphene microphone displaying flat frequency response over the entire human audible region as well as into the ultrasonic regime. Using the microphone, low-level ultrasonic bat calls are successfully recorded. The microphone can be paired with a similarly constructed electrostatic graphene loudspeaker to create a wideband ultrasonic radio. Materials Sciences Division, Lawrence Berkeley National Laboratory Kavli Energy NanoSciences Institute at the University of California - Berkeley.

  5. Study Of Adaptive-Array Signal Processing

    NASA Technical Reports Server (NTRS)

    Satorius, Edgar H.; Griffiths, Lloyd

    1990-01-01

    Report describes study of adaptive signal-processing techniques for suppression of mutual satellite interference in mobile (on ground)/satellite communication system. Presents analyses and numerical simulations of performances of two approaches to signal processing for suppression of interference. One approach, known as "adaptive side lobe canceling", second called "adaptive temporal processing".

  6. Multilayer graphene condenser microphone

    NASA Astrophysics Data System (ADS)

    Todorović, Dejan; Matković, Aleksandar; Milićević, Marijana; Jovanović, Djordje; Gajić, Radoš; Salom, Iva; Spasenović, Marko

    2015-12-01

    Vibrating membranes are the cornerstone of acoustic technology, forming the backbone of modern loudspeakers and microphones. Acoustic performance of a condenser microphone is derived mainly from the membrane’s size, surface mass and achievable static tension. The widely studied and available nickel has been a dominant membrane material for professional microphones for several decades. In this paper we introduce multilayer graphene as a membrane material for condenser microphones. The graphene device outperforms a high end commercial nickel-based microphone over a significant part of the audio spectrum, with a larger than 10 dB enhancement of sensitivity. Our experimental results are supported with numerical simulations, which also show that a 300 layer thick graphene membrane under maximum tension would offer excellent extension of the frequency range, up to 1 MHz.

  7. 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.

  8. 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

  9. 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

  10. Noise in miniature microphones.

    PubMed

    Thompson, Stephen C; LoPresti, Janice L; Ring, Eugene M; Nepomuceno, Henry G; Beard, John J; Ballad, William J; Carlson, Elmer V

    2002-02-01

    The internal noise spectrum in miniature electret microphones of the type used in the manufacture of hearing aids is measured. An analogous circuit model of the microphone is empirically fit to the measured data and used to determine the important sources of noise within the microphone. The dominant noise source is found to depend on the frequency. Below 40 Hz and above 9 kHz, the dominant source is electrical noise from the amplifier circuit needed to buffer the electrical signal from the microphone diaphragm. Between approximately 40 Hz and 1 kHz, the dominant source is thermal noise originating in the acoustic flow resistance of the small hole pierced in the diaphragm to equalize barometric pressure. Between approximately 1 kHz and 9 kHz, the noise originates in the acoustic flow resistances of sound entering the microphone and propagating to the diaphragm. To further reduce the microphone internal noise in the audio band requires attacking these sources. A prototype microphone having reduced acoustical noise is measured and discussed. PMID:11863188

  11. Noise in miniature microphones

    NASA Astrophysics Data System (ADS)

    Thompson, Stephen C.; Lopresti, Janice L.; Ring, Eugene M.; Nepomuceno, Henry G.; Beard, John J.; Ballad, William J.; Carlson, Elmer V.

    2002-02-01

    The internal noise spectrum in miniature electret microphones of the type used in the manufacture of hearing aids is measured. An analogous circuit model of the microphone is empirically fit to the measured data and used to determine the important sources of noise within the microphone. The dominant noise source is found to depend on the frequency. Below 40 Hz and above 9 kHz, the dominant source is electrical noise from the amplifier circuit needed to buffer the electrical signal from the microphone diaphragm. Between approximately 40 Hz and 1 kHz, the dominant source is thermal noise originating in the acoustic flow resistance of the small hole pierced in the diaphragm to equalize barometric pressure. Between approximately 1 kHz and 9 kHz, the noise originates in the acoustic flow resistances of sound entering the microphone and propagating to the diaphragm. To further reduce the microphone internal noise in the audio band requires attacking these sources. A prototype microphone having reduced acoustical noise is measured and discussed.

  12. Research in large adaptive antenna arrays

    NASA Technical Reports Server (NTRS)

    Berkowitz, R. S.; Dzekov, T.

    1976-01-01

    The feasibility of microwave holographic imaging of targets near the earth using a large random conformal array on the earth's surface and illumination by a CW source on a geostationary satellite is investigated. A geometrical formulation for the illuminator-target-array relationship is applied to the calculation of signal levels resulting from L-band illumination supplied by a satellite similar to ATS-6. The relations between direct and reflected signals are analyzed and the composite resultant signal seen at each antenna element is described. Processing techniques for developing directional beam formation as well as SNR enhancement are developed. The angular resolution and focusing characteristics of a large array covering an approximately circular area on the ground are determined. The necessary relations are developed between the achievable SNR and the size and number of elements in the array. Numerical results are presented for possible air traffic surveillance system. Finally, a simple phase correlation experiment is defined that can establish how large an array may be constructed.

  13. Adaptive and mobile ground sensor array.

    SciTech Connect

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

    2003-12-01

    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 autonomous 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.

  14. Efficient true-time-delay adaptive array processing

    NASA Astrophysics Data System (ADS)

    Wagner, Kelvin H.; Kraut, Shawn; Griffiths, Lloyd J.; Weaver, Samuel P.; Weverka, Robert T.; Sarto, Anthony W.

    1996-11-01

    We present a novel and efficient approach to true-time-delay (TTD) beamforming for large adaptive phased arrays with N elements, for application in radar, sonar, and communication. This broadband and efficient adaptive method for time-delay array processing algorithm decreases the number of tapped delay lines required for N-element arrays form N to only 2, producing an enormous savings in optical hardware, especially for large arrays. This new adaptive system provides the full NM degrees of freedom of a conventional N element time delay beamformer with M taps, each, enabling it to fully and optimally adapt to an arbitrary complex spatio-temporal signal environment that can contain broadband signals, noise, and narrowband and broadband jammers, all of which can arrive from arbitrary angles onto an arbitrarily shaped array. The photonic implementation of this algorithm uses index gratings produce in the volume of photorefractive crystals as the adaptive weights in a TTD beamforming network, 1 or 2 acousto-optic devices for signal injection, and 1 or 2 time-delay-and- integrate detectors for signal extraction. This approach achieves significant reduction in hardware complexity when compared to systems employing discrete RF hardware for the weights or when compared to alternative optical systems that typically use N channel acousto-optic deflectors.

  15. 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.

  16. 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.

  17. True-Time-Delay Adaptive Array Processing Using Photorefractive Crystals

    NASA Astrophysics Data System (ADS)

    Kriehn, G. R.; Wagner, K.

    Radio frequency (RF) signal processing has proven to be a fertile application area when using photorefractive-based, optical processing techniques. This is due to a photorefractive material's capability to record gratings and diffract off these gratings with optically modulated beams that contain a wide RF bandwidth, and include applications such as the bias-free time-integrating correlator [1], adaptive signal processing, and jammer excision, [2, 3, 4]. Photorefractive processing of signals from RF antenna arrays is especially appropriate because of the massive parallelism that is readily achievable in a photorefractive crystal (in which many resolvable beams can be incident on a single crystal simultaneously—each coming from an optical modulator driven by a separate RF antenna element), and because a number of approaches for adaptive array processing using photorefractive crystals have been successfully investigated [5, 6]. In these types of applications, the adaptive weight coefficients are represented by the amplitude and phase of the holographic gratings, and many millions of such adaptive weights can be multiplexed within the volume of a photorefractive crystal. RF modulated optical signals from each array element are diffracted from the adaptively recorded photorefractive gratings (which can be multiplexed either angularly or spatially), and are then coherently combined with the appropriate amplitude weights and phase shifts to effectively steer the angular receptivity pattern of the antenna array toward the desired arriving signal. Likewise, the antenna nulls can also be rotated toward unwanted narrowband jammers for extinction, thereby optimizing the signal-to-interference-plus-noise ratio.

  18. 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.

  19. 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.

  20. Dynamic Adaptive Neural Network Arrays: A Neuromorphic Architecture

    SciTech Connect

    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.

  1. NASA Adaptive Multibeam Phased Array (AMPA): An application study

    NASA Technical Reports Server (NTRS)

    Mittra, R.; Lee, S. W.; Gee, W.

    1982-01-01

    The proposed orbital geometry for the adaptive multibeam phased array (AMPA) communication system is reviewed and some of the system's capabilities and preliminary specifications are highlighted. Typical AMPA user link models and calculations are presented, the principal AMPA features are described, and the implementation of the system is demonstrated. System tradeoffs and requirements are discussed. Recommendations are included.

  2. 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.

  3. A recurrent neural network for adaptive beamforming and array correction.

    PubMed

    Che, Hangjun; Li, Chuandong; He, Xing; Huang, Tingwen

    2016-08-01

    In this paper, a recurrent neural network (RNN) is proposed for solving adaptive beamforming problem. In order to minimize sidelobe interference, the problem is described as a convex optimization problem based on linear array model. RNN is designed to optimize system's weight values in the feasible region which is derived from arrays' state and plane wave's information. The new algorithm is proven to be stable and converge to optimal solution in the sense of Lyapunov. So as to verify new algorithm's performance, we apply it to beamforming under array mismatch situation. Comparing with other optimization algorithms, simulations suggest that RNN has strong ability to search for exact solutions under the condition of large scale constraints.

  4. Adaptive Injection-locking Oscillator Array for RF Spectrum Analysis

    SciTech Connect

    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 achieved by on-board microcontroller.

  5. 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.

  6. 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.

  7. 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.

  8. 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.

  9. Adaptive sensor array algorithm for structural health monitoring of helmet

    NASA Astrophysics Data System (ADS)

    Zou, Xiaotian; Tian, Ye; Wu, Nan; Sun, Kai; Wang, Xingwei

    2011-04-01

    The adaptive neural network is a standard technique used in nonlinear system estimation and learning applications for dynamic models. In this paper, we introduced an adaptive sensor fusion algorithm for a helmet structure health monitoring system. The helmet structure health monitoring system is used to study the effects of ballistic/blast events on the helmet and human skull. Installed inside the helmet system, there is an optical fiber pressure sensors array. After implementing the adaptive estimation algorithm into helmet system, a dynamic model for the sensor array has been developed. The dynamic response characteristics of the sensor network are estimated from the pressure data by applying an adaptive control algorithm using artificial neural network. With the estimated parameters and position data from the dynamic model, the pressure distribution of the whole helmet can be calculated following the Bazier Surface interpolation method. The distribution pattern inside the helmet will be very helpful for improving helmet design to provide better protection to soldiers from head injuries.

  10. 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

  11. Toward a neuromorphic microphone

    PubMed Central

    Smith, Leslie S.

    2015-01-01

    Neuromorphic systems are used in variety of circumstances: as parts of sensory systems, for modeling parts of neural systems and for analog signal processing. In the sensory processing domain, neuromorphic systems can be considered in three parts: pre-transduction processing, transduction itself, and post-transduction processing. Neuromorphic systems include transducers for light, odors, and touch but so far neuromorphic applications in the sound domain have used standard microphones for transduction. We discuss why this is the case and describe what research has been done on neuromorphic approaches to transduction. We make a case for a change of direction toward systems where sound transduction itself has a neuromorphic component. PMID:26578861

  12. 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.

  13. 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.

  14. 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

  15. Hydrogel microphones for stealthy underwater listening

    NASA Astrophysics Data System (ADS)

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

    2016-08-01

    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.

  16. Hydrogel microphones for stealthy underwater listening.

    PubMed

    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

  17. Non-linear, adaptive array processing for acoustic interference suppression.

    PubMed

    Hoppe, Elizabeth; Roan, Michael

    2009-06-01

    A method is introduced where blind source separation of acoustical sources is combined with spatial processing to remove non-Gaussian, broadband interferers from space-time displays such as bearing track recorder displays. This differs from most standard techniques such as generalized sidelobe cancellers in that the separation of signals is not done spatially. The algorithm performance is compared to adaptive beamforming techniques such as minimum variance distortionless response beamforming. Simulations and experiments using two acoustic sources were used to verify the performance of the algorithm. Simulations were also used to determine the effectiveness of the algorithm under various signal to interference, signal to noise, and array geometry conditions. A voice activity detection algorithm was used to benchmark the performance of the source isolation.

  18. Adaptive array for weak interfering signals: Geostationary satellite experiments

    NASA Astrophysics Data System (ADS)

    Steadman, Karl

    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.

  19. Cylindrical Antenna With Partly Adaptive Phased-Array Feed

    NASA Technical Reports Server (NTRS)

    Hussein, Ziad; Hilland, Jeff

    2003-01-01

    A proposed design for a phased-array fed cylindrical-reflector microwave antenna would enable enhancement of the radiation pattern through partially adaptive amplitude and phase control of its edge radiating feed elements. Antennas based on this design concept would be attractive for use in radar (especially synthetic-aperture radar) and other systems that could exploit electronic directional scanning and in which there are requirements for specially shaped radiation patterns, including ones with low side lobes. One notable advantage of this design concept is that the transmitter/ receiver modules feeding all the elements except the edge ones could be identical and, as a result, the antenna would cost less than in the cases of prior design concepts in which these elements may not be identical.

  20. Techniques for radar imaging using a wideband adaptive array

    NASA Astrophysics Data System (ADS)

    Curry, Mark Andrew

    A microwave imaging approach is simulated and validated experimentally that uses a small, wideband adaptive array. The experimental 12-element linear array and microwave receiver uses stepped frequency CW signals from 2--3 GHz and receives backscattered energy from short range objects in a +/-90° field of view. Discone antenna elements are used due to their wide temporal bandwidth, isotropic azimuth beam pattern and fixed phase center. It is also shown that these antennas have very low mutual coupling, which significantly reduces the calibration requirements. The MUSIC spectrum is used as a calibration tool. Spatial resampling is used to correct the dispersion effects, which if not compensated causes severe reduction in detection and resolution for medium and large off-axis angles. Fourier processing provides range resolution and the minimum variance spectral estimate is employed to resolve constant range targets for improved angular resolution. Spatial smoothing techniques are used to generate signal plus interference covariance matrices at each range bin. Clutter affects the angular resolution of the array due to the increase in rank of the signal plus clutter covariance matrix, whereas at the same time the rank of this matrix is reduced for closely spaced scatterers due to signal coherence. A method is proposed to enhance angular resolution in the presence of clutter by an approximate signal subspace projection (ASSP) that maps the received signal space to a lower effective rank approximation. This projection operator has a scalar control parameter that is a function of the signal and clutter amplitude estimates. These operations are accomplished without using eigendecomposition. The low sidelobe levels allow the imaging of the integrated backscattering from the absorber cones in the chamber. This creates a fairly large clutter signature for testing ASSP. We can easily resolve 2 dihedrals placed at about 70% of a beamwidth apart, with a signal to clutter ratio

  1. 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

  2. Evolutionary Adaptive Discovery of Phased Array Sensor Signal Identification

    SciTech Connect

    Timothy R. McJunkin; Milos Manic

    2011-05-01

    Tomography, used to create images of the internal properties and features of an object, from phased array ultasonics is improved through many sophisiticated methonds of post processing of data. One approach used to improve tomographic results is to prescribe the collection of more data, from different points of few so that data fusion might have a richer data set to work from. This approach can lead to rapid increase in the data needed to be stored and processed. It also does not necessarily lead to have the needed data. This article describes a novel approach to utilizing the data aquired as a basis for adapting the sensors focusing parameters to locate more precisely the features in the material: specifically, two evolutionary methods of autofocusing on a returned signal are coupled with the derivations of the forumulas for spatially locating the feature are given. Test results of the two novel methods of evolutionary based focusing (EBF) illustrate the improved signal strength and correction of the position of feature using the optimized focal timing parameters, called Focused Delay Identification (FoDI).

  3. 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.

  4. Adaptive-array Electron Cyclotron Emission diagnostics using data streaming in a Software Defined Radio system

    NASA Astrophysics Data System (ADS)

    Idei, H.; Mishra, K.; Yamamoto, M. K.; Hamasaki, M.; Fujisawa, A.; Nagashima, Y.; Hayashi, Y.; Onchi, T.; Hanada, K.; Zushi, H.; the QUEST Team

    2016-04-01

    Measurement of the Electron Cyclotron Emission (ECE) spectrum is one of the most popular electron temperature diagnostics in nuclear fusion plasma research. A 2-dimensional ECE imaging system was developed with an adaptive-array approach. A radio-frequency (RF) heterodyne detection system with Software Defined Radio (SDR) devices and a phased-array receiver antenna was used to measure the phase and amplitude of the ECE wave. The SDR heterodyne system could continuously measure the phase and amplitude with sufficient accuracy and time resolution while the previous digitizer system could only acquire data at specific times. Robust streaming phase measurements for adaptive-arrayed continuous ECE diagnostics were demonstrated using Fast Fourier Transform (FFT) analysis with the SDR system. The emission field pattern was reconstructed using adaptive-array analysis. The reconstructed profiles were discussed using profiles calculated from coherent single-frequency radiation from the phase array antenna.

  5. Classical and adaptive control algorithms for the solar array pointing system of the Space Station Freedom

    NASA Technical Reports Server (NTRS)

    Ianculescu, G. D.; Klop, J. J.

    1992-01-01

    Classical and adaptive control algorithms for the solar array pointing system of the Space Station Freedom are designed using a continuous rigid body model of the solar array gimbal assembly containing both linear and nonlinear dynamics due to various friction components. The robustness of the design solution is examined by performing a series of sensitivity analysis studies. Adaptive control strategies are examined in order to compensate for the unfavorable effect of static nonlinearities, such as dead-zone uncertainties.

  6. Fiber optic microphone for harsh environment

    NASA Astrophysics Data System (ADS)

    Kots, Alexander; Paritsky, Alexander

    1999-12-01

    Fiber optic microphone is a new device developed on the basis of the new fiber optic technology for measuring distances. Very small in size microphone consists of glass and plastic without any metal. Microphone works very linear in wide frequency and dynamic range in very harsh environment like heavy magnetic, electric, RFI and radioactive fields where no one of known microphones can't work. Microphone may be successfully used in MRI system for audio connection between a patient in MRI equipment and medical personnel outside of it.

  7. 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.

  8. Dynamic Aspects of Cochlear Microphonic Potentials

    NASA Astrophysics Data System (ADS)

    Meenderink, Sebastiaan W. F.; van der Heijden, Marcel

    2011-11-01

    Cochlear microphonic potentials were recorded from the Mongolian gerbil in response to low-frequency auditory stimuli. Provided that contamination of the potentials by the phase-locked neurophonic is avoided, these recordings can be interpreted "as if recorded from a single outer hair cell". It is found that the instantaneous I/O-curves resemble the well-known Boltzmann activation curve. The dynamic aspect of the I/O-curves does reveal hysteresis and a level-dependent gain that is not observed in static measures of these curves. We explore a model that simulates CM generation from hair cell populations, but find it inadequate to reproduce the data. Rather, there seem to be fast, adaptive mechanisms probably at the level of the transduction channels themselves.

  9. Adaptive array technique for differential-phase reflectometry in QUEST

    SciTech Connect

    Idei, H. Hanada, K.; Zushi, H.; Nagata, K.; Mishra, K.; Itado, T.; Akimoto, R.; Yamamoto, M. K.

    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 effect 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.

  10. 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.

  11. Interference cancellation in RF signals using adaptive array techniques

    NASA Astrophysics Data System (ADS)

    Brown, Mark E.

    1990-12-01

    This study investigated the effectiveness of the least means squared (LMS) algorithm against various types of common jammers. The LMS algorithm was implemented using the block oriented systems simulator (BOSS). The LMS algorithm was inserted at the output of a two element antenna array. The array was configured so as to have one-half wavelength spacing. A quadrature hybrid signal structure was used. The array was then tested against a barrage and sweep jammer. The barrage jammer testing consisted of varying each of the three available jammer parameters: power, frequency, and angle of arrival individually. The sweep jammer testing consisted of varying each of the three available jammer parameters; power, sweep frequency and angle of arrival individually. The results of the simulation showed the LMS algorithm in combination with the quadrature hybrid was very effective against both the barrage and sweep jammers. It provided a 55 dB null in the barrage jammer cases and a 50 dB null in the sweep jammer case.

  12. Adaptive Waveform Correlation Detectors for Arrays: Algorithms for Autonomous Calibration

    SciTech Connect

    Ringdal, F; Harris, D B; Dodge, D; Gibbons, S J

    2009-07-23

    Waveform correlation detectors compare a signal template with successive windows of a continuous data stream and report a detection when the correlation coefficient, or some comparable detection statistic, exceeds a specified threshold. Since correlation detectors exploit the fine structure of the full waveform, they are exquisitely sensitive when compared to power (STA/LTA) detectors. The drawback of correlation detectors is that they require complete knowledge of the signal to be detected, which limits such methods to instances of seismicity in which a very similar signal has already been observed by every station used. Such instances include earthquake swarms, aftershock sequences, repeating industrial seismicity, and many other forms of controlled explosions. The reduction in the detection threshold is even greater when the techniques are applied to arrays since stacking can be performed on the individual channel correlation traces to achieve significant array gain. In previous years we have characterized the decrease in detection threshold afforded by correlation detection across an array or network when observations of a previous event provide an adequate template for signals from subsequent events located near the calibration event. Last year we examined two related issues: (1) the size of the source region calibration footprint afforded by a master event, and (2) the use of temporally incoherent detectors designed to detect the gross envelope structure of the signal to extend the footprint. In Case 1, results from the PETROBAR-1 marine refraction profile indicated that array correlation gain was usable at inter-source separations out to one or two wavelengths. In Case 2, we found that incoherent detectors developed from a magnitude 6 event near Svalbard were successful at detecting aftershocks where correlation detectors derived from individual aftershocks were not. Incoherent detectors might provide 'seed' events for correlation detectors that then could

  13. 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.

  14. Adaptive silver films toward bio-array applications

    NASA Astrophysics Data System (ADS)

    Drachev, Vladimir P.; Narasimhan, Meena L.; Yuan, Hsiao-Kuan; Thoreson, Mark D.; Xie, Yong; Davisson, V. J.; Shalaev, Vladimir M.

    2005-03-01

    Adaptive silver films (ASFs) have been studied as a substrate for protein microarrays. Vacuum evaporated silver films fabricated at certain range of evaporation parameters allow fine rearrangement of the silver nanostructure under protein depositions in buffer solution. Proteins restructure and stabilize the ASF to increase the surface-enhanced Raman scattering (SERS) signal from a monolayer of molecules. Preliminary evidence indicates that the adaptive property of the substrates make them appropriate for protein microarray assays. Head-to-head comparisons with two commercial substrates have been performed. Protein binding was quantified on the microarray using the streptavidinCy3/biotinylated goat IgG protein pair. With fluorescence detection, the performance of ASF substrates was comparable with SuperAldehyde and SuperEpoxy substrates. Additionally, the ASF is also a SERS substrate and this provides an additional tool for analysis. It is found that the SERS spectra of the streptavidinCy5 fluorescence reporter bound to true and bound to false sites show distinct difference.

  15. Regularized estimate of the weight vector of an adaptive antenna array

    NASA Astrophysics Data System (ADS)

    Ermolayev, V. T.; Flaksman, A. G.; Sorokin, I. S.

    2013-02-01

    We consider an adaptive antenna array (AAA) with the maximum signal-to-noise ratio (SNR) at the output. The antenna configuration is assumed to be arbitrary. A rigorous analytical solution for the optimal weight vector of the AAA is obtained if the input process is defined by the noise correlation matrix and the useful-signal vector. On the basis of this solution, the regularized estimate of the weight vector is derived by using a limited number of input noise samples, which can be either greater or smaller than the number of array elements. Computer simulation results of adaptive signal processing indicate small losses in the SNR compared with the optimal SNR value. It is shown that the computing complexity of the proposed estimate is proportional to the number of noise samples, the number of external noise sources, and the squared number of array elements.

  16. Multiple wall-reflection effect in adaptive-array differential-phase reflectometry on QUEST

    NASA Astrophysics Data System (ADS)

    Idei, H.; Mishra, K.; Yamamoto, M. K.; Fujisawa, A.; Nagashima, Y.; Hamasaki, M.; Hayashi, Y.; Onchi, T.; Hanada, K.; Zushi, H.; QUEST Team

    2016-01-01

    A phased array antenna and Software-Defined Radio (SDR) heterodyne-detection systems have been developed for adaptive array approaches in reflectometry on the QUEST. In the QUEST device considered as a large oversized cavity, standing wave (multiple wall-reflection) effect was significantly observed with distorted amplitude and phase evolution even if the adaptive array analyses were applied. The distorted fields were analyzed by Fast Fourier Transform (FFT) in wavenumber domain to treat separately the components with and without wall reflections. The differential phase evolution was properly obtained from the distorted field evolution by the FFT procedures. A frequency derivative method has been proposed to overcome the multiple-wall reflection effect, and SDR super-heterodyned components with small frequency difference for the derivative method were correctly obtained using the FFT analysis.

  17. Improvement in adaptive nonuniformity correction method with nonlinear model for infrared focal plane arrays

    NASA Astrophysics Data System (ADS)

    Rui, Lai; Yin-Tang, Yang; Qing, Li; Hui-Xin, Zhou

    2009-09-01

    The scene adaptive nonuniformity correction (NUC) technique is commonly used to decrease the fixed pattern noise (FPN) in infrared focal plane arrays (IRFPA). However, the correction precision of existing scene adaptive NUC methods is reduced by the nonlinear response of IRFPA detectors seriously. In this paper, an improved scene adaptive NUC method that employs "S"-curve model to approximate the detector response is presented. The performance of the proposed method is tested with real infrared video sequence, and the experimental results validate that our method can promote the correction precision considerably.

  18. 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.

  19. Iterative Robust Capon Beamforming with Adaptively Updated Array Steering Vector Mismatch Levels

    PubMed Central

    Sun, Liguo

    2014-01-01

    The performance of the conventional adaptive beamformer is sensitive to the array steering vector (ASV) mismatch. And the output signal-to interference and noise ratio (SINR) suffers deterioration, especially in the presence of large direction of arrival (DOA) error. To improve the robustness of traditional approach, we propose a new approach to iteratively search the ASV of the desired signal based on the robust capon beamformer (RCB) with adaptively updated uncertainty levels, which are derived in the form of quadratically constrained quadratic programming (QCQP) problem based on the subspace projection theory. The estimated levels in this iterative beamformer present the trend of decreasing. Additionally, other array imperfections also degrade the performance of beamformer in practice. To cover several kinds of mismatches together, the adaptive flat ellipsoid models are introduced in our method as tight as possible. In the simulations, our beamformer is compared with other methods and its excellent performance is demonstrated via the numerical examples. PMID:27355008

  20. An adaptive microwave phased array for targeted heating of deep tumours in intact breast: animal study results.

    PubMed

    Fenn, A J; Wolf, G L; Fogle, R M

    1999-01-01

    It has previously been reported in phantoms, that an adaptive radiofrequency phased array can generate deep focused heating distributions without overheating the skin and superficial healthy tissues. The present study involves adaptive microwave phased array hyperthermia tests in animals (rabbits) with and without tumours. The design of the adaptive phased array as applied to the treatment of tumours in intact breast, is described. The adaptive phased array concept uses breast compression and dual-opposing 915 MHz air-cooled waveguide applicators with electronic phase shifters and electric-field feedback, to focus automatically by computer control the microwave radiation in deep tissue. Temperature measurements for a clinical adaptive phased array hyperthermia system demonstrate tissue heating at depth with reduced skin heating.

  1. Optoelectronic implementation of a 256-channel sonar adaptive-array processor.

    PubMed

    Silveira, Paulo E X; Pati, Gour S; Wagner, Kelvin H

    2004-12-10

    We present an optoelectronic implementation of an adaptive-array processor that is capable of performing beam forming and jammer nulling in signals of wide fractional bandwidth that are detected by an array of arbitrary topology. The optical system makes use of a two-dimensional scrolling spatial light modulator to represent an array of input signals in 256 tapped delay lines, two acousto-optic modulators for modulating the feedback error signal, and a photorefractive crystal for representing the adaptive weights as holographic gratings. Gradient-descent learning is used to dynamically adapt the holographic weights to optimally form multiple beams and to null out multiple interference sources, either in the near field or in the far field. Space-integration followed by differential heterodyne detection is used for generating the system's output. The processor is analyzed to show the effects of exponential weight decay on the optimum solution and on the convergence conditions. Several experimental results are presented that validate the system's capacity for broadband beam forming and jammer nulling for linear and circular arrays.

  2. Carbon granule probe microphone for leak detection

    NASA Astrophysics Data System (ADS)

    Parthasarathy, S. P.

    1985-02-01

    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.

  3. Effect of reference microphone location and loudspeaker azimuth on probe tube microphone measurements.

    PubMed

    Ickes, M A; Hawkins, D B; Cooper, W A

    1991-07-01

    The effects of loudspeaker azimuth and reference microphone location on probe tube microphone measures were assessed. The real ear unaided response (REUR), real ear aided response (REAR), and real ear insertion response (REIR) were obtained on a KEMAR. Aided measures were obtained with both a behind-the-ear and an in-the-ear hearing aid. All three measurements were affected by changes in the loudspeaker azimuth and reference microphone location. Responses obtained with a 90 degree loudspeaker azimuth or with the reference microphone located at-the-ear revealed greater disparity than those obtained under other conditions. Most of the differences occurred at frequencies above 2000 Hz, with measurements utilizing the behind-the-ear hearing aid showing greater dispersion. These results suggest that the location of the loudspeaker and the reference microphone are important variables when utilizing probe tube microphone measurements.

  4. 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

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

    NASA Astrophysics Data System (ADS)

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

    2016-10-01

    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.

  6. Adaptive optics wavefront sensors based on photon-counting detector arrays

    NASA Astrophysics Data System (ADS)

    Aull, Brian F.; Schuette, Daniel R.; Reich, Robert K.; Johnson, Robert L.

    2010-07-01

    For adaptive optics systems, there is a growing demand for wavefront sensors that operate at higher frame rates and with more pixels while maintaining low readout noise. Lincoln Laboratory has been investigating Geiger-mode avalanche photodiode arrays integrated with CMOS readout circuits as a potential solution. This type of sensor counts photons digitally within the pixel, enabling data to be read out at high rates without the penalty of readout noise. After a brief overview of adaptive optics sensor development at Lincoln Laboratory, we will present the status of silicon Geigermode- APD technology along with future plans to improve performance.

  7. Fiber Optic Flexural Disk Microphone

    NASA Astrophysics Data System (ADS)

    Brown, David A.; Hofler, T.; Garrett, S. L.

    1989-02-01

    A microphone consisting of a hollow cylinder whose flexible, circular endplates are bonded to pairs of flat spiral wound coils of optical fiber is described. When the endplate/disk is deformed due to a pressure difference, the outer and inner fiber coils experience opposite strains resulting in a "push-pull" optical path length difference which is detected in an all-fiber Michelson interferometer. The close proximity of the interferometric fiber coils, separated by the thin thermally conducting end plate, rejects thermal gradient induced signals. The addition of a second identical endplate and fiber coil pair at the opposite end of the cylinder doubles the acoustic sensitivity while canceling acceleration induced signals. The calculated and measured optical strain of a single plate, single coil sensor using static pressure, acoustic pressure, and acceleration are in good agreement and yield a sensitivity of 21 milliradians per Pascal per meter of optical fiber for an 8.0 cm diameter, 3.0 mm thick plate below its resonance frequency of 3 KHz.

  8. 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.

  9. 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

  10. 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%.

  11. 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. PMID:23319710

  12. Adaptive non-uniformity correction method based on temperature for infrared detector array

    NASA Astrophysics Data System (ADS)

    Zhang, Zhijie; Yue, Song; Hong, Pu; Jia, Guowei; Lei, Bo

    2013-09-01

    The existence of non-uniformities in the responsitivity of the element array is a severe problem typical to common infrared detector. These non-uniformities result in a "curtain'' like fixed pattern noises (FPN) that appear in the image. Some random noise can be restrained by the method kind of equalization method. But the fixed pattern noise can only be removed by .non uniformity correction method. The produce of non uniformities of detector array is the combined action of infrared detector array, readout circuit, semiconductor device performance, the amplifier circuit and optical system. Conventional linear correction techniques require costly recalibration due to the drift of the detector or changes in temperature. Therefore, an adaptive non-uniformity method is needed to solve this problem. A lot factors including detectors and environment conditions variety are considered to analyze and conduct the cause of detector drift. Several experiments are designed to verify the guess. Based on the experiments, an adaptive non-uniformity correction method is put forward in this paper. The strength of this method lies in its simplicity and low computational complexity. Extensive experimental results demonstrate the disadvantage of traditional non-uniformity correct method is conquered by the proposed scheme.

  13. 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.

  14. Color filter array demosaicing: an adaptive progressive interpolation based on the edge type

    NASA Astrophysics Data System (ADS)

    Dong, Qiqi; Liu, Zhaohui

    2015-10-01

    Color filter array (CFA) is one of the key points for single-sensor digital cameras to produce color images. Bayer CFA is the most commonly used pattern. In this array structure, the sampling frequency of green is two times of red or blue, which is consistent with the sensitivity of human eyes to colors. However, each sensor pixel only samples one of three primary color values. To render a full-color image, an interpolation process, commonly referred to CFA demosaicing, is required to estimate the other two missing color values at each pixel. In this paper, we explore an adaptive progressive interpolation based on the edge type algorithm. The proposed demosaicing method consists of two successive steps: an interpolation step that estimates missing color values according to various edges and a post-processing step by iterative interpolation.

  15. Microphonics Measurements in SRF Cavities for RIA

    SciTech Connect

    Kelly, M.P.; Fuerst, Joel; Kedzie, M.; Sharamentov, S.I.; Shepard, Kenneth; Delayen, Jean

    2003-05-01

    Phase stabilization of the RIA drift tube cavities in the presence of microphonics will be a key issue for RIA. Due to the relatively low beam currents (lte 0.5 pmA) required for the RIA driver, microphonics will impact the rf power required to control the cavity fields. Microphonics measurements on the ANL Beta=0.4 single spoke cavity and on the ANL Beta=0.4 two-cell spoke cavity have been performed many at high fields and using a new "cavity resonance monitor" device developed in collaboration with JLAB. Tests on a cold two-cell spoke are the first ever on a multi-cell spoke geometry. The design is essentially a production model with an integral stainless steel housing to hold the liquid helium bath.

  16. Control algorithms of liquid crystal phased arrays used as adaptive optic correctors

    NASA Astrophysics Data System (ADS)

    Dayton, David; Gonglewski, John; Browne, Stephen

    2006-08-01

    Multi-segment liquid crystal phased arrays have been demonstrated as adaptive optics elements for correction of atmospheric turbulence. High speed dual-frequency nematic liquid crystal has sufficient bandwidth to keep up with moderate atmospheric Greenwood frequencies. However the segmented piston correction only spatial nature of the devices requires novel approaches to control algorithms especially when used with Shack-Hartmann wave front sensors. In this presentation we explore approaches and their effects on closed loop Strehl ratios. A Zernike modal based approach has produced the best results. The presentation will contain results from experiments with a Meadowlark optics liquid crystal device.

  17. Probe-microphone measurements with body-worn instruments: loudspeaker and reference microphone effects.

    PubMed

    Nelson, J A; Hawkins, D B

    1994-03-01

    Probe-microphone measurements are typically made with behind-the-ear (BTE) and in-the-ear (ITE) hearing aids with the loudspeaker located at 0-degrees or 45-degrees azimuth at head level and the reference microphone positioned on the head near the hearing aid microphone. With body-worn instruments, these conditions may not accurately reflect in situ hearing aid performance. This study compared the real-ear aided response (REAR) and real-ear insertion response (REIR) for a body-worn hearing aid using the substitution method and an off-line equalization modified pressure method with three different loudspeaker locations (0 degrees and 45 degrees at head level and 0 degrees at body hearing aid level) and two reference microphone positions (over-the-ear [OTE] and next to the body hearing aid microphone). Results indicated that each of the responses was affected by changes in loudspeaker and reference microphone location. If the substitution method measured from 0-degrees azimuth at head level is considered to be the most realistic representation of hearing aid performance, the closest agreement with body-worn hearing aids was obtained with the modified pressure method when the loudspeaker was located at 0-degrees azimuth at head level and the reference microphone was located over the ear. If the clinician uses the modified pressure method and desires to approximate results with the substitution method, correction values are needed for REAR measurements but not for REIR measurements.

  18. 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.

  19. Fiberoptic microphone using a polymeric cavity

    NASA Astrophysics Data System (ADS)

    Wang, Wei-Chih; Soetanto, William; Gu, Kebin

    2011-04-01

    The fabrication and experimental investigation of a fiberoptic microphone is described. The sensing element is a silicon diaphragm with gold thin film coating that is positioned inside a silicone rubber mold at the end of a single mode optical fiber. Thus, a Fabry-Perot interferometer is formed between the inner fiber and the diaphragm. An acoustic pressure change is detected by using the developed microphone. The polymeric cavity and silicon diaphragm-based system exhibits excellent physicochemical properties with a small, simple, low cost, and lightweight design. The system is also electromagnetic interference / radio frequency interference immunity due to the use of fiberoptics.

  20. Array model interpolation and subband iterative adaptive filters applied to beamforming-based acoustic echo cancellation.

    PubMed

    Bai, Mingsian R; Chi, Li-Wen; Liang, Li-Huang; Lo, Yi-Yang

    2016-02-01

    In this paper, an evolutionary exposition is given in regard to the enhancing strategies for acoustic echo cancellers (AECs). A fixed beamformer (FBF) is utilized to focus on the near-end speaker while suppressing the echo from the far end. In reality, the array steering vector could differ considerably from the ideal freefield plane wave model. Therefore, an experimental procedure is developed to interpolate a practical array model from the measured frequency responses. Subband (SB) filtering with polyphase implementation is exploited to accelerate the cancellation process. Generalized sidelobe canceller (GSC) composed of an FBF and an adaptive blocking module is combined with AEC to maximize cancellation performance. Another enhancement is an internal iteration (IIT) procedure that enables efficient convergence in the adaptive SB filters within a sample time. Objective tests in terms of echo return loss enhancement (ERLE), perceptual evaluation of speech quality (PESQ), word recognition rate for automatic speech recognition (ASR), and subjective listening tests are conducted to validate the proposed AEC approaches. The results show that the GSC-SB-AEC-IIT approach has attained the highest ERLE without speech quality degradation, even in double-talk scenarios. PMID:26936567

  1. Device enables calibration of microphones at high sound pressure levels

    NASA Technical Reports Server (NTRS)

    Gillen, A.

    1967-01-01

    Coupling device accurately calibrates microphones at high sound pressure intensities. The system which uses a liquid as the coupling medium can operate in an automatic mode by using a standard microphone as a control sensor. Feedback from the standard microphone controls the calibration signal level.

  2. Transient Microphonic Effects In Superconducting Cavities

    SciTech Connect

    Thomas Powers; G. Davis; Lawrence King

    2005-07-10

    A number of experiments were performed on an installed and operational 5-cell CEBAF cavity to determine the minimum time required to reestablish stable gradient after a cavity window arc trip. Once it was determined that gradient could be reestablished within 10 ms by applying constant power RF signal in and a voltage controlled Oscillator-phase locked loop based system (VCO-PLL), a second experiment was performed to determine if stable gradient could be reestablished using a fixed frequency RF system with a simple gradient based closed loop control system. During this test, instabilities were observed in the cavity forward power signal, which were determined to be microphonic in nature. These microphonic effects were quantified using a cavity resonance monitor and a VCO{_}PLL RF system. Two types of microphonic effects were observed depending on the type of arc event. If the arc occurred in the vacuum space between the warm and cold windows, the transient frequency shift was about 75 Hz peak-to-peak. If the arc occurred on the cavity side of the cold window the transient frequency shift was about 400 Hz peak-to-peak. The background microphonics level for the tested cavity was approximately 30 Hz peak-to-peak. Experimental results, analysis of the resultant klystron power transients, the decay time of the transients, and the implications with respect to fast reset algorithms will be presented.

  3. 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…

  4. 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.

  5. Adaptive scene-based nonuniformity correction method for infrared-focal plane arrays

    NASA Astrophysics Data System (ADS)

    Torres, Sergio N.; Vera, Esteban M.; Reeves, Rodrigo A.; Sobarzo, Sergio K.

    2003-08-01

    The non-uniform response in infrared focal plane array (IRFPA) detectors produces corrupted images with a fixed-pattern noise. In this paper we present an enhanced adaptive scene-based non-uniformity correction (NUC) technique. The method simultaneously estimates detector's parameters and performs the non-uniformity compensation using a neural network approach. In addition, the proposed method doesn't make any assumption on the kind or amount of non-uniformity presented on the raw data. The strength and robustness of the proposed method relies in avoiding the presence of ghosting artifacts through the use of optimization techniques in the parameter estimation learning process, such as: momentum, regularization, and adaptive learning rate. The proposed method has been tested with video sequences of simulated and real infrared data taken with an InSb IRFPA, reaching high correction levels, reducing the fixed pattern noise, decreasing the ghosting, and obtaining an effective frame by frame adaptive estimation of each detector's gain and offset.

  6. High-resolution optical coherence tomography using self-adaptive FFT and array detection

    NASA Astrophysics Data System (ADS)

    Zhao, Yonghua; Chen, Zhongping; Xiang, Shaohua; Ding, Zhihua; Ren, Hongwu; Nelson, J. Stuart; Ranka, Jinendra K.; Windeler, Robert S.; Stentz, Andrew J.

    2001-05-01

    We developed a novel optical coherence tomographic (OCT) system which utilized broadband continuum generation for high axial resolution and a high numeric-aperture (N.A.) Objective for high lateral resolution (<5 micrometers ). The optimal focusing point was dynamically compensated during axial scanning so that it can be kept at the same position as the point that has an equal optical path length as that in the reference arm. This gives us uniform focusing size (<5 mum) at different depths. A new self-adaptive fast Fourier transform (FFT) algorithm was developed to digitally demodulate the interference fringes. The system employed a four-channel detector array for speckle reduction that significantly improved the image's signal-to-noise ratio.

  7. 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.

  8. 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.

  9. Adaptive ground implemented phased array. [evaluation to overcome radio frequency interference characteristics of TDRS VHF return link

    NASA Technical Reports Server (NTRS)

    Smith, J. M.

    1973-01-01

    Tests were conducted to determine the feasibility of using an adaptive ground implemented phased array (AGIPA) to overcome the limitations of the radio frequency interference limited low data Tracking and Data Relay Satellite VHF return link. A feasibility demonstration model of a single user channel AFIPA system was designed, developed, fabricated, and evaluated. By scaling the frequency and aperture geometry from VHF to S-band, the system performance was more easily demonstrated in the controlled environment of an anechoic chamber. The testing procedure employs an AGIPA in which received signals from each element of the array are processed on the ground to form an adaptive, independent, computer controlled beam for each user.

  10. Wideband micromachined microphones with radio frequency detection

    NASA Astrophysics Data System (ADS)

    Hansen, Sean Thomas

    There are many commercial, scientific, and military applications for miniature wideband acoustic sensors, including monitoring the condition or wear of equipment, collecting scientific data, and identifying and localizing military targets. The application of semiconductor micromachining techniques to sensor fabrication has the potential to transform acoustic sensing with small, reproducible, and inexpensive silicon-based microphones. However, such sensors usually suffer from limited bandwidth and from non-uniformities in their frequency response due to squeeze-film damping effects and narrow air gaps. Furthermore, they may be too fragile to be left unattended in a humid or dusty outdoor environment. Silicon microphones that incorporate capacitive micromachined ultrasonic transducer membranes overcome some of the drawbacks of conventional microphones. These micromachined membranes are small and robust enough to be vacuum-sealed, and can withstand atmospheric pressure and submersion in water. In addition, the membrane mechanical response is flat from dc up to ultrasonic frequencies, resulting in a wideband sensor for accurate spectral analysis of acoustic signals. However, a sensitive detection scheme is necessary to detect the small changes in membrane displacement that result from using smaller, stiffer membranes than do conventional microphones. We propose a radio frequency detection technique, in which the capacitive membranes are incorporated into a transmission line. Variations in membrane capacitance due to impinging sound pressure are sensed through the phase variations of a carrier signal that propagates along the line. This dissertation examines the design, fabrication, modeling, and experimental measurements of wideband micromachined microphones using sealed ultrasonic membranes and RF detection. Measurements of fabricated microphones demonstrate less than 0.5 dB variation in their output responses between 0.1 Hz to 100 kHz under electrostatic actuation of

  11. Microphonics in biopotential measurements with capacitive electrodes.

    PubMed

    Luna-Lozano, Pablo S; Pallas-Areny, Ramon

    2010-01-01

    Biopotential measurements with capacitive electrodes do not need any direct contact between electrode and skin, which saves the time devoted to expose and prepare the contact area when measuring with conductive electrodes. However, mechanical vibrations resulting from physiological functions such as respiration and cardiac contraction can change the capacitance of the electrode and affect the recordings. This transformation of mechanical vibrations into undesired electric signals is termed microphonics. We have evaluated microphonics in capacitive ECG recordings obtained from a dressed subject seated on a common chair with electrodes placed on the front side of the backrest of the chair. Depending on the softness of the backrest, the recordings may be clearly affected by the displacement of the thorax back wall due to the respiration and to the heart's mechanical activity.

  12. Graphene electrostatic microphone and ultrasonic radio.

    PubMed

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

    2015-07-21

    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

  13. 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

  14. Locally adaptive regression filter-based infrared focal plane array non-uniformity correction

    NASA Astrophysics Data System (ADS)

    Li, Jia; Qin, Hanlin; Yan, Xiang; Huang, He; Zhao, Yingjuan; Zhou, Huixin

    2015-10-01

    Due to the limitations of the manufacturing technology, the response rates to the same infrared radiation intensity in each infrared detector unit are not identical. As a result, the non-uniformity of infrared focal plane array, also known as fixed pattern noise (FPN), is generated. To solve this problem, correcting the non-uniformity in infrared image is a promising approach, and many non-uniformity correction (NUC) methods have been proposed. However, they have some defects such as slow convergence, ghosting and scene degradation. To overcome these defects, a novel non-uniformity correction method based on locally adaptive regression filter is proposed. First, locally adaptive regression method is used to separate the infrared image into base layer containing main scene information and the detail layer containing detailed scene with FPN. Then, the detail layer sequence is filtered by non-linear temporal filter to obtain the non-uniformity. Finally, the high quality infrared image is obtained by subtracting non-uniformity component from original image. The experimental results show that the proposed method can significantly eliminate the ghosting and the scene degradation. The results of correction are superior to the THPF-NUC and NN-NUC in the aspects of subjective visual and objective evaluation index.

  15. Passive wireless MEMS microphones for biomedical applications.

    PubMed

    Sezen, A S; Sivaramakrishnan, S; Hur, S; Rajamani, R; Robbins, W; Nelson, B J

    2005-11-01

    This paper introduces passive wireless telemetry based operation for high frequency acoustic sensors. The focus is on the development, fabrication, and evaluation of wireless, battery-less SAW-IDT MEMS microphones for biomedical applications. Due to the absence of batteries, the developed sensors are small and as a result of the batch manufacturing strategy are inexpensive which enables their utilization as disposable sensors. A pulse modulated surface acoustic wave interdigital transducer (SAW-IDT) based sensing strategy has been formulated. The sensing strategy relies on detecting the ac component of the acoustic pressure signal only and does not require calibration. The proposed sensing strategy has been successfully implemented on an in-house fabricated SAW-IDT sensor and a variable capacitor which mimics the impedance change of a capacitive microphone. Wireless telemetry distances of up to 5 centimeters have been achieved. A silicon MEMS microphone which will be used with the SAW-IDT device is being microfabricated and tested. The complete passive wireless sensor package will include the MEMS microphone wire-bonded on the SAW substrate and interrogated through an on-board antenna. This work on acoustic sensors breaks new ground by introducing high frequency (i.e., audio frequencies) sensor measurement utilizing SAW-IDT sensors. The developed sensors can be used for wireless monitoring of body sounds in a number of different applications, including monitoring breathing sounds in apnea patients, monitoring chest sounds after cardiac surgery, and for feedback sensing in compression (HFCC) vests used for respiratory ventilation. Another promising application is monitoring chest sounds in neonatal care units where the miniature sensors will minimize discomfort for the newborns.

  16. NORSAR Final Scientific Report Adaptive Waveform Correlation Detectors for Arrays: Algorithms for Autonomous Calibration

    SciTech Connect

    Gibbons, S J; Ringdal, F; Harris, D B

    2009-04-16

    Correlation detection is a relatively new approach in seismology that offers significant advantages in increased sensitivity and event screening over standard energy detection algorithms. The basic concept is that a representative event waveform is used as a template (i.e. matched filter) that is correlated against a continuous, possibly multichannel, data stream to detect new occurrences of that same signal. These algorithms are therefore effective at detecting repeating events, such as explosions and aftershocks at a specific location. This final report summarizes the results of a three-year cooperative project undertaken by NORSAR and Lawrence Livermore National Laboratory. The overall objective has been to develop and test a new advanced, automatic approach to seismic detection using waveform correlation. The principal goal is to develop an adaptive processing algorithm. By this we mean that the detector is initiated using a basic set of reference ('master') events to be used in the correlation process, and then an automatic algorithm is applied successively to provide improved performance by extending the set of master events selectively and strategically. These additional master events are generated by an independent, conventional detection system. A periodic analyst review will then be applied to verify the performance and, if necessary, adjust and consolidate the master event set. A primary focus of this project has been the application of waveform correlation techniques to seismic arrays. The basic procedure is to perform correlation on the individual channels, and then stack the correlation traces using zero-delay beam forming. Array methods such as frequency-wavenumber analysis can be applied to this set of correlation traces to help guarantee the validity of detections and lower the detection threshold. In principle, the deployment of correlation detectors against seismically active regions could involve very large numbers of very specific detectors. To

  17. Self-adapting root-MUSIC algorithm and its real-valued formulation for acoustic vector sensor array

    NASA Astrophysics Data System (ADS)

    Wang, Peng; Zhang, Guo-jun; Xue, Chen-yang; Zhang, Wen-dong; Xiong, Ji-jun

    2012-12-01

    In this paper, based on the root-MUSIC algorithm for acoustic pressure sensor array, a new self-adapting root-MUSIC algorithm for acoustic vector sensor array is proposed by self-adaptive selecting the lead orientation vector, and its real-valued formulation by Forward-Backward(FB) smoothing and real-valued inverse covariance matrix is also proposed, which can reduce the computational complexity and distinguish the coherent signals. The simulation experiment results show the better performance of two new algorithm with low Signal-to-Noise (SNR) in direction of arrival (DOA) estimation than traditional MUSIC algorithm, and the experiment results using MEMS vector hydrophone array in lake trails show the engineering practicability of two new algorithms.

  18. Effect of microphone location in ITE versus BTE hearing aids.

    PubMed

    Gartrell, E L; Church, G T

    1990-07-01

    Sound pressure measurements were made at the hearing aid microphones of 20 subjects with their in-the-ear (ITE) hearing aids and a behind-the-ear (BTE) hearing aid to determine the influence of microphone location on hearing aid input. A probe tube microphone was used to measure the difference in dB SPL between the ITE and BTE microphone locations. ITE microphone location resulted in a maximum high frequency advantage of 9.2 dB in the 2500 to 5000 Hz range. However, the frequency location of this maximal advantage varied a great deal between individuals, precluding the use of a standard ITE microphone correction factor for 2cc coupler to functional gain conversions.

  19. Forward Interference Avoidance in Ad Hoc Communications Using Adaptive Array Antennas

    NASA Astrophysics Data System (ADS)

    Sakaguchi, Tomofumi; Kamiya, Yukihiro; Fujii, Takeo; Suzuki, Yasuo

    Wireless ad hoc communications such as ad hoc networks have been attracting researchers' attention. They are expected to become a key technology for “ubiquitous” networking because of the ability to configure wireless links by nodes autonomously, without any centralized control facilities. Adaptive array antennas (AAA) have been expected to improve the network efficiency by taking advantage of its adaptive beamforming capability. However, it should be noted that AAA is not almighty. Its interference cancellation capability is limited by the degree-of-freedom (DOF) and the angular resolution as a function of the number of element antennas. Application of AAA without attending to these problems can degrade the efficiency of the network. Let us consider wireless ad hoc communication as a target application for AAA, taking advantage of AAA's interference cancellation capability. The low DOF and insufficient resolution will be crucial problems compared to other wireless systems, since there is no centralized facility to control the nodes to avoid interferences in such systems. A number of interferences might impinge on a node from any direction of arrival (DOA) without any timing control. In this paper, focusing on such limitations of AAA applied in ad hoc communications, we propose a new scheme, Forward Interference Avoidance (FIA), using AAA for ad hoc communications in order to avoid problems caused by the limitation of the AAA capability. It enables nodes to avoid interfering with other nodes so that it increases the number of co-existent wireless links. The performance improvement of ad hoc communications in terms of the number of co-existent links is investigated through computer simulations.

  20. Bulk micro-machined wide-band aero-acoustic microphone and its application to acoustic ranging

    NASA Astrophysics Data System (ADS)

    Zhou, Z. J.; Rufer, L.; Salze, E.; Yuldashev, P.; Ollivier, S.; Wong, M.

    2013-10-01

    A wide-band aero-acoustic microphone was realized using a bulk micro-machining process based on the deep reactive-ion etching of silicon. The sensing diaphragm is completely sealed, thus eliminating the loss of low-frequency response resulting from pressure equalization through the release etch-holes present on the diaphragm of a previously reported microphone implemented using a surface-micro-machining process. A dynamic sensitivity of ∼0.33 µV/V/Pa was estimated using an acoustic shockwave (‘N-wave’) generated using a custom-built high-voltage electrical spark-discharge system. This value is comparable to the effective static sensitivity of ∼0.28 µV/V/Pa measured using a commercial nano-indenter system. The response of the microphone is relatively flat from 6 to 500 kHz, with a resonance frequency of ∼715 kHz. An array of three microphones was also constructed and tested to demonstrate the application of these microphones to the localization of high frequency and short duration acoustic sources.

  1. 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

  2. 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.

  3. A self-adaptive thermal switch array for rapid temperature stabilization under various thermal power inputs

    NASA Astrophysics Data System (ADS)

    Geng, Xiaobao; Patel, Pragnesh; Narain, Amitabh; Desheng Meng, Dennis

    2011-08-01

    A self-adaptive thermal switch array (TSA) based on actuation by low-melting-point alloy droplets is reported to stabilize the temperature of a heat-generating microelectromechanical system (MEMS) device at a predetermined range (i.e. the optimal working temperature of the device) with neither a control circuit nor electrical power consumption. When the temperature is below this range, the TSA stays off and works as a thermal insulator. Therefore, the MEMS device can quickly heat itself up to its optimal working temperature during startup. Once this temperature is reached, TSA is automatically turned on to increase the thermal conductance, working as an effective thermal spreader. As a result, the MEMS device tends to stay at its optimal working temperature without complex thermal management components and the associated parasitic power loss. A prototype TSA was fabricated and characterized to prove the concept. The stabilization temperatures under various power inputs have been studied both experimentally and theoretically. Under the increment of power input from 3.8 to 5.8 W, the temperature of the device increased only by 2.5 °C due to the stabilization effect of TSA.

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

    PubMed

    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

  5. 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.

  6. 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

  7. A High-Speed Adaptively-Biased Current-to-Current Front-End for SSPM Arrays

    NASA Astrophysics Data System (ADS)

    Zheng, Bob; Walder, Jean-Pierre; Lippe, Henrik vonder; Moses, William; Janecek, Martin

    Solid-state photomultiplier (SSPM) arrays are an interesting technology for use in PET detector modules due to their low cost, high compactness, insensitivity to magnetic fields, and sub-nanosecond timing resolution. However, the large intrinsic capacitance of SSPM arrays results in RC time constants that can severely degrade the response time, which leads to a trade-off between array size and speed. Instead, we propose a front-end that utilizes an adaptively biased current-to-current converter that minimizes the resistance seen by the SSPM array, thus preserving the timing resolution for both large and small arrays. This enables the use of large SSPM arrays with resistive networks, which creates position information and minimizes the number of outputs for compatibility with general PET multiplexing schemes. By tuning the bias of the feedback amplifier, the chip allows for precise control of the close-loop gain, ensuring stability and fast operation from loads as small as 50pF to loads as large as 1nF. The chip has 16 input channels, and 4 outputs capable of driving 100 n loads. The power consumption is 12mW per channel and 360mW for the entire chip. The chip has been designed and fabricated in an AMS 0.35um high-voltage technology, and demonstrates a fast rise-time response and low noise performances.

  8. 77 FR 2087 - Certain Silicon Microphone Packages and Products Containing Same; Institution of Investigation

    Federal Register 2010, 2011, 2012, 2013, 2014

    2012-01-13

    ... COMMISSION Certain Silicon Microphone Packages and Products Containing Same; Institution of Investigation... importation, and the sale within the United States after importation of certain silicon microphone packages... after importation of certain silicon microphone packages and products containing same that infringe...

  9. 76 FR 78042 - Certain Silicon Microphone Packages and Products Containing Same Receipt of Complaint...

    Federal Register 2010, 2011, 2012, 2013, 2014

    2011-12-15

    ... COMMISSION Certain Silicon Microphone Packages and Products Containing Same Receipt of Complaint... complaint entitled In Re Certain Silicon Microphone Packages and Products Containing Same, DN 2864; the... importation of certain silicon microphone packages and products containing same. The complaint names...

  10. 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.

  11. Shooter Localization in Wireless Microphone Networks

    NASA Astrophysics Data System (ADS)

    Lindgren, David; Wilsson, Olof; Gustafsson, Fredrik; Habberstad, Hans

    2010-12-01

    Shooter localization in a wireless network of microphones is studied. Both the acoustic muzzle blast (MB) from the gunfire and the ballistic shock wave (SW) from the bullet can be detected by the microphones and considered as measurements. The MB measurements give rise to a standard sensor network problem, similar to time difference of arrivals in cellular phone networks, and the localization accuracy is good, provided that the sensors are well synchronized compared to the MB detection accuracy. The detection times of the SW depend on both shooter position and aiming angle and may provide additional information beside the shooter location, but again this requires good synchronization. We analyze the approach to base the estimation on the time difference of MB and SW at each sensor, which becomes insensitive to synchronization inaccuracies. Cramér-Rao lower bound analysis indicates how a lower bound of the root mean square error depends on the synchronization error for the MB and the MB-SW difference, respectively. The estimation problem is formulated in a separable nonlinear least squares framework. Results from field trials with different types of ammunition show excellent accuracy using the MB-SW difference for both the position and the aiming angle of the shooter.

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

    SciTech Connect

    Cochran, J.R.; McDonald, J.R.; Russell, R.J.; Robertson, R.; Hensel, E.

    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).

  13. A new microphonics measurement method for superconducting RF cavities

    NASA Astrophysics Data System (ADS)

    Gao, Zheng; He, Yuan; Chang, Wei; Powers, Tom; Yue, Wei-ming; Zhu, Zheng-long; Chen, Qi

    2014-12-01

    Mechanical vibrations of the superconducting cavity, also known as microphonics, cause shifts in the resonant frequency of the cavity. In addition to requiring additional RF power, these frequency shifts can contribute to errors in the closed loop phase and amplitude regulation. In order to better understand these effects, a new microphonics measurement method was developed, and the method was successfully used to measure microphonics on the half-wave superconducting cavity when it was operated in a production style cryostat. The test cryostat held a single β=0.1 half-wave cavity which was operated at 162.5 MHz Yue et al. (2013) and Wang et al. (2013) [1,2]. It is the first time that the National Instruments PXIe-5641R intermediate frequency transceiver has been used for microphonics measurements in superconducting cavities. The new microphonics measurement method and results will be shown and analyzed in this paper.

  14. A new microphonics measurement method for superconducting RF cavities

    SciTech Connect

    Gao, Zheng; He, Yuan; Chang, Wei; Powers, Tom; Yue, Wei-ming; Zhu, Zheng-long; Chen, Qi

    2014-09-01

    Mechanical vibrations of the superconducting cavity, also known as microphonics, cause shifts in the resonant frequency of the cavity. In addition to requiring additional RF power, these frequency shifts can contribute to errors in the closed loop phase and amplitude regulation. In order to better understand these effects, a new microphonics measurement method was developed, and the method was successfully used to measure microphonics on the half-wave superconducting cavity when it was operated in a production style cryostat. The test cryostat held a single β=0.1 half-wave cavity which was operated at 162.5 MHz [1] and [2]. It's the first time that the National Instruments PXIe-5641R intermediate frequency transceiver has been used for microphonics measurements in superconducting cavities. The new microphonics measurement method and results will be shown and analyzed in this paper.

  15. A Dual-Microphone Speech Enhancement Algorithm Based on the Coherence Function

    PubMed Central

    2011-01-01

    A novel dual-microphone speech enhancement technique is proposed in the present paper. The technique utilizes the coherence between the target and noise signals as a criterion for noise reduction and can be generally applied to arrays with closely-spaced microphones, where noise captured by the sensors is highly correlated. The proposed algorithm is simple to implement and requires no estimation of noise statistics. In addition, it offers the capability of coping with multiple interfering sources that might be located at different azimuths. The proposed algorithm was evaluated with normal hearing listeners using intelligibility listening tests and compared against a well-established beamforming algorithm. Results indicated large gains in speech intelligibility relative to the baseline (front microphone) algorithm in both single and multiple-noise source scenarios. The proposed algorithm was found to yield substantially higher intelligibility than that obtained by the beamforming algorithm, particularly when multiple noise sources or competing talker(s) were present. Objective quality evaluation of the proposed algorithm also indicated significant quality improvement over that obtained by the beamforming algorithm. The intelligibility and quality benefits observed with the proposed coherence-based algorithm make it a viable candidate for hearing aid and cochlear implant devices. PMID:22207823

  16. Virtual Microphones for Multichannel Audio Resynthesis

    NASA Astrophysics Data System (ADS)

    Mouchtaris, Athanasios; Narayanan, Shrikanth S.; Kyriakakis, Chris

    2003-12-01

    Multichannel audio offers significant advantages for music reproduction, including the ability to provide better localization and envelopment, as well as reduced imaging distortion. On the other hand, multichannel audio is a demanding media type in terms of transmission requirements. Often, bandwidth limitations prohibit transmission of multiple audio channels. In such cases, an alternative is to transmit only one or two reference channels and recreate the rest of the channels at the receiving end. Here, we propose a system capable of synthesizing the required signals from a smaller set of signals recorded in a particular venue. These synthesized "virtual" microphone signals can be used to produce multichannel recordings that accurately capture the acoustics of that venue. Applications of the proposed system include transmission of multichannel audio over the current Internet infrastructure and, as an extension of the methods proposed here, remastering existing monophonic and stereophonic recordings for multichannel rendering.

  17. Adapting physically complete models to vehicle-based EMI array sensor data: data inversion and discrimination studies

    NASA Astrophysics Data System (ADS)

    Shubitidze, Fridon; Miller, Jonathan S.; Schultz, Gregory M.; Marble, Jay A.

    2010-04-01

    This paper reports vehicle based electromagnetic induction (EMI) array sensor data inversion and discrimination results. Recent field studies show that EMI arrays, such as the Minelab Single Transmitter Multiple Receiver (STMR), and the Geophex GEM-5 EMI array, provide a fast and safe way to detect subsurface metallic targets such as landmines, unexploded ordnance (UXO) and buried explosives. The array sensors are flexible and easily adaptable for a variety of ground vehicles and mobile platforms, which makes them very attractive for safe and cost effective detection operations in many applications, including but not limited to explosive ordnance disposal and humanitarian UXO and demining missions. Most state-of-the-art EMI arrays measure the vertical or full vector field, or gradient tensor fields and utilize them for real-time threat detection based on threshold analysis. Real field practice shows that the threshold-level detection has high false alarms. One way to reduce these false alarms is to use EMI numerical techniques that are capable of inverting EMI array data in real time. In this work a physically complete model, known as the normalized volume/surface magnetic sources (NV/SMS) model is adapted to the vehicle-based EMI array, such as STMR and GEM-5, data. The NV/SMS model can be considered as a generalized volume or surface dipole model, which in a special limited case coincides with an infinitesimal dipole model approach. According to the NV/SMS model, an object's response to a sensor's primary field is modeled mathematically by a set of equivalent magnetic dipoles, distributed inside the object (i.e. NVMS) or over a surface surrounding the object (i.e. NSMS). The scattered magnetic field of the NSMS is identical to that produced by a set of interacting magnetic dipoles. The amplitudes of the magnetic dipoles are normalized to the primary magnetic field, relating induced magnetic dipole polarizability and the primary magnetic field. The magnitudes of

  18. 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.

  19. 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.

  20. 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.

  1. 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.

  2. 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.

  3. 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.

  4. Study on a flexoelectric microphone using barium strontium titanate

    NASA Astrophysics Data System (ADS)

    Kwon, S. R.; Huang, W. B.; Zhang, S. J.; Yuan, F. G.; Jiang, X. N.

    2016-04-01

    In this study, a flexoelectric microphone was, for the first time, designed and fabricated in a bridge structure using barium strontium titanate (Ba0.65Sr0.35TiO3) ceramic and tested afterwards. The prototyped flexoelectric microphone consists of a 1.5 mm  ×  768 μm  ×  50 μm BST bridge structure and a silicon substrate with a cavity. The sensitivity and resonance frequency were designed to be 0.92 pC/Pa and 98.67 kHz, respectively. The signal to noise ratio was measured to be 74 dB. The results demonstrate that the flexoelectric microphone possesses high sensitivity and a wide working frequency range simultaneously, suggesting that flexoelectricity could be an excellent alternative sensing mechanism for microphone applications.

  5. 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.

  6. Shape optimization of pressure gradient microphones

    NASA Technical Reports Server (NTRS)

    Norum, T. D.; Seiner, J. M.

    1977-01-01

    Recently developed finite element computer programs were utilized to investigate the influence of the shape of a body on its scattering field with the aim of determining the optimal shape for a Pressure Gradient Microphone (PGM). Circular cylinders of various aspect ratios were evaluated to choose the length to diameter ratio best suited for a dual element PGM application. Alterations of the basic cylindrical shape by rounding the edges and recessing at the centerline were also studied. It was found that for a + or - 1 db deviation from a linear pressure gradient response, a circular cylinder of aspect ratio near 0.5 was most suitable, yielding a useful upper frequency corresponding to ka = 1.8. The maximum increase in this upper frequency limit obtained through a number of shape alterations was only about 20 percent. An initial experimental evaluation of a single element cylindrical PGM of aspect ratio 0.18 utilizing a piezoresistive type sensor was also performed and is compared to the analytical results.

  7. Faraday-effect light-valve arrays for adaptive optical instruments

    SciTech Connect

    Hirleman, E.D.; Dellenback, P.A.

    1987-01-01

    The ability to adapt to a range of measurement conditions by autonomously configuring software or hardware on-line will be an important attribute of next-generation intelligent sensors. This paper reviews the characteristics of spatial light modulators (SLM) with an emphasis on potential integration into adaptive optical instruments. The paper focuses on one type of SLM, a magneto-optic device based on the Faraday effect. Finally, the integration of the Faraday-effect SLM into a laser-diffraction particle-sizing instrument giving it some ability to adapt to the measurement context is discussed.

  8. Precise calibration of a GNSS antenna array for adaptive beamforming applications.

    PubMed

    Daneshmand, Saeed; Sokhandan, Negin; Zaeri-Amirani, Mohammad; Lachapelle, Gérard

    2014-05-30

    The use of global navigation satellite system (GNSS) antenna arrays for applications such as interference counter-measure, attitude determination and signal-to-noise ratio (SNR) enhancement is attracting significant attention. However, precise antenna array calibration remains a major challenge. This paper proposes a new method for calibrating a GNSS antenna array using live signals and an inertial measurement unit (IMU). Moreover, a second method that employs the calibration results for the estimation of steering vectors is also proposed. These two methods are applied to the receiver in two modes, namely calibration and operation. In the calibration mode, a two-stage optimization for precise calibration is used; in the first stage, constant uncertainties are estimated while in the second stage, the dependency of each antenna element gain and phase patterns to the received signal direction of arrival (DOA) is considered for refined calibration. In the operation mode, a low-complexity iterative and fast-converging method is applied to estimate the satellite signal steering vectors using the calibration results. This makes the technique suitable for real-time applications employing a precisely calibrated antenna array. The proposed calibration method is applied to GPS signals to verify its applicability and assess its performance. Furthermore, the data set is used to evaluate the proposed iterative method in the receiver operation mode for two different applications, namely attitude determination and SNR enhancement.

  9. Precise calibration of a GNSS antenna array for adaptive beamforming applications.

    PubMed

    Daneshmand, Saeed; Sokhandan, Negin; Zaeri-Amirani, Mohammad; Lachapelle, Gérard

    2014-01-01

    The use of global navigation satellite system (GNSS) antenna arrays for applications such as interference counter-measure, attitude determination and signal-to-noise ratio (SNR) enhancement is attracting significant attention. However, precise antenna array calibration remains a major challenge. This paper proposes a new method for calibrating a GNSS antenna array using live signals and an inertial measurement unit (IMU). Moreover, a second method that employs the calibration results for the estimation of steering vectors is also proposed. These two methods are applied to the receiver in two modes, namely calibration and operation. In the calibration mode, a two-stage optimization for precise calibration is used; in the first stage, constant uncertainties are estimated while in the second stage, the dependency of each antenna element gain and phase patterns to the received signal direction of arrival (DOA) is considered for refined calibration. In the operation mode, a low-complexity iterative and fast-converging method is applied to estimate the satellite signal steering vectors using the calibration results. This makes the technique suitable for real-time applications employing a precisely calibrated antenna array. The proposed calibration method is applied to GPS signals to verify its applicability and assess its performance. Furthermore, the data set is used to evaluate the proposed iterative method in the receiver operation mode for two different applications, namely attitude determination and SNR enhancement. PMID:24887043

  10. Precise Calibration of a GNSS Antenna Array for Adaptive Beamforming Applications

    PubMed Central

    Daneshmand, Saeed; Sokhandan, Negin; Zaeri-Amirani, Mohammad; Lachapelle, Gérard

    2014-01-01

    The use of global navigation satellite system (GNSS) antenna arrays for applications such as interference counter-measure, attitude determination and signal-to-noise ratio (SNR) enhancement is attracting significant attention. However, precise antenna array calibration remains a major challenge. This paper proposes a new method for calibrating a GNSS antenna array using live signals and an inertial measurement unit (IMU). Moreover, a second method that employs the calibration results for the estimation of steering vectors is also proposed. These two methods are applied to the receiver in two modes, namely calibration and operation. In the calibration mode, a two-stage optimization for precise calibration is used; in the first stage, constant uncertainties are estimated while in the second stage, the dependency of each antenna element gain and phase patterns to the received signal direction of arrival (DOA) is considered for refined calibration. In the operation mode, a low-complexity iterative and fast-converging method is applied to estimate the satellite signal steering vectors using the calibration results. This makes the technique suitable for real-time applications employing a precisely calibrated antenna array. The proposed calibration method is applied to GPS signals to verify its applicability and assess its performance. Furthermore, the data set is used to evaluate the proposed iterative method in the receiver operation mode for two different applications, namely attitude determination and SNR enhancement. PMID:24887043

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

    SciTech Connect

    Joyner, Dominique; Fortney, Julian; Chakraborty, Romy; Hazen, Terry

    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 reducing 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 not

  12. Outline of a multiple-access communication network based on adaptive arrays

    NASA Technical Reports Server (NTRS)

    Zohar, S.

    1982-01-01

    Attention is given to a narrow-band communication system consisting of a central station trying to receive signals simultaneously from K spatially distinct mobile users sharing the same frequencies. One example of such a system is a group of aircraft and ships transmitting messages to a communication satellite. A reasonable approach to such a multiple access system may be based on equipping the central station with an n-element antenna array where n is equal to or greater than K. The array employs K sets of n weights to segregate the signals received from the K users. The weights are determined by direct computation based on position information transmitted by the users. A description is presented of an improved technique which makes it possible to reduce significantly the number of required computer operations in comparison to currently known techniques.

  13. 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).

  14. Guided filter and adaptive learning rate based non-uniformity correction algorithm for infrared focal plane array

    NASA Astrophysics Data System (ADS)

    Sheng-Hui, Rong; Hui-Xin, Zhou; Han-Lin, Qin; Rui, Lai; Kun, Qian

    2016-05-01

    Imaging non-uniformity of infrared focal plane array (IRFPA) behaves as fixed-pattern noise superimposed on the image, which affects the imaging quality of infrared system seriously. In scene-based non-uniformity correction methods, the drawbacks of ghosting artifacts and image blurring affect the sensitivity of the IRFPA imaging system seriously and decrease the image quality visibly. This paper proposes an improved neural network non-uniformity correction method with adaptive learning rate. On the one hand, using guided filter, the proposed algorithm decreases the effect of ghosting artifacts. On the other hand, due to the inappropriate learning rate is the main reason of image blurring, the proposed algorithm utilizes an adaptive learning rate with a temporal domain factor to eliminate the effect of image blurring. In short, the proposed algorithm combines the merits of the guided filter and the adaptive learning rate. Several real and simulated infrared image sequences are utilized to verify the performance of the proposed algorithm. The experiment results indicate that the proposed algorithm can not only reduce the non-uniformity with less ghosting artifacts but also overcome the problems of image blurring in static areas.

  15. Analytical approach to transforming filter design for sound field recording and reproduction using circular arrays with a spherical baffle.

    PubMed

    Koyama, Shoichi; Furuya, Ken'ichi; Wakayama, Keigo; Shimauchi, Suehiro; Saruwatari, Hiroshi

    2016-03-01

    A sound field recording and reproduction method using circular arrays of microphones and loudspeakers with a spherical baffle is proposed. The spherical baffle is an acoustically rigid object on which the microphone array is mounted. The driving signals of the loudspeakers must be obtained from the signals received by the microphones. A transform filter for this signal conversion is analytically derived, which is referred to as the wave field reconstruction filter. The proposed method using a spherical baffle is compared with methods using an array of directional microphones and a microphone array mounted on a cylindrical baffle. Numerical simulations indicated that the proposed method is advantageous for sound field recording and reproduction compared with the other two methods. The results of measurement experiments in a real environment are also demonstrated. PMID:27036240

  16. A comparison of deghosting techniques in adaptive nonuniformity correction for IR focal-plane array systems

    NASA Astrophysics Data System (ADS)

    Rossi, Alessandro; Diani, Marco; Corsini, Giovanni

    2010-10-01

    Focal-plane array (FPA) IR systems are affected by fixed-pattern noise (FPN) which is caused by the nonuniformity of the responses of the detectors that compose the array. Due to the slow temporal drift of FPN, several scene-based nonuniformity correction (NUC) techniques have been developed that operate calibration during the acquisition only by means of the collected data. Unfortunately, such algorithms are affected by a collateral damaging problem: ghosting-like artifacts are generated by the edges in the scene and appear as a reverse image in the original position. In this paper, we compare the performance of representative methods for reducing ghosting. Such methods relate to the least mean square (LMS)-based NUC algorithm proposed by D.A. Scribner. In particular, attention is focused on a recently proposed technique which is based on the computation of the temporal statistics of the error signal in the aforementioned LMS-NUC algorithm. In this work, the performances of the deghosting techniques have been investigated by means of IR data corrupted with simulated nonuniformity noise over the detectors of the FPA. Finally, we have made some considerations on the computational aspect which is a challenging task for the employment of such techniques in real-time systems.

  17. Improved Open-Microphone Speech Recognition

    NASA Astrophysics Data System (ADS)

    Abrash, Victor

    2002-12-01

    Many current and future NASA missions make extreme demands on mission personnel both in terms of work load and in performing under difficult environmental conditions. In situations where hands are impeded or needed for other tasks, eyes are busy attending to the environment, or tasks are sufficiently complex that ease of use of the interface becomes critical, spoken natural language dialog systems offer unique input and output modalities that can improve efficiency and safety. They also offer new capabilities that would not otherwise be available. For example, many NASA applications require astronauts to use computers in micro-gravity or while wearing space suits. Under these circumstances, command and control systems that allow users to issue commands or enter data in hands-and eyes-busy situations become critical. Speech recognition technology designed for current commercial applications limits the performance of the open-ended state-of-the-art dialog systems being developed at NASA. For example, today's recognition systems typically listen to user input only during short segments of the dialog, and user input outside of these short time windows is lost. Mistakes detecting the start and end times of user utterances can lead to mistakes in the recognition output, and the dialog system as a whole has no way to recover from this, or any other, recognition error. Systems also often require the user to signal when that user is going to speak, which is impractical in a hands-free environment, or only allow a system-initiated dialog requiring the user to speak immediately following a system prompt. In this project, SRI has developed software to enable speech recognition in a hands-free, open-microphone environment, eliminating the need for a push-to-talk button or other signaling mechanism. The software continuously captures a user's speech and makes it available to one or more recognizers. By constantly monitoring and storing the audio stream, it provides the spoken

  18. Improved Open-Microphone Speech Recognition

    NASA Technical Reports Server (NTRS)

    Abrash, Victor

    2002-01-01

    Many current and future NASA missions make extreme demands on mission personnel both in terms of work load and in performing under difficult environmental conditions. In situations where hands are impeded or needed for other tasks, eyes are busy attending to the environment, or tasks are sufficiently complex that ease of use of the interface becomes critical, spoken natural language dialog systems offer unique input and output modalities that can improve efficiency and safety. They also offer new capabilities that would not otherwise be available. For example, many NASA applications require astronauts to use computers in micro-gravity or while wearing space suits. Under these circumstances, command and control systems that allow users to issue commands or enter data in hands-and eyes-busy situations become critical. Speech recognition technology designed for current commercial applications limits the performance of the open-ended state-of-the-art dialog systems being developed at NASA. For example, today's recognition systems typically listen to user input only during short segments of the dialog, and user input outside of these short time windows is lost. Mistakes detecting the start and end times of user utterances can lead to mistakes in the recognition output, and the dialog system as a whole has no way to recover from this, or any other, recognition error. Systems also often require the user to signal when that user is going to speak, which is impractical in a hands-free environment, or only allow a system-initiated dialog requiring the user to speak immediately following a system prompt. In this project, SRI has developed software to enable speech recognition in a hands-free, open-microphone environment, eliminating the need for a push-to-talk button or other signaling mechanism. The software continuously captures a user's speech and makes it available to one or more recognizers. By constantly monitoring and storing the audio stream, it provides the spoken

  19. 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.

  20. Digital Cavity Resonance Monitor, alternative method of measuring cavity microphonics

    SciTech Connect

    Tomasz Plawski; G. Davis; Hai Dong; J. Hovater; John Musson; Thomas Powers

    2005-09-20

    As is well known, mechanical vibration or microphonics in a cryomodule causes the cavity resonance frequency to change at the vibration frequency. One way to measure the cavity microphonics is to drive the cavity with a Phase Locked Loop. Measurement of the instantaneous frequency or PLL error signal provides information about the cavity microphonic frequencies. Although the PLL error signal is available directly, precision frequency measurements require additional instrumentation, a Cavity Resonance Monitor (CRM). The analog version of such a device has been successfully used for several cavity tests [1]. In this paper we present a prototype of a Digital Cavity Resonance Monitor designed and built in the last year. The hardware of this instrument consists of an RF downconverter, digital quadrature demodulator and digital processor motherboard (Altera FPGA). The motherboard processes received data and computes frequency changes with a resolution of 0.2 Hz, with a 3 kHz output bandwidth.

  1. Optimized mirror supports, active primary mirrors and adaptive secondaries for the Optical Very Large Array (OVLA)

    NASA Astrophysics Data System (ADS)

    Arnold, Luc

    1994-06-01

    This article first deals with general aspects of optimizing mirror supports. A wide variety of support topologies have been optimized by Nelson et al for unobscured entrance pupils. Optical forces and locations of point supports have been calculated here for annular pupils. Efficient topologies introducing a small amount of defocusing are also proposed for unobscured and annular pupils. Support efficiencies are given for each topology. Wavefront errors are estimated in the case of a defective cell, in order to specify tolerances on forces and geometries. The OVLA active optics is then discussed. The very thin, meniscus-shaped primary will be actively supported by 29 actuators and 3 fixed points. Actuator locations and forces have been calculated to minimize the mirror deflection under its own weight but also to allow a good control of astigmatism. We finally present a study of a concave adaptive secondary for the OVLA telescopes. As an initial result, we propose a defocus adaptive corrector with a variable thickness distribution. Conditions of use are defined, and performances are evaluated.

  2. Small foamed polystyrene shield protects low-frequency microphones from wind noise

    NASA Technical Reports Server (NTRS)

    Tedrick, R. N.

    1964-01-01

    A foamed polystyrene noise shield for microphones has been designed in teardrop shape to minimize air turbulence. The shield slips on and off the microphone head easily and is very effective in low-frequency sound intensity measurements.

  3. 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.

  4. Cochlear Implant Microphone Location Affects Speech Recognition in Diffuse Noise

    PubMed Central

    Kolberg, Elizabeth R.; Sheffield, Sterling W.; Davis, Timothy J.; Sunderhaus, Linsey W.; Gifford, René H.

    2015-01-01

    Background Despite improvements in cochlear implants (CIs), CI recipients continue to experience significant communicative difficulty in background noise. Many potential solutions have been proposed to help increase signal-to-noise ratio in noisy environments, including signal processing and external accessories. To date, however, the effect of microphone location on speech recognition in noise has focused primarily on hearing aid users. Purpose The purpose of this study was to (1) measure physical output for the T-Mic as compared with the integrated behind-the-ear(BTE) processor mic for various source azimuths, and (2) to investigate the effect of CI processor mic location for speech recognition in semi-diffuse noise with speech originating from various source azimuths as encountered in everyday communicative environments. Research Design A repeated-measures, within-participant design was used to compare performance across listening conditions. Study Sample A total of 11 adults with Advanced Bionics CIs were recruited for this study. Data Collection and Analysis Physical acoustic output was measured on a Knowles Experimental Mannequin for Acoustic Research (KEMAR) for the T-Mic and BTE mic, with broadband noise presented at 0 and 90° (directed toward the implant processor). In addition to physical acoustic measurements, we also assessed recognition of sentences constructed by researchers at Texas Instruments, the Massachusetts Institute of Technology, and the Stanford Research Institute (TIMIT sentences) at 60 dBA for speech source azimuths of 0, 90, and 270°. Sentences were presented in a semi-diffuse restaurant noise originating from the R-SPACE 8-loudspeaker array. Signal-to-noise ratio was determined individually to achieve approximately 50% correct in the unilateral implanted listening condition with speech at 0°. Performance was compared across the T-Mic, 50/50, and the integrated BTE processor mic. Results The integrated BTE mic provided approximately 5

  5. 78 FR 38734 - Certain Silicon Microphone Packages and Products Containing Same; Notice of Receipt of Complaint...

    Federal Register 2010, 2011, 2012, 2013, 2014

    2013-06-27

    ... COMMISSION Certain Silicon Microphone Packages and Products Containing Same; Notice of Receipt of Complaint... complaint entitled Certain Silicon Microphone Packages and Products Containing Same, DN 2962; the Commission... importation of certain silicon microphone packages and products containing same. The complaint names...

  6. 78 FR 45272 - Certain Silicon Microphone Packages and Products Containing Same Institution of Investigation...

    Federal Register 2010, 2011, 2012, 2013, 2014

    2013-07-26

    ... COMMISSION Certain Silicon Microphone Packages and Products Containing Same Institution of Investigation... importation, and the sale within the United States after importation of certain silicon microphone packages... importation, or the sale within the United States after importation of certain silicon microphone packages...

  7. 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)

  8. 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

  9. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2013 CFR

    2013-10-01

    ... 49 Transportation 5 2013-10-01 2013-10-01 false Microphone distance correction factors. 1 325.73 Section 325.73 Transportation Other Regulations Relating to Transportation (Continued) FEDERAL MOTOR CARRIER SAFETY ADMINISTRATION, DEPARTMENT OF TRANSPORTATION GENERAL REGULATIONS COMPLIANCE WITH...

  10. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2011 CFR

    2011-10-01

    ... 49 Transportation 5 2011-10-01 2011-10-01 false Microphone distance correction factors. 1 325.73 Section 325.73 Transportation Other Regulations Relating to Transportation (Continued) FEDERAL MOTOR CARRIER SAFETY ADMINISTRATION, DEPARTMENT OF TRANSPORTATION GENERAL REGULATIONS COMPLIANCE WITH...

  11. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2012 CFR

    2012-10-01

    ... 49 Transportation 5 2012-10-01 2012-10-01 false Microphone distance correction factors. 1 325.73 Section 325.73 Transportation Other Regulations Relating to Transportation (Continued) FEDERAL MOTOR CARRIER SAFETY ADMINISTRATION, DEPARTMENT OF TRANSPORTATION GENERAL REGULATIONS COMPLIANCE WITH...

  12. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2014 CFR

    2014-10-01

    ... 49 Transportation 5 2014-10-01 2014-10-01 false Microphone distance correction factors. 1 325.73 Section 325.73 Transportation Other Regulations Relating to Transportation (Continued) FEDERAL MOTOR CARRIER SAFETY ADMINISTRATION, DEPARTMENT OF TRANSPORTATION GENERAL REGULATIONS COMPLIANCE WITH...

  13. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2010 CFR

    2010-10-01

    ... 49 Transportation 5 2010-10-01 2010-10-01 false Microphone distance correction factors. 1 325.73 Section 325.73 Transportation Other Regulations Relating to Transportation (Continued) FEDERAL MOTOR CARRIER SAFETY ADMINISTRATION, DEPARTMENT OF TRANSPORTATION GENERAL REGULATIONS COMPLIANCE WITH...

  14. Single and multiple microphone noise reduction strategies in cochlear implants.

    PubMed

    Kokkinakis, Kostas; Azimi, Behnam; Hu, Yi; Friedland, David R

    2012-06-01

    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.

  15. 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…

  16. 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…

  17. Feasible pickup from intact ossicular chain with floating piezoelectric microphone

    PubMed Central

    2012-01-01

    Objectives 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. Methods 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. Results 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. Conclusions 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. PMID:22353161

  18. Analysis and active compensation of microphonics in continuous wave narrow-bandwidth superconducting cavities

    NASA Astrophysics Data System (ADS)

    Neumann, A.; Anders, W.; Kugeler, O.; Knobloch, J.

    2010-08-01

    Many proposals for next generation light sources based on single pass free electron lasers or energy recovery linac facilities require a continuous wave (cw) driven superconducting linac. The effective beam loading in such machines is very small and in principle the cavities can be operated at a bandwidth of a few Hz and with less than a few kW of rf power. However, a power reserve is required to ensure field stability. A major error source is the mechanical microphonics detuning of the niobium cavities. To understand the influence of cavity detuning on longitudinal beam stability, a measurement program has been started at the horizontal cavity test facility HoBiCaT at HZB to study TESLA-type cavities. The microphonics detuning spectral content, peak detuning values, and the driving terms for these mechanical oscillations have been analyzed. In combination with the characterization of cw-adapted fast tuning systems based on the piezoelectric effect this information has been used to design a detuning compensation algorithm. It has been shown that a compensation factor between 2-7 is achievable, reducing the typical detuning of 2-3 Hz rms to below 0.5 Hz rms. These results were included in rf-control simulations of the cavities, and it was demonstrated that a phase stability below 0.02° can be achieved.

  19. Design and evaluation of a higher-order spherical microphone/ambisonic sound reproduction system for the acoustical assessment of concert halls

    NASA Astrophysics Data System (ADS)

    Clapp, Samuel W.

    Previous studies of the perception of concert hall acoustics have generally employed two methods for soliciting listeners' judgments. One method is to have listeners rate the sound in a hall while physically present in that hall. The other method is to make recordings of different halls and seat positions, and then recreate the environment for listeners in a laboratory setting via loudspeakers or headphones. In situ evaluations offer a completely faithful rendering of all aspects of the concert hall experience. However, many variables cannot be controlled and the short duration of auditory memory precludes an objective comparison of different spaces. Simulation studies allow for more control over various aspects of the evaluations, as well as A/B comparisons of different halls and seat positions. The drawback is that all simulation methods suffer from limitations in the accuracy of reproduction. If the accuracy of the simulation system is improved, then the advantages of the simulation method can be retained, while mitigating its disadvantages. Spherical microphone array technology has received growing interest in the acoustics community in recent years for many applications including beamforming, source localization, and other forms of three-dimensional sound field analysis. These arrays can decompose a measured sound field into its spherical harmonic components, the spherical harmonics being a set of spatial basis functions on the sphere that are derived from solving the wave equation in spherical coordinates. Ambisonics is a system for two- and three-dimensional spatialized sound that is based on recreating a sound field from its spherical harmonic components. Because of these shared mathematical underpinnings, ambisonics provides a natural way to present fully spatialized renderings of recordings made with a spherical microphone array. Many of the previously studied applications of spherical microphone arrays have used a narrow frequency range where the array

  20. High-performance liquid chromatography with diode-array detection cotinine method adapted for the assessment of tobacco smoke exposure.

    PubMed

    Bartolomé, Mónica; Gallego-Picó, Alejandrina; Huetos, Olga; Castaño, Argelia

    2014-06-01

    Smoking is considered to be one of the main risk factors for cancer and other diseases and is the second leading cause of death worldwide. As the anti-tobacco legislation implemented in Europe has reduced secondhand smoke exposure levels, analytical methods must be adapted to these new levels. Recent research has demonstrated that cotinine is the best overall discriminator when biomarkers are used to determine whether a person has ongoing exposure to tobacco smoke. This work proposes a sensitive, simple and low-cost method based on solid-phase extraction and liquid chromatography with diode array detection for the assessment of tobacco smoke exposure by cotinine determination in urine. The analytical procedure is simple and fast (20 min) when compared to other similar methods existing in the literature, and it is cheaper than the mass spectrometry techniques usually used to quantify levels in nonsmokers. We obtained a quantification limit of 12.30 μg/L and a recovery of over 90%. The linearity ranges used were 12-250 and 250-4000 μg/L. The method was successfully used to determine cotinine in urine samples collected from different volunteers and is clearly an alternative routine method that allows active and passive smokers to be distinguished.

  1. 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.

  2. Investigation of continuously traversing microphone system for mode measurement

    NASA Technical Reports Server (NTRS)

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

    1982-01-01

    The continuously Traversing Microphone System consists of a data acquisition and processing method for obtaining the modal coefficients of the discrete, coherent acoustic field in a fan inlet duct. The system would be used in fan rigs or full scale engine installations where present measurement methods, because of the excessive number of microphones and long test times required, are not feasible. The purpose of the investigation reported here was to develop a method for defining modal structure by means of a continuously traversing microphone system and to perform an evaluation of the method, based upon analytical studies and computer simulated tests. A variety of system parameters were examined, and the effects of deviations from ideal were explored. Effects of traverse speed, digitizing rate, run time, roundoff error, calibration errors, and random noise background level were determined. For constant fan operating speed, the sensitivity of the method to normal errors and deviations was determined to be acceptable. Good recovery of mode coefficients was attainable. Fluctuating fan speed conditions received special attention, and it was concluded that by employing suitable time delay procedures, satisfactory information on mode coefficients can be obtained under realistic conditions. A plan for further development involving fan rig tests was prepared.

  3. 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.

  4. Feedback characteristics between implantable microphone and transducer in middle ear cavity.

    PubMed

    Arman Woo, S H; Woo, Seong Tak; Song, Byung Seop; Cho, Jin-Ho

    2013-10-01

    With the advent of implantable hearing aids, implementation and acoustic sensing strategy of the implantable microphone becomes an important issue; among the many types of implantable microphone, placing the microphone in middle ear cavity (MEC) has advantages including simple operation and insensitive to skin touching or chewing motion. In this paper, an implantable microphone was implemented and researched feedback characteristic when both the implantable microphone and the transducer were placed in the MEC. Analytical and finite element analysis were conducted to design the microphone to have a natural frequency of 7 kHz and showed good characteristics of SNR and sensitivity. For the feedback test, simple analytical and finite element analysis were calculated and compared with in vitro experiments (n = 4). From the experiments, the open-loop gain and feedback factor were measured and the minimum gain margin measured as 14.3 dB.

  5. Adapt

    NASA Astrophysics Data System (ADS)

    Bargatze, L. F.

    2015-12-01

    Active Data Archive Product Tracking (ADAPT) is a collection of software routines that permits one to generate XML metadata files to describe and register data products in support of the NASA Heliophysics Virtual Observatory VxO effort. ADAPT is also a philosophy. The ADAPT concept is to use any and all available metadata associated with scientific data to produce XML metadata descriptions in a consistent, uniform, and organized fashion to provide blanket access to the full complement of data stored on a targeted data server. In this poster, we present an application of ADAPT to describe all of the data products that are stored by using the Common Data File (CDF) format served out by the CDAWEB and SPDF data servers hosted at the NASA Goddard Space Flight Center. These data servers are the primary repositories for NASA Heliophysics data. For this purpose, the ADAPT routines have been used to generate data resource descriptions by using an XML schema named Space Physics Archive, Search, and Extract (SPASE). SPASE is the designated standard for documenting Heliophysics data products, as adopted by the Heliophysics Data and Model Consortium. The set of SPASE XML resource descriptions produced by ADAPT includes high-level descriptions of numerical data products, display data products, or catalogs and also includes low-level "Granule" descriptions. A SPASE Granule is effectively a universal access metadata resource; a Granule associates an individual data file (e.g. a CDF file) with a "parent" high-level data resource description, assigns a resource identifier to the file, and lists the corresponding assess URL(s). The CDAWEB and SPDF file systems were queried to provide the input required by the ADAPT software to create an initial set of SPASE metadata resource descriptions. Then, the CDAWEB and SPDF data repositories were queried subsequently on a nightly basis and the CDF file lists were checked for any changes such as the occurrence of new, modified, or deleted

  6. Photoacoustic Spectroscopy Using a MEMS Microphone with Inter-IC Sound Digital Output

    NASA Astrophysics Data System (ADS)

    Bruhns, H.; Marianovich, A.; Wolff, M.

    2014-12-01

    In photoacoustic spectroscopy that is adopted for gas sensing, microphones are usually used to detect the pressure variation inside the sample cell. The application of a new commercial inter-integrated circuit sound digital micro-electro-mechanical systems (MEMS) microphone that is enhanced with a filter and an analog-to-digital converter in a single package is presented. The utilization of the described MEMS microphone together with an embedded microcontroller significantly reduces the required space and costs for the components needed to realize the signal detection path of the spectrometer. The measurement results of this signal detection path are compared with those of a conventional photoacoustic spectrometer that is equipped with a capacitive microphone, a microphone preamplifier, and a lock-in amplifier for signal processing. At the first stage of our study, the recorded 24 bit data streams of both microphones are evaluated. At a second stage, the digital output signals of the MEMS microphone are processed with the Goertzel algorithm. The results are compared with the digital output of a lock-in amplifier that is connected to the microphone preamplifier's output of the condenser microphone.

  7. 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.

  8. Perception and automatic detection of wind-induced microphone noise.

    PubMed

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

    2014-09-01

    Wind can induce noise on microphones, causing problems for users of hearing aids and for those making recordings outdoors. Perceptual tests in the laboratory and via the Internet were carried out to understand what features of wind noise are important to the perceived audio quality of speech recordings. The average A-weighted sound pressure level of the wind noise was found to dominate the perceived degradation of quality, while gustiness was mostly unimportant. Large degradations in quality were observed when the signal to noise ratio was lower than about 15 dB. A model to allow an estimation of wind noise level was developed using an ensemble of decision trees. The model was designed to work with a single microphone in the presence of a variety of foreground sounds. The model outputted four classes of wind noise: none, low, medium, and high. Wind free examples were accurately identified in 79% of cases. For the three classes with noise present, on average 93% of samples were correctly assigned. A second ensemble of decision trees was used to estimate the signal to noise ratio and thereby infer the perceived degradation caused by wind noise. PMID:25190392

  9. Perception and automatic detection of wind-induced microphone noise.

    PubMed

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

    2014-09-01

    Wind can induce noise on microphones, causing problems for users of hearing aids and for those making recordings outdoors. Perceptual tests in the laboratory and via the Internet were carried out to understand what features of wind noise are important to the perceived audio quality of speech recordings. The average A-weighted sound pressure level of the wind noise was found to dominate the perceived degradation of quality, while gustiness was mostly unimportant. Large degradations in quality were observed when the signal to noise ratio was lower than about 15 dB. A model to allow an estimation of wind noise level was developed using an ensemble of decision trees. The model was designed to work with a single microphone in the presence of a variety of foreground sounds. The model outputted four classes of wind noise: none, low, medium, and high. Wind free examples were accurately identified in 79% of cases. For the three classes with noise present, on average 93% of samples were correctly assigned. A second ensemble of decision trees was used to estimate the signal to noise ratio and thereby infer the perceived degradation caused by wind noise.

  10. 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.

  11. MagicPlate-512: A 2D silicon detector array for quality assurance of stereotactic motion adaptive radiotherapy

    SciTech Connect

    Petasecca, M. Newall, M. K.; Aldosari, A. H.; Fuduli, I.; Espinoza, A. A.; Porumb, C. S.; Guatelli, S.; Metcalfe, P.; Lerch, M. L. F.; Rosenfeld, A. B.; Booth, J. T.; Colvill, E.; Duncan, M.; Cammarano, D.; Carolan, M.; Oborn, B.; Perevertaylo, V.; Keall, P. J.

    2015-06-15

    Purpose: Spatial and temporal resolutions are two of the most important features for quality assurance instrumentation of motion adaptive radiotherapy modalities. The goal of this work is to characterize the performance of the 2D high spatial resolution monolithic silicon diode array named “MagicPlate-512” for quality assurance of stereotactic body radiation therapy (SBRT) and stereotactic radiosurgery (SRS) combined with a dynamic multileaf collimator (MLC) tracking technique for motion compensation. Methods: MagicPlate-512 is used in combination with the movable platform HexaMotion and a research version of radiofrequency tracking system Calypso driving MLC tracking software. The authors reconstruct 2D dose distributions of small field square beams in three modalities: in static conditions, mimicking the temporal movement pattern of a lung tumor and tracking the moving target while the MLC compensates almost instantaneously for the tumor displacement. Use of Calypso in combination with MagicPlate-512 requires a proper radiofrequency interference shielding. Impact of the shielding on dosimetry has been simulated by GEANT4 and verified experimentally. Temporal and spatial resolutions of the dosimetry system allow also for accurate verification of segments of complex stereotactic radiotherapy plans with identification of the instant and location where a certain dose is delivered. This feature allows for retrospective temporal reconstruction of the delivery process and easy identification of error in the tracking or the multileaf collimator driving systems. A sliding MLC wedge combined with the lung motion pattern has been measured. The ability of the MagicPlate-512 (MP512) in 2D dose mapping in all three modes of operation was benchmarked by EBT3 film. Results: Full width at half maximum and penumbra of the moving and stationary dose profiles measured by EBT3 film and MagicPlate-512 confirm that motion has a significant impact on the dose distribution. Motion

  12. An Eye-adapted Beamforming for Axial B-scans Free from Crystalline Lens Aberration: In vitro and ex vivo Results with a 20 MHz Linear Array

    NASA Astrophysics Data System (ADS)

    Matéo, Tony; Mofid, Yassine; Grégoire, Jean-Marc; Ossant, Frédéric

    In ophtalmic ultrasonography, axial B-scans are seriously deteriorated owing to the presence of the crystalline lens. This strongly aberrating medium affects both spatial and contrast resolution and causes important distortions. To deal with this issue, an adapted beamforming (BF) has been developed and experimented with a 20 MHz linear array working with a custom US research scanner. The adapted BF computes focusing delays that compensate for crystalline phase aberration, including refraction effects. This BF was tested in vitro by imaging a wire phantom through an eye phantom consisting of a synthetic gelatin lens, shaped according to the unaccommodated state of an adult human crystalline lens, anatomically set up in an appropriate liquid (turpentine) to approach the in vivo velocity ratio. Both image quality and fidelity from the adapted BF were assessed and compared with conventional delay-and-sum BF over the aberrating medium. Results showed 2-fold improvement of the lateral resolution, greater sensitivity and 90% reduction of the spatial error (from 758 μm to 76 μm) with adapted BF compared to conventional BF. Finally, promising first ex vivo axial B-scans of a human eye are presented.

  13. Optimized vector sound intensity measurements with a tetrahedral arrangement of microphones in a spherical shell.

    PubMed

    Sondergaard, Thomas; Wille, Morten

    2015-11-01

    Recent times have seen the introduction of small spherical arrays whose usefulness as sound intensity probes is the focus of this paper. The presented probe consists of a spherical shell, 30 mm in diameter, housing four 14 in. microphones arranged in a regular tetrahedral configuration. Classical formulae may be used to estimate the sound intensity vector, as may methods based on spherical harmonics decomposition. Results are shown to be comparable to those obtained from classical sound intensity probes. The existence of an analytical model for a plane wave's diffraction about a sphere provides a means for adopting common optimization techniques for potentially improving the intensity vector estimate, however. This paper examines the validity of non-linear least squares optimization in conjunction with the proposed spherical sound intensity probe when placed in the following sound fields: (1) a simple plane wave; (2) a plane wave corrupted by noise; and (3) multiple incident plane waves. Under certain conditions, the probe is shown to greatly extend the operational frequency range of classical sound intensity probes. The optimization algorithm is found to lack robustness against deviations from plane wave conditions, however.

  14. All-optical low noise fiber Bragg grating microphone.

    PubMed

    Bandutunga, Chathura P; Fleddermann, Roland; Gray, Malcolm B; Close, John D; Chow, Jong H

    2016-07-20

    We present an all-fiber design for a microphone using a fiber Bragg grating Fabry-Perot resonator attached to a diaphragm transducer. We analytically model and verify the fiber-diaphragm mechanical interaction, using the Hänsch-Couillaud readout technique to provide necessary sensitivity. We achieved a noise-equivalent strain sensitivity of 7.1×10-12  ϵ/Hz, which corresponds to a sound pressure of 74  μPa/Hz at 1 kHz limited by laser frequency noise and yielding a signal-to-noise ratio of 47±2  dB with a 1 Pa drive at 1 kHz, in close agreement with modeled results. PMID:27463906

  15. Adaptive lenticular microlens array based on voltage-induced waves at the surface of polyvinyl chloride/dibutyl phthalate gels.

    PubMed

    Xu, Miao; Jin, Boya; He, Rui; Ren, Hongwen

    2016-04-18

    We report a new approach to preparing a lenticular microlens array (LMA) using polyvinyl chloride (PVC)/dibutyl phthalate (DBP) gels. The PVD/DBP gels coated on a glass substrate form a membrane. With the aid of electrostatic repulsive force, the surface of the membrane can be reconfigured with sinusoidal waves by a DC voltage. The membrane with wavy surface functions as a LMA. By switching over the anode and cathode, the convex shape of each lenticular microlens in the array can be converted to the concave shape. Therefore, the LMA can present a large dynamic range. The response time is relatively fast and the driving voltage is low. With the advantages of compact structure, optical isotropy, and good mechanical stability, our LMA has potential applications in imaging, information processing, biometrics, and displays. PMID:27137253

  16. 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.

  17. 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.

  18. Modeling distributed electrostatic effects in silicon microphones and their impact on the performance

    NASA Astrophysics Data System (ADS)

    Kuenzig, Thomas; Schrag, Gabriele; Dehé, Alfons; Wachutka, Gerhard

    2015-05-01

    We present a system-level model for fast and efficient investigations of distributed electrostatic effects in state-of-the-art silicon microphones. Combining lumped and distributed submodels it accounts for electrostatic forces and capacitive read-out, including non-linearities, fringing fields and parasitics. The derived model is calibrated using electrostatic finite element (FE) simulations and validated by measurements. The non-linearities caused by electrostatic effects have a decisive impact on the sensitivity of the microphone and the distortion of the transduced acoustical signal. Hence, the proposed model provides important insights into the operation of the device, which can be employed to optimize the microphone characteristics.

  19. Investigation of microphones as near-ground sensors for seismic detection of buried landmines.

    PubMed

    Larson, Gregg D; Martin, James S; Scott, Waymond R

    2007-07-01

    Commercially available microphones were investigated as near-ground sensors to measure the acoustic pressure and the vertical pressure gradient of evanescent air-acoustic waves associated with audio-frequency seismic waves. Measurements in close proximity to the surface and the use of waveguides were found to improve the microphone signal's quality, the comparison of its seismic sensitivity to its sensitivity to propagating sound (ambient acoustic noise and nonseismic reverberation). Landmine images formed using microphone data collected in a laboratory experimental model clearly locate buried inert landmines but exhibit more clutter than images of the same objects formed with seismic displacement data collected using other techniques.

  20. A Two-dimensional Position Estimate of Two Sound Sources Using Two Microphones with Reflectors

    NASA Astrophysics Data System (ADS)

    Nakashima, Hiromichi; Kawamoto, Mitsuru; Ito, Masanori; Mukai, Toshiharu

    Human beings and living things have the capability of identifying the directions of two or more sounds by a certain amount of correctness with only two ears. However it is difficult to give this capability to robots. Almost all the robots which have been proposed until now have three or more microphones in order to localize sound sources. In this paper, we propose a technique of estimating two kinds of directions, that is, vertical and horizontal directions, using a robot head consisted of two microphones, where the microphones of the robot head have reflectors working like the pinna.

  1. Simulation Study of Electronic Damping of Microphonic Vibrations in Superconducting Cavities

    SciTech Connect

    Alicia Hofler; Jean Delayen

    2005-05-01

    Electronic damping of microphonic vibrations in superconducting rf cavities involves an active modulation of the cavity field amplitude in order to induce ponderomotive forces that counteract the effect of ambient vibrations on the cavity frequency. In lightly beam loaded cavities, a reduction of the microphonics-induced frequency excursions leads directly to a reduction of the rf power required for phase and amplitude stabilization. Jefferson Lab is investigating such an electronic damping scheme that could be applied to the JLab 12 GeV upgrade, the RIA driver, and possibly to energy-recovering superconducting linacs. This paper discusses a model and presents simulation results for electronic damping of microphonic vibrations.

  2. 78 FR 21977 - Certain Silicon Microphone Packages and Products Containing the Same; Commission Determination...

    Federal Register 2010, 2011, 2012, 2013, 2014

    2013-04-12

    ...,049, 77 FR 2087 (Jan. 13, 2012). The respondents are Analog Devices Inc. of Norwood, Massachusetts... From the Federal Register Online via the Government Publishing Office INTERNATIONAL TRADE COMMISSION Certain Silicon Microphone Packages and Products Containing the Same; Commission Determination...

  3. 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.

  4. A New Trans-Tympanic Microphone Approach for Fully Implantable Hearing Devices

    PubMed Central

    Woo, Seong Tak; Shin, Dong Ho; Lim, Hyung-Gyu; Seong, Ki-Woong; Gottlieb, Peter; Puria, Sunil; Lee, Kyu-Yup; Cho, Jin-Ho

    2015-01-01

    Fully implantable hearing devices (FIHDs) have been developed as a new technology to overcome the disadvantages of conventional acoustic hearing aids. The implantable microphones currently used in FIHDs, however, have difficulty achieving high sensitivity to environmental sounds, low sensitivity to body noise, and ease of implantation. In general, implantable microphones may be placed under the skin in the temporal bone region of the skull. In this situation, body noise picked up during mastication and touching can be significant, and the layer of skin and hair can both attenuate and distort sounds. The new approach presently proposed is a microphone implanted at the tympanic membrane. This method increases the microphone’s sensitivity by utilizing the pinna’s directionally dependent sound collection capabilities and the natural resonances of the ear canal. The sensitivity and insertion loss of this microphone were measured in human cadaveric specimens in the 0.1 to 16 kHz frequency range. In addition, the maximum stable gain due to feedback between the trans-tympanic microphone and a round-window-drive transducer, was measured. The results confirmed in situ high-performance capabilities of the proposed trans-tympanic microphone. PMID:26371007

  5. Open Microphone Speech Understanding: Correct Discrimination Of In Domain Speech

    NASA Technical Reports Server (NTRS)

    Hieronymus, James; Aist, Greg; Dowding, John

    2006-01-01

    An ideal spoken dialogue system listens continually and determines which utterances were spoken to it, understands them and responds appropriately while ignoring the rest This paper outlines a simple method for achieving this goal which involves trading a slightly higher false rejection rate of in domain utterances for a higher correct rejection rate of Out of Domain (OOD) utterances. The system recognizes semantic entities specified by a unification grammar which is specialized by Explanation Based Learning (EBL). so that it only uses rules which are seen in the training data. The resulting grammar has probabilities assigned to each construct so that overgeneralizations are not a problem. The resulting system only recognizes utterances which reduce to a valid logical form which has meaning for the system and rejects the rest. A class N-gram grammar has been trained on the same training data. This system gives good recognition performance and offers good Out of Domain discrimination when combined with the semantic analysis. The resulting systems were tested on a Space Station Robot Dialogue Speech Database and a subset of the OGI conversational speech database. Both systems run in real time on a PC laptop and the present performance allows continuous listening with an acceptably low false acceptance rate. This type of open microphone system has been used in the Clarissa procedure reading and navigation spoken dialogue system which is being tested on the International Space Station.

  6. 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.

  7. Laser diode edge sensors for adaptive optics segmented arrays: Part 1--external cavity coupling and detector current

    NASA Astrophysics Data System (ADS)

    Remo, John L.

    1994-05-01

    An analytical study of laser diode (LD) operation coupled to external cavity scattering elements, which function as variably coupling reflectors (VCRs), is carried out with the purpose of determining the interrelationship between cavity coupling and intracavity optical intensity which determine the current generated at the rear facet PIN detector. If the external cavity coupling is position sensitive it can allow the relative position between the LD and the external cavity to be determined from the PIN or other detector mounted with the LD. If the LD and external cavity element are placed on opposite edges of two adjacent adaptive optics segments they can provide the basis for a self aligning position sensor; the amount of current detected at the PIN or other detector will depend on the relative displacement between the LD and external coupling element. Schematics of the edge sensors, the basic electronic configuration, and the optics of the external cavity are given. The ratio of the internal cavity intensity, Ic, to the saturation intensity, Is, is plotted as a function of the external cavity coupling. When this ratio approaches one, large-signal output is not a linear function of large-signal output. For operation well below saturation, the PIN detector current is directly related to Ic and may serve as a reliable detector.

  8. 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.

  9. Active structural acoustic control of a smart cylindrical shell using a virtual microphone

    NASA Astrophysics Data System (ADS)

    Loghmani, Ali; Danesh, Mohammad; Kwak, Moon K.; Keshmiri, Mehdi

    2016-04-01

    This paper investigates the active structural acoustic control of sound radiated from a smart cylindrical shell. The cylinder is equipped with piezoelectric sensors and actuators to estimate and control the sound pressure that radiates from the smart shell. This estimated pressure is referred to as a virtual microphone, and it can be used in control systems instead of actual microphones to attenuate noise due to structural vibrations. To this end, the dynamic model for the smart cylinder is derived using the extended Hamilton’s principle, the Sanders shell theory and the assumed mode method. The simplified Kirchhoff-Helmholtz integral estimates the far-field sound pressure radiating from the baffled cylindrical shell. A modified higher harmonic controller that can cope with a harmonic disturbance is designed and experimentally evaluated. The experimental tests were carried out on a baffled cylindrical aluminum shell in an anechoic chamber. The frequency response for the theoretical virtual microphone and the experimental actual microphone are in good agreement with each other, and the results show the effectiveness of the designed virtual microphone and controller in attenuating the radiated sound.

  10. Two clover-shaped piezoresistive silicon microphones for photo acoustic gas sensors

    NASA Astrophysics Data System (ADS)

    Grinde, C.; Sanginario, A.; Ohlckers, P. A.; Jensen, G. U.; Mielnik, M. M.

    2010-04-01

    Low cost CO2 gas sensors for demand-controlled ventilation can lower the energy consumption and increase comfort and hence productivity in office buildings and schools. The photo aoustic principle offers very high sensitivity and selectivity when used for gas trace analysis. Current systems are too expensive and large for in-duct mounting. Here, the design, modeling, fabrication and characterization of two micromachined silicon microphones with piezoresistive readout designed for low cost photo acoustic gas sensors are presented. The microphones have been fabricated using a foundry MPW service. One of the microphones has been fabricated using an additional etching step that allows etching through membranes with large variations in thickness. To increase sensitivity and resolution, a design based on a released membrane suspended by four beams was chosen. The microphones have been characterized for frequencies up to 1 kHz and 100 Hz, respectively. Averaged sensitivities are measured to be 30 µV/(V × Pa) and 400 µV/(V × Pa). The presented microphones offer increased sensitivities compared to similar sensors.

  11. Free-field calibration of measurement microphones at frequencies up to 80 kHz

    NASA Astrophysics Data System (ADS)

    Zuckerwar, Allan J.; Herring, Gregory C.

    2002-11-01

    Civil-aviation noise-reduction programs, that make use of scaled-down aircraft models in wind tunnel tests, require knowledge of microphone pressure (i.e., not free-field) sensitivities beyond 20 kHz--since noise wavelengths also scale down with decreasing model size. Furthermore, not all microphone types (e.g., electrets) are easily calibrated with the electrostatic technique, while enclosed cavity calibrations typically have an upper limit for the useful frequency range. Thus, work was initiated to perform a high-frequency pressure calibration of Panasonic electret microphones using a substitution free-field method in a small anechoic chamber. First, a standard variable-frequency pistonphone was used to obtain the pressure calibration up to 16 kHz. Above 16 kHz, to avoid spatially irregular sound fields due to dephasing of loudspeaker diaphragms, a series of resonant ceramic piezoelectric crystals was used at five specific ultrasonic frequencies as the free-field calibration sound source. Then, the free-field sensitivity was converted to a pressure sensitivity with an electrostatic calibration of the reference microphone (an air condenser type), for which the free-field correction is known. Combining the low- and high-frequency data sets, a full frequency calibration of pressure sensitivity for an electret microphone was generated from 63 Hz to 80 kHz.

  12. Sound scattering by rigid oblate spheroids, with implication to pressure gradient microphones

    NASA Technical Reports Server (NTRS)

    Maciulaitis, A.; Seiner, J.; Norum, T. D.

    1976-01-01

    The frequency limit below which sound scattering by a microphone body is sufficiently small to permit accurate pressure gradient measurements was determined. The sound pressure was measured at various points on the surface of a rigid oblate spheroid illuminated by spherical waves generated by a point source at a large distance from the spheroid, insuring an essentially plane sound field. The measurements were made with small pressure microphones flush mounted from the inside of the spheroid model. Numerical solutions were obtained for a variety of spheroid shapes, including that of the experimental model. Very good agreement was achieved between the experimental and theoretical results. It was found that scattering effects are insignificant if the ratio of the major circumference of the spheroid to the wavelength of the incident sound is less than about 0.7, this number being dependent upon the shape of the spheroid. This finding can be utilized in the design of pressure gradient microphones.

  13. 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.

  14. Standoff photoacoustic detection of explosives using quantum cascade laser and an ultrasensitive microphone.

    PubMed

    Chen, Xing; Guo, Dingkai; Choa, Fow-Sen; Wang, Chen-Chia; Trivedi, Sudhir; Snyder, A Peter; Ru, Guoyun; Fan, Jenyu

    2013-04-20

    Standoff detections of explosives using quantum cascade lasers (QCLs) and the photoacoustic (PA) technique were studied. In our experiment, a mid-infrared QCL with emission wavelength near 7.35 μm was used as a laser source. Direct standoff PA detection of trinitrotoluene (TNT) was achieved using an ultrasensitive microphone. The QCL output light was focused on explosive samples in powder form. PA signals were generated and detected directly by an ultrasensitive low-noise microphone with 1 in. diameter. A detection distance up to 8 in. was obtained using the microphone alone. With increasing detection distance, the measured PA signal not only decayed in amplitude but also presented phase delays, which clearly verified the source location. To further increase the detection distance, a parabolic sound reflector was used for effective sound collection. With the help of the sound reflector, standoff PA detection of TNT with distance of 8 ft was demonstrated.

  15. Study of a porous surface microphone sensor in an aerofoil. [air flow

    NASA Technical Reports Server (NTRS)

    Noiseux, D. U.; Noiseux, N. B.; Kadman, Y.

    1975-01-01

    The porous microphone in an airfoil is described as a directional sensor which rejects flow noise. The airfoil allows the sensor to be rotated in the airflow over a wide range of yaw angles, 0 to 90 degrees, avoiding flow separation over the surface of the sensor and its associated additional flow noise. The microphone is discussed in terms of its acoustic properties, vibration sensitivity, effect of Mach number on the directivity function, and flow noise. Additional information on the acoustic calibration of the microphone, the acceleration sensitivity of the airfoil, stationary source and receiver in a moving gas, acoustic tests in airflow, and flow noise tests of the airfoil porous surface sensor is included.

  16. 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.

  17. Constrained adaptation for feedback cancellation in hearing aids.

    PubMed

    Kates, J M

    1999-08-01

    In feedback cancellation in hearing aids, an adaptive filter is used to model the feedback path. The output of the adaptive filter is subtracted from the microphone signal to cancel the acoustic and mechanical feedback picked up by the microphone, thus allowing more gain in the hearing aid. In general, the feedback-cancellation filter adapts on the hearing-aid input signal, and signal cancellation and coloration artifacts can occur for a narrow-band input. In this paper, two procedures for LMS adaptation with a constraint on the magnitude of the adaptive weight vector are derived. The constraints greatly reduce the probability that the adaptive filter will cancel a narrow-band input. Simulation results are used to demonstrate the efficacy of the constrained adaptation. PMID:10462806

  18. Study of porous surface microphones for acoustic measurements in wind tunnels

    NASA Technical Reports Server (NTRS)

    Noiseux, D. U.

    1973-01-01

    Porous surface sensors acting as directional microphones in subsonic airflow were investigated. The first part of the report deals with the design of a porous strip sensor set in an aerofoil. The second part presents the experimental results of frequency response, directivity, and flow noise of a porous pipe sensor and a porous strip sensor. For flow noise, these sensors were compared with the Bruel and Kjaer half-inch condenser microphone with a nose cone. The flow noise was examined under two conditions of flow: in a very quiet flow where the turbulence was approximately 0.3% and in a spoiled flow where the turbulence was approximately 5%.

  19. Dynamic microphones M-87/AIC and M-101/AIC and earphone H-143/AIC. [for space shuttle

    NASA Technical Reports Server (NTRS)

    Reiff, F. H.

    1975-01-01

    The electrical characteristics of the M-87/AIC and M-101/AIC dynamic microphone and H-143 earphones were tested for the purpose of establishing the relative performance levels of units supplied by four vendors. The microphones and earphones were tested for frequency response, sensitivity, linearity, impedance and noise cancellation. Test results are presented and discussed.

  20. 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...

  1. A Micro-Machined Microphone Based on a Combination of Electret and Field-Effect Transistor.

    PubMed

    Shin, Kumjae; Jeon, Junsik; West, James Edward; Moon, Wonkyu

    2015-08-18

    Capacitive-type transduction is now widely used in MEMS microphones. However, its sensitivity decreases with reducing size, due to decreasing air gap capacitance. In the present study, we proposed and developed the Electret Gate of Field Effect Transistor (ElGoFET) transduction based on an electret and FET (field-effect-transistor) as a novel mechanism of MEMS microphone transduction. The ElGoFET transduction has the advantage that the sensitivity is dependent on the ratio of capacitance components in the transduction structure. Hence, ElGoFET transduction has high sensitivity even with a smaller air gap capacitance, due to a miniaturization of the transducer. A FET with a floating-gate electrode embedded on a membrane was designed and fabricated and an electret was fabricated by ion implantation with Ga(+) ions. During the assembly process between the FET and the electret, the operating point of the FET was characterized using the static response of the FET induced by the electric field due to the trapped positive charge at the electret. Additionally, we evaluated the microphone performance of the ElGoFET by measuring the acoustic response in air using a semi-anechoic room. The results confirmed that the proposed transduction mechanism has potential for microphone applications.

  2. A Micro-Machined Microphone Based on a Combination of Electret and Field-Effect Transistor

    PubMed Central

    Shin, Kumjae; Jeon, Junsik; West, James Edward; Moon, Wonkyu

    2015-01-01

    Capacitive-type transduction is now widely used in MEMS microphones. However, its sensitivity decreases with reducing size, due to decreasing air gap capacitance. In the present study, we proposed and developed the Electret Gate of Field Effect Transistor (ElGoFET) transduction based on an electret and FET (field-effect-transistor) as a novel mechanism of MEMS microphone transduction. The ElGoFET transduction has the advantage that the sensitivity is dependent on the ratio of capacitance components in the transduction structure. Hence, ElGoFET transduction has high sensitivity even with a smaller air gap capacitance, due to a miniaturization of the transducer. A FET with a floating-gate electrode embedded on a membrane was designed and fabricated and an electret was fabricated by ion implantation with Ga+ ions. During the assembly process between the FET and the electret, the operating point of the FET was characterized using the static response of the FET induced by the electric field due to the trapped positive charge at the electret. Additionally, we evaluated the microphone performance of the ElGoFET by measuring the acoustic response in air using a semi-anechoic room. The results confirmed that the proposed transduction mechanism has potential for microphone applications. PMID:26295231

  3. 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…

  4. 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…

  5. Noise-induced reduction of inner-ear microphonic response: dependence on body temperature.

    PubMed

    Drescher, D G

    1974-07-19

    The rate of reduction of chinchilla cochlear microphonic response with exposure to steady noise is less at lower body temperatures and greater at higher body temperatures. Before exposure to noise, this auditory response is invariant within the range of temperatures employed. The mechanism of reduction of cochlear response appears to involve processes sensitive to body temperature.

  6. Nonnegative signal factorization with learnt instrument models for sound source separation in close-microphone recordings

    NASA Astrophysics Data System (ADS)

    Carabias-Orti, Julio J.; Cobos, Máximo; Vera-Candeas, Pedro; Rodríguez-Serrano, Francisco J.

    2013-12-01

    Close-microphone techniques are extensively employed in many live music recordings, allowing for interference rejection and reducing the amount of reverberation in the resulting instrument tracks. However, despite the use of directional microphones, the recorded tracks are not completely free from source interference, a problem which is commonly known as microphone leakage. While source separation methods are potentially a solution to this problem, few approaches take into account the huge amount of prior information available in this scenario. In fact, besides the special properties of close-microphone tracks, the knowledge on the number and type of instruments making up the mixture can also be successfully exploited for improved separation performance. In this paper, a nonnegative matrix factorization (NMF) method making use of all the above information is proposed. To this end, a set of instrument models are learnt from a training database and incorporated into a multichannel extension of the NMF algorithm. Several options to initialize the algorithm are suggested, exploring their performance in multiple music tracks and comparing the results to other state-of-the-art approaches.

  7. Hearing Performance Benefits of a Programmable Power Baha® Sound Processor with a Directional Microphone for Patients with a Mixed Hearing Loss

    PubMed Central

    Hedin, Annelen; Halvarsson, Glenn; Good, Tobias; Sadeghi, Andre

    2012-01-01

    Objectives New signal processing technologies have recently become available for Baha® sound processors. These technologies have led to an increase in power and to the implementation of directional microphones. For any new technology, it is important to evaluate the degree of benefit under different listening situations. Methods Twenty wearers of the Baha osseointegrated hearing system participated in the investigation. The control sound processor was the Baha Intenso and the test sound processor was the Cochlear™ Baha® BP110power. Performance was evaluated in terms of free-field audibility with narrow band noise stimuli. Speech recognition of monosyllabic phonetically balanced (PB) words in quiet was performed at three intensity settings (50, 65, and 80 dB sound pressure level [SPL]) with materials presented at 0 degrees azimuth. Speech recognition of sentences in noise using the Hearing in Noise Test (HINT) in an adaptive framework was performed with speech from 0 degrees and noise held constant at 65 dB SPL from 180 degrees. Testing was performed in both the omni and directional microphone settings. Loudness growth was assessed in randomly presented 10 dB steps between 30 and 90 dB SPL to narrow band noise stimuli at 500 Hz and 3,000 Hz. Results The test sound processor had significantly improved high frequency audibility (3,000-8,000 Hz). Speech recognition of PB words in quiet at three different intensity levels (50, 65, and 80 dB SPL) indicated a significant difference in terms of level (P<0.0001) but not for sound processor type (P>0.05). Speech recognition of sentences in noise demonstrated a 2.5 dB signal-to-noise ratio (SNR) improvement in performance for the test sound processor. The directional microphone provided an additional 2.3 dB SNR improvement in speech recognition (P<0.0001). Loudness growth functions demonstrated similar performance, indicating that both sound processors had sufficient headroom and amplification for the required hearing

  8. Coherent optical adaptive techniques.

    PubMed

    Bridges, W B; Brunner, P T; Lazzara, S P; Nussmeier, T A; O'Meara, T R; Sanguinet, J A; Brown, W P

    1974-02-01

    The theory of multidither adaptive optical radar phased arrays is briefly reviewed as an introduction to the experimental results obtained with seven-element linear and three-element triangular array systems operating at 0.6328 microm. Atmospheric turbulence compensation and adaptive tracking capabilities are demonstrated.

  9. 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.

  10. Localization of multiple acoustic sources with small arrays using a coherence test.

    PubMed

    Mohan, Satish; Lockwood, Michael E; Kramer, Michael L; Jones, Douglas L

    2008-04-01

    Direction finding of more sources than sensors is appealing in situations with small sensor arrays. Potential applications include surveillance, teleconferencing, and auditory scene analysis for hearing aids. A new technique for time-frequency-sparse sources, such as speech and vehicle sounds, uses a coherence test to identify low-rank time-frequency bins. These low-rank bins are processed in one of two ways: (1) narrowband spatial spectrum estimation at each bin followed by summation of directional spectra across time and frequency or (2) clustering low-rank covariance matrices, averaging covariance matrices within clusters, and narrowband spatial spectrum estimation of each cluster. Experimental results with omnidirectional microphones and colocated directional microphones demonstrate the algorithm's ability to localize 3-5 simultaneous speech sources over 4 s with 2-3 microphones to less than 1 degree of error, and the ability to localize simultaneously two moving military vehicles and small arms gunfire. PMID:18397021

  11. 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.

  12. 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.

  13. 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. PMID:27475217

  14. Planar microphone based on piezoelectric electrospun poly(γ-benzyl-α,L-glutamate) nanofibers.

    PubMed

    Ren, Kailiang; West, James E; Yu, S Michael

    2014-06-01

    Velocity and pressure microphones composed of piezoelectric poly(γ-benzyl-α,L-glutamate) (PBLG) nanofibers were produced by adhering a single layer of PBLG film to a Mylar diaphragm. The device exhibited a sensitivity of -60 dBV/Pa in air, and both pressure and velocity response showed a broad frequency response that was primarily controlled by the stiffness of the supporting diaphragm. The pressure microphone response was ±3 dB between 200 Hz and 4 kHz when measured in a semi-anechoic chamber. Thermal stability, easy fabrication, and simple design make this single element transducer ideal for various applications including those for underwater and high temperature use.

  15. Fiber Bragg grating microphone system for condition-based maintenance of industrial facilities

    NASA Astrophysics Data System (ADS)

    Tosi, D.; Olivero, M.; Perrone, G.; Vallan, A.

    2011-05-01

    This paper presents a multipoint fiber Bragg grating (FBG) sensing system operating as a precision microphone. This instrument aims to become the best performing technology for condition-based maintenance (CBM) of critical elements, like ball bearings and cogwheels, embedded in industrial manufacturing machineries. The system architecture is based on the simple matched-laser principle, leading to a low-cost and high-sensitivity system, operating in time and wavelength multiplexing mode. Then, heavy signal processing is applied, providing an outstanding performance improvement of 59 dB in terms of signal-to-noise ratio. A demonstration of condition-based maintenance operation has been performed using standard models of ball bearing sound spectra. Compared to traditional microphones applied to CBM, the signal processing-powered FBG system provides remarkable advantages in terms of sensitivity and rejection of environment noise, providing an improvement of cost-effectiveness of CBM.

  16. Acoustic sensor engineering evaluation test report. [microphones for monitoring inside the space shuttle orbiter

    NASA Technical Reports Server (NTRS)

    Phillips, E. L., Jr.; Bronson, R. D.

    1976-01-01

    Two types of one-inch diameter sound pressure level sensors, which are candidates for monitoring ambient noise in the shuttle orbiter crew compartment during rest periods, were exposed to temperature, passive humidity, and vibration. One unexposed sensor of each type served as a reference unit. Except for the humidity exposures, each of the three capacitive microphones was individually tested in sequence with the essential voltage power supply and preamplifier. One unit exibited anomalous characteristics after the humidity exposure but returned to normal after being dried in an oven at 115 deg for two hours. Except for the humidity exposures, each of the three piezoelectric microphones was individually tested with a laboratory type amplifier. Two apparent failures occurred during these tests. The diaphragm on one was found ruptured after the fourth cycle of the humidity test. A second sensor showed an anomaly after the random vibration tests at which time its sensitivity was consistent at about one-half its former value.

  17. Optimisation of photoacoustic resonant cells with commercial microphones for diode laser gas detection

    NASA Astrophysics Data System (ADS)

    Kapitanov, V. A.; Zeninari, V.; Parvitte, B.; Courtois, D.; Ponomarev, Yu. N.

    2002-09-01

    The theoretical and experimental study of the differential Helmholtz resonant (DHR) cell sensitivity under variation of the total gas pressure is made for various commercial microphones. Near-infrared lasers (room-temperature diode lasers) were used to measure the response of DHR cell versus pressure of the absorbing gas and frequency of the laser radiation modulation. Several molecular absorbers like H 2O, CH 4, mixed with molecular buffer gases were used to investigate the behavior of the photoacoustic (PA) signal characteristics with a DHR cell. The experimental data are compared with the results of computer simulation. The minimal detectable concentrations of gases were determined for the DHR cell for each commercial microphone.

  18. 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.

  19. 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. PMID:19894812

  20. Compliant membranes for the development of MEMS dual-backplate capacitive microphone using the SUMMiT V fabrication process.

    SciTech Connect

    Martin, David

    2005-11-01

    The objective of this project is the investigation of compliant membranes for the development of a MicroElectrical Mechanical Systems (MEMS) microphone using the Sandia Ultraplanar, Multilevel MEMS Technology (SUMMiT V) fabrication process. The microphone is a dual-backplate capacitive microphone utilizing electrostatic force feedback. The microphone consists of a diaphragm and two porous backplates, one on either side of the diaphragm. This forms a capacitor between the diaphragm and each backplate. As the incident pressure deflects the diaphragm, the value of each capacitor will change, thus resulting in an electrical output. Feedback may be used in this device by applying a voltage between the diaphragm and the backplates to balance the incident pressure keeping the diaphragm stationary. The SUMMiT V fabrication process is unique in that it can meet the fabrication requirements of this project. All five layers of polysilicon are used in the fabrication of this device. The SUMMiT V process has been optimized to provide low-stress mechanical layers that are ideal for the construction of the microphone's diaphragm. The use of chemical mechanical polishing in the SUMMiT V process results in extremely flat structural layers and uniform spacing between the layers, both of which are critical to the successful fabrication of the MEMS microphone. The MEMS capacitive microphone was fabricated at Sandia National Laboratories and post-processed, packaged, and tested at the University of Florida. The microphone demonstrates a flat frequency response, a linear response up to the designed limit, and a sensitivity that is close to the designed value. Future work will focus on characterization of additional devices, extending the frequency response measurements, and investigating the use of other types of interface circuitry.

  1. 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.

  2. 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.

  3. Microphonics detuning compensation in 3.9 GHZ superconducting RF cavities

    SciTech Connect

    Ruben Carcagno et al.

    2003-10-20

    Mechanical vibrations can detune superconducting radio frequency (SCRF) cavities unless a tuning mechanism counteracting the vibrations is present. Due to their narrow operating bandwidth and demanding mechanical structure, the 13-cell 3.9GHz SCRF cavities for the Charged Kaons at Main Injector (CKM) experiment at Fermilab are especially susceptible to this microphonic phenomena. We present early results correlating RF frequency detuning with cavity vibration measurements for CKM cavities; initial detuning compensation results with piezoelectric actuators are also presented.

  4. STS-39 MS McMonagle adjusts CCA microphones prior to simulation in JSC's WETF

    NASA Technical Reports Server (NTRS)

    1990-01-01

    STS-39 Mission Specialist (MS) Donald R. McMonagle, wearing extravehicular mobility unit (EMU), adjusts the microphones on his communications carrier assembly (CCA) prior to underwater simulation in JSC's Weightless Environment Training Facility (WETF) Bldg 29. McMonagle will be lowered into the WETF's 25 ft deep pool for an underwater simulation of contingency extravehicular activity (EVA) procedures. He is scheduled as a crewmember aboard Discovery, Orbiter Vehicle (OV) 103 in the spring of 1991.

  5. Selective-Tap Blind Dereverberation for Two-Microphone Enhancement of Reverberant Speech

    PubMed Central

    Kokkinakis, Kostas; Loizou, Philipos C.

    2009-01-01

    In this letter we propose a novel approach for two-microphone enhancement of speech corrupted by reverberation. Our approach steers computational resources to filter coefficients having the largest impact on the error surface and therefore only updates a subset of coefficients in every iteration. Experimental results carried out in a realistically reverberant setup indicate that the performance of the proposed algorithm is comparable to the performance of its full-update counterpart. PMID:19885386

  6. 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

  7. Thermal-stress modeling of an optical microphone at high temperature.

    SciTech Connect

    Barnard, Casey Anderson

    2010-08-01

    To help determine the capability range of a MEMS optical microphone design in harsh conditions computer simulations were carried out. Thermal stress modeling was performed up to temperatures of 1000 C. Particular concern was over stress and strain profiles due to the coefficient of thermal expansion mismatch between the polysilicon device and alumina packaging. Preliminary results with simplified models indicate acceptable levels of deformation within the device.

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

    PubMed

    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

  9. 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.

  10. Implementation of the CMOS MEMS condenser microphone with corrugated metal diaphragm and silicon back-plate.

    PubMed

    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

  11. 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.

  12. Analytical modeling of squeeze air film damping of biomimetic MEMS directional microphone

    NASA Astrophysics Data System (ADS)

    Ishfaque, Asif; Kim, Byungki

    2016-08-01

    Squeeze air film damping is introduced in microelectromechanical systems due to the motion of the fluid between two closely spaced oscillating micro-structures. The literature is abundant with different analytical models to address the squeeze air film damping effects, however, there is a lack of work in modeling the practical sensors like directional microphones. Here, we derive an analytical model of squeeze air film damping of first two fundamental vibration modes, namely, rocking and bending modes, of a directional microphone inspired from the fly Ormia ochracea's ear anatomy. A modified Reynolds equation that includes compressibility and rarefaction effects is used in the analysis. Pressure distribution under the vibrating diaphragm is derived by using Green's function. From mathematical modeling of the fly's inspired mechanical model, we infer that bringing the damping ratios of both modes in the critical damping range enhance the directional sensitivity cues. The microphone parameters are varied in derived damping formulas to bring the damping ratios in the vicinity of critical damping, and to show the usefulness of the analytical model in tuning the damping ratios of both modes. The accuracy of analytical damping results are also verified by finite element method (FEM) using ANSYS. The FEM results are in full compliance with the analytical results.

  13. Model based prediction of the existence of the spontaneous cochlear microphonic

    NASA Astrophysics Data System (ADS)

    Ayat, Mohammad; Teal, Paul D.

    2015-12-01

    In the mammalian cochlea, self-sustaining oscillation of the basilar membrane in the cochlea can cause vibration of the ear drum, and produce spontaneous narrow-band air pressure fluctuations in the ear canal. These spontaneous fluctuations are known as spontaneous otoacoustic emissions. Small perturbations in feedback gain of the cochlear amplifier have been proposed to be the generation source of self-sustaining oscillations of the basilar membrane. We hypothesise that the self-sustaining oscillation resulting from small perturbations in feedback gain produce spontaneous potentials in the cochlea. We demonstrate that according to the results of the model, a measurable spontaneous cochlear microphonic must exist in the human cochlea. The existence of this signal has not yet been reported. However, this spontaneous electrical signal could play an important role in auditory research. Successful or unsuccessful recording of this signal will indicate whether previous hypotheses about the generation source of spontaneous otoacoustic emissions are valid or should be amended. In addition according to the proposed model spontaneous cochlear microphonic is basically an electrical analogue of spontaneous otoacoustic emissions. In certain experiments, spontaneous cochlear microphonic may be more easily detected near its generation site with proper electrical instrumentation than is spontaneous otoacoustic emission.

  14. The Detection System for Existence and Direction of Emergency Vehicle using Microphone on-Vehicle

    NASA Astrophysics Data System (ADS)

    Otsuka, Shinichiro; Hara, Hironori; Ozawa, Shinji

    Problem may arise when the drivers are unable to notice or late in noticing emergency vehicles due to the sealing nature of the vehicles, masking by the car audio system or car navigation operation, which poses a threat for common vehicles and emergency vehicles to collide at many crossing. Every minute, problem may arise at every crossing and it is dangerous to wait for the completion of the information infrastructure system, which requires suitable cost and time. Therefore, we propose a system for common vehicles, which detects emergency vehicles using microphone on-vehicle and produce warning information to the driver. In this research, our focus is for “Ambulance", “Fire Truck" and “Police Car". The siren for these vehicles is detected by means of a microphone on-vehicle. This project utilizes two microphones to in order to detect the direction in which incoming emergency vehicles are present. The system, which we propose, must be helpful to response quickly to emergency and rescue activities.

  15. 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. PMID:22712924

  16. 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.

  17. Magnetic arrays

    DOEpatents

    Trumper, David L.; Kim, Won-jong; Williams, Mark E.

    1997-05-20

    Electromagnet arrays which can provide selected field patterns in either two or three dimensions, and in particular, which can provide single-sided field patterns in two or three dimensions. These features are achieved by providing arrays which have current densities that vary in the windings both parallel to the array and in the direction of array thickness.

  18. Magnetic arrays

    DOEpatents

    Trumper, D.L.; Kim, W.; Williams, M.E.

    1997-05-20

    Electromagnet arrays are disclosed which can provide selected field patterns in either two or three dimensions, and in particular, which can provide single-sided field patterns in two or three dimensions. These features are achieved by providing arrays which have current densities that vary in the windings both parallel to the array and in the direction of array thickness. 12 figs.

  19. 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

  20. Genetic optimisation of a plane array geometry for beamforming. Application to source localisation in a high speed train

    NASA Astrophysics Data System (ADS)

    Le Courtois, Florent; Thomas, Jean-Hugh; Poisson, Franck; Pascal, Jean-Claude

    2016-06-01

    Thanks to its easy implementation and robust performance, beamforming is applied for source localisation in several fields. Its effectiveness depends greatly on the array sensor configuration. This paper introduces a criterion to improve the array beampattern and increase the accuracy of sound source localisation. The beamwidth and the maximum sidelobe level are used to quantify the spatial variation of the beampattern through a new criterion. This criterion is shown to be useful, especially for the localisation of moving sources. A genetic algorithm is proposed for the optimisation of microphone placement. Statistical analysis of the optimised arrays provides original results on the algorithm performance and on the optimal microphone placement. An optimised array is tested to localise the sound sources of a high speed train. The results show an accurate separation.

  1. Development of high-speed, low-noise NIR HgCdTe avalanche photodiode arrays for adaptive optics and interferometry

    NASA Astrophysics Data System (ADS)

    Finger, Gert; Baker, Ian; Dorn, Reinhold; Eschbaumer, Siegfried; Ives, Derek; Mehrgan, Leander; Meyer, Manfred; Stegmeier, Jörg

    2010-07-01

    The most promising way to overcome the CMOS noise barrier of infrared AO sensors is the amplification of the photoelectron signal directly at the point of absorption inside the infrared pixel by means of the avalanche gain. HgCdTe eAPD arrays with cut off wavelengths of λc ~2.64 μm produced by SELEX-Galileo have been evaluated at ESO. The arrays were hybridized to an existing non-optimized ROIC developed for laser gated imaging which has a format of 320×256 pixels and four parallel video outputs. The avalanche gain makes it possible to reduce the read noise to < 7 e rms. The dark current requirements of IR wavefront sensing are also met.

  2. 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.

  3. 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. PMID:26627773

  4. Flexible retinal electrode array

    DOEpatents

    Okandan, Murat; Wessendorf, Kurt O.; Christenson, Todd R.

    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.

  5. 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

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

    PubMed

    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

  7. Kokkos Array

    SciTech Connect

    Edwards Daniel Sunderland, Harold Carter

    2012-09-12

    The Kokkos Array library implements shared-memory array data structures and parallel task dispatch interfaces for data-parallel computational kernels that are performance-portable to multicore-CPU and manycore-accelerator (e.g., GPGPU) devices.

  8. 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.

  9. 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. PMID:25373956

  10. Systolic arrays

    SciTech Connect

    Moore, W.R.; McCabe, A.P.H.; Vrquhart, R.B.

    1987-01-01

    Selected Contents of this book are: Efficient Systolic Arrays for the Solution of Toeplitz Systems, The Derivation and Utilization of Bit Level Systolic Array Architectures, an Efficient Systolic Array for Distance Computation Required in a Video-Codec Based Motion-Detection, On Realizations of Least-Squares Estimation and Kalman Filtering by Systolic Arrays, and Comparison of Systolic and SIMD Architectures for Computer Vision Computations.

  11. Nanocylinder arrays

    DOEpatents

    Tuominen, Mark; Schotter, Joerg; Thurn-Albrecht, Thomas; Russell, Thomas P.

    2009-08-11

    Pathways to rapid and reliable fabrication of nanocylinder arrays are provided. Simple methods are described for the production of well-ordered arrays of nanopores, nanowires, and other materials. This is accomplished by orienting copolymer films and removing a component from the film to produce nanopores, that in turn, can be filled with materials to produce the arrays. The resulting arrays can be used to produce nanoscale media, devices, and systems.

  12. Nanocylinder arrays

    DOEpatents

    Tuominen, Mark; Schotter, Joerg; Thurn-Albrecht, Thomas; Russell, Thomas P.

    2007-03-13

    Pathways to rapid and reliable fabrication of nanocylinder arrays are provided. Simple methods are described for the production of well-ordered arrays of nanopores, nanowires, and other materials. This is accomplished by orienting copolymer films and removing a component from the film to produce nanopores, that in turn, can be filled with materials to produce the arrays. The resulting arrays can be used to produce nanoscale media, devices, and systems.

  13. Primary calibration of measurement microphones in the world: state of art

    NASA Astrophysics Data System (ADS)

    Milhomem, T. A. B.; Soares, Z. M. D.

    2016-07-01

    This paper presents an overview of state of art of measurement microphones primary calibration in the world with emphasis on Brazil practices. Initially, pressure field calibration is summarized being discussed mainly the couplers used to create pressure field conditions. After that, free-field calibration is presented being commented especially the anechoic chambers used to create free-field conditions. Concluding, it is showed diffuse-field calibration that is being investigated. It is presented, in particular, the reverberant chambers used to create diffuse-field conditions.

  14. Recognition of Devoiced Vowels Using Optical Microphone Made of Multipled POF-Type Moisture Sensors

    NASA Astrophysics Data System (ADS)

    Morisawa, Masayuki; Natori, Yoichi; Taki, Tomohito; Muto, Shinzo

    A novel optical fiber microphone system for recognizing devoiced vowels has been studied. This system consists of the optical detection of moisture pattern formed by devoiced breath and its recognization process using a modified DP-matching. To detect moisture pattern of devoiced vowels, five plastic optical fiber moisture sensors with fast response were developed and used. Using this system, high discernment rate over 93% was obtained for the devoiced vowels. This system will be used for verbally handicapped people to create sounds with a small effort in the near future.

  15. Low-Level RF Control of Microphonics in Superconducting Spoke-Loaded Cavities

    SciTech Connect

    Conway, Z.A.; Kelly, M.P.; Sharamentov, S.I.; Shepard, K.W.; Davis, G.; Delayen, Jean; Doolittle, Lawrence

    2007-10-01

    This paper presents the results of cw RF frequency control and RF phase-stabilization experiments performed with a piezoelectric fast tuner mechanically coupled to a superconducting, 345 MHz, Ë = 0.5 triple-spoke-loaded cavity operating at 4.2K. The piezoelectric fast tuner damped low-frequency microphonic-noise by an order of magnitude. Two methods of RF phase-stabilization were characterized: overcoupling with negative phase feedback, and also fast mechanical tuner feedback. The Ë = 0.5 triple-spoke-loaded cavity RF field amplitude and phase errors were controlled to ±0.5% and ±30 respectively.

  16. Reducing the impact of wind noise on cochlear implant processors with two microphones

    PubMed Central

    Kokkinakis, Kostas; Cox, Casey

    2014-01-01

    Behind-the-ear (BTE) processors of cochlear implant (CI) devices offer little to almost no protection from wind noise in most incidence angles. To assess speech intelligibility, eight CI recipients were tested in 3 and 9 m/s wind. Results indicated that speech intelligibility decreased substantially when the wind velocity, and in turn the wind sound pressure level, increased. A two-microphone wind noise suppression strategy was developed. Scores obtained with this strategy indicated substantial gains in speech intelligibility over other conventional noise reduction strategies tested. PMID:24815292

  17. Low-level RF control of superconducting microphonics in spoke-loaded cavities.

    SciTech Connect

    Conway, Z. A.; Kelly, M. P.; Sharamentov, S. I.; Shepard, K. W.; Davis, G. K.; Delayen, J. R.; Doolittle, L. R.; TJNAF; LBNL

    2007-01-01

    This paper presents the results of cw RF frequency control and RF phase-stabilization experiments performed with a piezoelectric fast tuner mechanically coupled to a superconducting, 345 MHz, {beta} = 0.5 triple-spoke-loaded cavity operating at 4.2K. The piezoelectric fast tuner damped low-frequency microphonic-noise by an order of magnitude. Two methods of RF phase-stabilization were characterized: overcoupling with negative phase feedback, and also fast mechanical tuner feedback. The {beta} = 0.5 triple-spoke-loaded cavity RF field amplitude and phase errors were controlled to {+-} 0.5% and {+-} 30 respectively.

  18. 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.

  19. Numerical simulation of acoustic holography with propagator adaptation. Application to a 3D disc

    NASA Astrophysics Data System (ADS)

    Martin, Vincent; Le Bourdon, Thibault; Pasqual, Alexander Mattioli

    2011-08-01

    Acoustical holography can be used to identify the vibration velocity of an extended vibrating body. Such an inverse problem relies on the radiated acoustic pressure measured by a microphone array and on an a priori knowledge of the way the body radiates sound. Any perturbation on the radiation model leads to a perturbation on the velocity identified by the inversion process. Thus, to obtain the source vibration velocity with a good precision, it is useful to identify also an appropriate propagation model. Here, this identification, or adaptation, procedure rests on a geometrical interpretation of the acoustic holography in the objective space (here the radiated pressure space equipped with the L2-norm) and on a genetic algorithm. The propagator adaptation adds information to the holographic process, so it is not a regularisation method, which approximates the inverse of the model but does not affect the model. Moreover regularisations act in the variables space, here the velocities space. It is shown that an adapted model significantly decreases the quantity of regularisation needed to obtain a good reconstructed velocity, and that model adaptation improves significantly the acoustical holography results. In the presence of perturbations on the radiated pressure, some indications will be given on the interest or not to adapt the model, again thanks to the geometrical interpretation of holography in the objective space. As a numerical example, a disc whose vibration velocity on one of its sides is identified by acoustic holography is presented. On an industrial scale, this problem occurs due to the noise radiated by car wheels. The assessment of the holographic results has not yet been rigorously performed in such situations due to the complexity of the wheel environment made up of the car body, road and rolling conditions.

  20. Control of phased-array antennas

    NASA Astrophysics Data System (ADS)

    Samoilenko, V. I.; Shishov, Iu. A.

    Principles and algorithms for the control of phased arrays are described. Particular consideration is given to algorithms for the control of phase distribution, adaptive arrays, beam-steerable arrays, the design of phase shifters, the compensation of beam-pointing errors, and the calibration of high-gain antenna pointing.

  1. Estimation of low-altitude moving target trajectory using single acoustic array.

    PubMed

    Tong, Jianfei; Xie, Wei; Hu, Yu-Hen; Bao, Ming; Li, Xiaodong; He, Wei

    2016-04-01

    An acoustic-signature based method of estimating the flight trajectory of low-altitude flying aircraft that only requires a stationary microphone array is proposed. This method leverages the Doppler shifts of engine sound to estimate the closest point of approach distance, time, and speed. It also leverages the acoustic phase shift over the microphone array to estimate the direction of arrival of the target. Combining these parameters, this algorithm provides a total least square estimate of the target trajectory under the assumption of constant target height, direction, and speed. Analytical bounds of potential performance degradation due to noise are derived and the estimation error caused by signal propagation delay is analyzed, and both are verified with extensive simulation. The proposed algorithm is also validated by processing the data collected in field experiments. PMID:27106332

  2. 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. PMID:19655971

  3. Lamb-wave (X, Y) giant tap screen panel with built-in microphone and loudspeaker.

    PubMed

    Nikolovski, Jean-Pierre

    2013-06-01

    This paper presents a passive (X, Y) giant tap screen panel (GTP). Based on the time difference of arrival principle (TDOA), the device localizes low-energy impacts of around 1 mJ generated by fingernail taps. Selective detection of A0 Lamb waves generated in the upper frequency spectrum, around 100 kHz, makes it possible to detect light to strong impacts with equal resolution or precision, close to 1 cm and 2 mm, respectively, for a 10-mm-thick and 1-m(2) glass plate. Additionally, with glass, symmetrical beveling of the edges is used to create a tsunami effect that reduces the minimum impacting speed for light taps by a factor of three. Response time is less than 1 ms. Maximum panel size is of the order of 10 m(2). A rugged integrated flat design with embedded transducers in an electrically shielding frame features waterproof and sticker/ tag proof operation. Sophisticated electronics with floating amplification maintains the panel at its maximum possible sensitivity according to the surrounding noise. Amplification and filtering turns the panel into a microphone and loudspeaker featuring 50 mV/Pa as a microphone and up to 80 dBlin between 500 Hz and 8 kHz as a loudspeaker.

  4. A Fully On-Chip Gm-Opamp-RC Based Preamplifier for Electret Condenser Microphones

    NASA Astrophysics Data System (ADS)

    Le, Huy-Binh; Ryu, Seung-Tak; Lee, Sang-Gug

    An on-chip CMOS preamplifier for direct signal readout from an electret capacitor microphone has been designed with high immunity to common-mode and supply noise. The Gm-Opamp-RC based high impedance preamplifier helps to remove all disadvantages of the conventional JFET based amplifier and can drive a following switched-capacitor sigma-delta modulator in order to realize a compact digital electret microphone. The proposed chip is designed based on 0.18µm CMOS technology, and the simulation results show 86dB of dynamic range with 4.5µVrms of input-referred noise for an audio bandwidth of 20kHz and a total harmonic distortion (THD) of 1% at 90mVrms input. Power supply rejection ratio (PSRR) and common-mode rejection ration (CMRR) are more than 95dB at 1kHz. The proposed design dissipates 125µA and can operate over a wide supply voltage range of 1.6V to 3.3V.

  5. Acoustic scattering by circular cylinders of various aspect ratios. [pressure gradient microphones

    NASA Technical Reports Server (NTRS)

    Maciulaitis, A.

    1979-01-01

    The effects of acoustic scattering on the useful frequency range of pressure gradient microphones were investigated experimentally between ka values of 0.407 and 4.232 using two circular cylindrical models (L/D = 0.5 and 0.25) having a 25 cm outside diameter. Small condenser microphones, attached to preamplifiers by flexible connectors, were installed from inside the cylindrical bodies, and flush mounted on the exterior surface of the cylinders. A 38 cm diameter woofer in a large speaker enclosure was used as the sound source. Surface pressure augmentation and phase differences were computed from measured data for various sound wave incidence angles. Results are graphically compared with theoretical predictions supplied by NASA for ka = 0.407, 2.288, and 4.232. All other results are tabulated in the appendices. With minor exceptions, the experimentally determined pressure augmentations agreed within 0.75 dB with theoretical predictions. The agreement for relative phase angles was within 5 percent without any exceptions. Scattering parameter variations with ka and L/D ratio, as computed from experimental data, are also presented.

  6. 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.

  7. Lamb-wave (X, Y) giant tap screen panel with built-in microphone and loudspeaker.

    PubMed

    Nikolovski, Jean-Pierre

    2013-06-01

    This paper presents a passive (X, Y) giant tap screen panel (GTP). Based on the time difference of arrival principle (TDOA), the device localizes low-energy impacts of around 1 mJ generated by fingernail taps. Selective detection of A0 Lamb waves generated in the upper frequency spectrum, around 100 kHz, makes it possible to detect light to strong impacts with equal resolution or precision, close to 1 cm and 2 mm, respectively, for a 10-mm-thick and 1-m(2) glass plate. Additionally, with glass, symmetrical beveling of the edges is used to create a tsunami effect that reduces the minimum impacting speed for light taps by a factor of three. Response time is less than 1 ms. Maximum panel size is of the order of 10 m(2). A rugged integrated flat design with embedded transducers in an electrically shielding frame features waterproof and sticker/ tag proof operation. Sophisticated electronics with floating amplification maintains the panel at its maximum possible sensitivity according to the surrounding noise. Amplification and filtering turns the panel into a microphone and loudspeaker featuring 50 mV/Pa as a microphone and up to 80 dBlin between 500 Hz and 8 kHz as a loudspeaker. PMID:25004480

  8. Echo-intensity compensation in echolocating bats (Pipistrellus abramus) during flight measured by a telemetry microphone.

    PubMed

    Hiryu, Shizuko; Hagino, Tomotaka; Riquimaroux, Hiroshi; Watanabe, Yoshiaki

    2007-03-01

    An onboard microphone (Telemike) was developed to examine changes in the basic characteristics of echolocation sounds of small frequency-modulated echolocating bats, Pipistrellus abramus. Using a dual high-speed video camera system, spatiotemporal observations of echolocation characteristics were conducted on bats during a landing flight task in the laboratory. The Telemike allowed us to observe emitted pulses and returning echoes to which the flying bats listened during flight, and the acoustic parameters could be precisely measured without traditional problems such as the directional properties of the recording microphone and the emitted pulse, or traveling loss of the sound in the air. Pulse intensity in bats intending to land exhibited a marked decrease by 30 dB within 2 m of the target wall, and the reduction rate was approximately 6.5 dB per halving of distance. The intensity of echoes returning from the target wall indicated a nearly constant intensity (-42.6 +/- 5.5 dB weaker than the pulse emitted in search phase) within a target distance of 2 m. These findings provide direct evidence that bats adjust pulse intensity to compensate for changes in echo intensity to maintain a constant intensity of the echo returned from the approaching target at an optimal range.

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

    NASA Technical Reports Server (NTRS)

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

    1990-01-01

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

  10. 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.

  11. 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.

  12. 47 CFR 15.216 - Disclosure requirements for wireless microphones and other low power auxiliary stations capable...

    Code of Federal Regulations, 2012 CFR

    2012-10-01

    ... 47 Telecommunication 1 2012-10-01 2012-10-01 false Disclosure requirements for wireless... wireless microphones and other low power auxiliary stations capable of operating in the core TV bands. (a... by the Wireless Telecommunications Bureau and the Consumer and Governmental Affairs Bureau, at...

  13. 47 CFR 15.216 - Disclosure requirements for wireless microphones and other low power auxiliary stations capable...

    Code of Federal Regulations, 2013 CFR

    2013-10-01

    ... 47 Telecommunication 1 2013-10-01 2013-10-01 false Disclosure requirements for wireless... wireless microphones and other low power auxiliary stations capable of operating in the core TV bands. (a... by the Wireless Telecommunications Bureau and the Consumer and Governmental Affairs Bureau, at...

  14. 47 CFR 15.216 - Disclosure requirements for wireless microphones and other low power auxiliary stations capable...

    Code of Federal Regulations, 2014 CFR

    2014-10-01

    ... 47 Telecommunication 1 2014-10-01 2014-10-01 false Disclosure requirements for wireless... wireless microphones and other low power auxiliary stations capable of operating in the core TV bands. (a... by the Wireless Telecommunications Bureau and the Consumer and Governmental Affairs Bureau, at...

  15. 47 CFR 15.216 - Disclosure requirements for wireless microphones and other low power auxiliary stations capable...

    Code of Federal Regulations, 2011 CFR

    2011-10-01

    ... 47 Telecommunication 1 2011-10-01 2011-10-01 false Disclosure requirements for wireless... wireless microphones and other low power auxiliary stations capable of operating in the core TV bands. (a... by the Wireless Telecommunications Bureau and the Consumer and Governmental Affairs Bureau, at...

  16. 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

  17. 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…

  18. A laboratory study on a capacitive displacement sensor as an implant microphone in totally implant cochlear hearing aid systems.

    PubMed

    Huang, Ping; Guo, Jun; Megerian, Cliff A; Young, Darrin J; Ko, Wen H

    2007-01-01

    A totally implant cochlear hearing aids system, integrating an implant microphone, interface electronics, a speech processor, a stimulator, and cochlear electrodes, can overcome the uncomfortable, inconvenient, and stigma problems associated with the conventional and semi-implantable hearing aids. This paper presents a laboratory feasibility study on the use of an electret condenser microphone (ECM) displacement sensor, serving as an implant microphone, and combined with a spring coupler to directly sense the umbo acoustic vibration. The umbo vibration characteristics were extracted from literature to determine the coupler and sensor requirements. A laboratory model was built to simulate the vibration source and experimentally study the transmission coefficient. Experimental data demonstrate that by using a 5 N/m stiffness spring, the umbo vibration amplitude as high as 67% can be transmitted to the sensor. Measurement of the sensor system on the temporal bone was also made. The minimum detectable sound pressure level (SPL) at 1 kHz is 41 and 67 dB for laboratory and 38 and 64 dB for temporal bone measurement for 1 and 388 Hz bandwidth, respectively. Better performance was achieved in a higher frequency. Results and analysis of this study can be used as a guideline for the future design of displacement sensors as implant microphones. PMID:18003304

  19. 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.

  20. 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

  1. Note: Electronic damping of microphonics in superconducting resonators of a continuous wave linac

    SciTech Connect

    Joshi, Gopal; Sahu, Bhuban Kumar; Agarwal, Vivek; Kumar, Girish

    2014-02-15

    The paper presents an implementation technique to damp the microphonics in superconducting resonators utilizing the coupling between the electromagnetic and the mechanical modes of a resonator. In the technique used the resonant frequency variations are fed back to modulate the field amplitude through a suitable transfer function. Of the two transfer functions used in the experiments, one emulates a derivative action and is placed in a negative feedback configuration. The other transfer function is essentially a parallel combination of second order low pass filters and is used in a positive feedback configuration. Experiments with the Quarter Wave resonators of IUAC, New Delhi linac demonstrate that the damping of some of the modes increases significantly with the introduction of this feedback leading to a reduction in power required for field stabilization and quieter operation of the RF control system.

  2. Effect of intense sound exposure on cochlear microphonics and whole nerve action potential

    NASA Astrophysics Data System (ADS)

    Yamamura, K.; Yamamoto, N.; Kohyama, A.; Sawada, Y.; Ohno, H.; Saitoh, Y.

    1989-06-01

    An investigation was carried out to determine whether or not the critical band with Temporary Threshold Shift (TTS) is affected by exposure to high frequency sound. The function of the cochlea and the 8th nerve in guinea pigs was estimated by the intensity function and maximum output voltage of cochlear microphonics (CM) and by whole nerve action potential (Ap). Our results showed that both the intensity function and the maximum output voltage of CM and Ap decreased. Ap obtained at the test frequency higher, by half an octave, than the center frequency of the exposure noise was especially lowered. These results suggest that the critical band with TTS of both Ap and CM may be affected in exposure to high frequency sound.

  3. Design and Development of a High Impedance Amplifier For Use With Piezoelectric Infrasound Microphones

    NASA Astrophysics Data System (ADS)

    Kleinert, D. E.; Talmadge, C. L.

    2011-12-01

    The National Center for Physical Acoustics (NCPA) has developed a new class of high fidelity low cost piezoelectric infrasound sensors. One of the key electronic issues has been the design and development of the appropriate high impedance amplifiers including material specification as well as circuit layout and fabrication. The high impedance amplifier is required to allow the piezoelectronic sensor to operate over its entire bandwidth as the sensor itself has high impedance at the low frequency end of its operation. The specifications include a flat frequency response from at least .01 Hz to 500 Hz, a dynamic range suitable to feed a 24 bit ADC and reasonably low power (mW levels). There has been extensive field testing of the resulting amplifier in conjunction with the piezoelectric microphone, also developed at NCPA, in a variety of locations and climates using various sources, including hurricanes, tornados and high explosive detonations.

  4. Infrastructure-Less Indoor Localization Using the Microphone, Magnetometer and Light Sensor of a Smartphone.

    PubMed

    Galván-Tejada, Carlos E; García-Vázquez, Juan Pablo; Galván-Tejada, Jorge I; Delgado-Contreras, J Rubén; Brena, Ramon F

    2015-01-01

    In this paper, we present the development of an infrastructure-less indoor location system (ILS), which relies on the use of a microphone, a magnetometer and a light sensor of a smartphone, all three of which are essentially passive sensors, relying on signals available practically in any building in the world, no matter how developed the region is. In our work, we merge the information from those sensors to estimate the user's location in an indoor environment. A multivariate model is applied to find the user's location, and we evaluate the quality of the resulting model in terms of sensitivity and specificity. Our experiments were carried out in an office environment during summer and winter, to take into account changes in light patterns, as well as changes in the Earth's magnetic field irregularities. The experimental results clearly show the benefits of using the information fusion of multiple sensors when contrasted with the use of a single source of information. PMID:26295237

  5. Effects of lead acetate on guinea pig - cochear microphonics, action potential, and motor nerve conduction velocity

    SciTech Connect

    Yamamura, K.; Maehara, N.; Terayama, K.; Ueno, N.; Kohyama, A.; Sawada, Y.; Kishi, R.

    1987-04-01

    Segmental demyelination and axonal degeneration of motor nerves induced by lead exposure is well known in man, and animals. The effect of lead acetate exposure to man may involve the cranial nerves, since vertigo and sensory neuronal deafness have been reported among lead workers. However, there are few reports concerning the dose-effects of lead acetate both to the peripheral nerve and the cranial VII nerve with measurement of blood lead concentration. The authors investigated the effects of lead acetate to the cochlea and the VIII nerve using CM (cochlear microphonics) and AP (action potential) of the guinea pigs. The effects of lead acetate to the sciatic nerve were measured by MCV of the sciatic nerve with measurement of blood lead concentration.

  6. Infrastructure-Less Indoor Localization Using the Microphone, Magnetometer and Light Sensor of a Smartphone.

    PubMed

    Galván-Tejada, Carlos E; García-Vázquez, Juan Pablo; Galván-Tejada, Jorge I; Delgado-Contreras, J Rubén; Brena, Ramon F

    2015-08-18

    In this paper, we present the development of an infrastructure-less indoor location system (ILS), which relies on the use of a microphone, a magnetometer and a light sensor of a smartphone, all three of which are essentially passive sensors, relying on signals available practically in any building in the world, no matter how developed the region is. In our work, we merge the information from those sensors to estimate the user's location in an indoor environment. A multivariate model is applied to find the user's location, and we evaluate the quality of the resulting model in terms of sensitivity and specificity. Our experiments were carried out in an office environment during summer and winter, to take into account changes in light patterns, as well as changes in the Earth's magnetic field irregularities. The experimental results clearly show the benefits of using the information fusion of multiple sensors when contrasted with the use of a single source of information.

  7. Development, fabrication and calibration of a porous surface microphone in an aerofoil

    NASA Technical Reports Server (NTRS)

    Noiseux, D. U.; Noiseux, N. B.; Kadman, Y.

    1975-01-01

    The development of a porous surface microphone in an airfoil intended to measure acoustic signals in a turbulent airflow and to minimize the flow noise is described. The sensor because of its airfoil operates over a wide range of yaw angles and flow velocities without excessive flow noise. The acoustic properties of the porous materials used in the airfoil sensor and their effects on the frequency response of the sensor were analyzed and tested. An accurate airfoil was selected, having a smaller thickness-to-chord ratio and an airfoil sensor was designed. The sensor was calibrated acoustically and its flow noise evaluated in the quiet BBN wind tunnel at flow velocities up to 70 m/sec. Results are presented.

  8. A Novel Vibration Mode Testing Method for Cylindrical Resonators Based on Microphones

    PubMed Central

    Zhang, Yongmeng; Wu, Yulie; Wu, Xuezhong; Xi, Xiang; Wang, Jianqiu

    2015-01-01

    Non-contact testing is an important method for the study of the vibrating characteristic of cylindrical resonators. For the vibratory cylinder gyroscope excited by piezo-electric electrodes, mode testing of the cylindrical resonator is difficult. In this paper, a novel vibration testing method for cylindrical resonators is proposed. This method uses a MEMS microphone, which has the characteristics of small size and accurate directivity, to measure the vibration of the cylindrical resonator. A testing system was established, then the system was used to measure the vibration mode of the resonator. The experimental results show that the orientation resolution of the node of the vibration mode is better than 0.1°. This method also has the advantages of low cost and easy operation. It can be used in vibration testing and provide accurate results, which is important for the study of the vibration mode and thermal stability of vibratory cylindrical gyroscopes. PMID:25602269

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

    SciTech Connect

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

    2008-11-15

    An active device for superheated emulsion detector is described. A capacitive diaphragm sensor or condenser microphone is used to convert the acoustic pulse of drop nucleation to electrical signal. An active peak detector is included in the circuit to avoid multiple triggering of the counter. The counts are finally recorded by a microprocessor based data acquisition system. Genuine triggers, missed by the sensor, were studied using a simulated clock pulse. The neutron energy spectrum of {sup 252}Cf fission neutron source was measured using the device with R114 as the sensitive liquid and compared with the calculated fission neutron energy spectrum of {sup 252}Cf. Frequency analysis of the detected signals was also carried out.

  10. How to measure snoring? A comparison of the microphone, cannula and piezoelectric sensor.

    PubMed

    Arnardottir, Erna S; Isleifsson, Bardur; Agustsson, Jon S; Sigurdsson, Gunnar A; Sigurgunnarsdottir, Magdalena O; Sigurđarson, Gudjon T; Saevarsson, Gudmundur; Sveinbjarnarson, Atli T; Hoskuldsson, Sveinbjorn; Gislason, Thorarinn

    2016-04-01

    The objective of this study was to compare to each other the methods currently recommended by the American Academy of Sleep Medicine (AASM) to measure snoring: an acoustic sensor, a piezoelectric sensor and a nasal pressure transducer (cannula). Ten subjects reporting habitual snoring were included in the study, performed at Landspitali-University Hospital, Iceland. Snoring was assessed by listening to the air medium microphone located on a patient's chest, compared to listening to two overhead air medium microphones (stereo) and manual scoring of a piezoelectric sensor and nasal cannula vibrations. The chest audio picked up the highest number of snore events of the different snore sensors. The sensitivity and positive predictive value of scoring snore events from the different sensors was compared to the chest audio: overhead audio (0.78, 0.98), cannula (0.55, 0.67) and piezoelectric sensor (0.78, 0.92), respectively. The chest audio was capable of detecting snore events with lower volume and higher fundamental frequency than the other sensors. The 200 Hz sampling rate of the cannula and piezoelectric sensor was one of their limitations for detecting snore events. The different snore sensors do not measure snore events in the same manner. This lack of consistency will affect future research on the clinical significance of snoring. Standardization of objective snore measurements is therefore needed. Based on this paper, snore measurements should be audio-based and the use of the cannula as a snore sensor be discontinued, but the piezoelectric sensor could possibly be modified for improvement.

  11. Use of adaptive network burst detection methods for multielectrode array data and the generation of artificial spike patterns for method evaluation

    NASA Astrophysics Data System (ADS)

    Mendis, G. D. C.; Morrisroe, E.; Petrou, S.; Halgamuge, S. K.

    2016-04-01

    Objective. Multielectrode arrays are an informative extracellular recording technology that enables the analysis of cultured neuronal networks and network bursts (NBs) are a dominant feature observed in these recordings. This paper focuses on the validation of NB detection methods on different network activity patterns and developing a detection method that performs robustly across a wide variety of activity patterns. Approach. A firing rate based approach was used to generate artificial spike timestamps where NBs were introduced as episodes where the probability of spiking increases. Variations in firing and bursting characteristics were also included. In addition, an improved methodology of detecting NBs is proposed, based on time-binned average firing rates and time overlaps of single channel bursts. The robustness of the proposed method was compared against three existing algorithms using simulated, publicly available and newly acquired data. Main results. A range of activity patterns were generated by changing simulation variables that correspond to NB duration (40-2200 ms), intervals (0.3-16 s), firing rates (0.1-1 spikes s-1), local burst percentage (0%-90%), number of channels in local bursts (20-40) as well as the number of tonic and frequently-bursting channels. By extracting simulation parameters directly from real data, we generated synthetic data that closely resemble activity of mouse and rat cortical cultures at native and chemically perturbed states. In 50 simulated data sets with randomly selected parameter values, the improved NB detection method performed better (ascertained by the f-measure) than three existing methods (p < 0.005). The improved method was also able to detect clustered, long-tailed and short-frequent NBs on real data. Significance. This work presents an objective method of assessing the applicability of NB detection methods for different neuronal activity patterns. Furthermore, it proposes an improved NB detection method that can

  12. 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.

  13. Adaptive behavior for texture discrimination by the free-flying big brown bat, Eptesicus fuscus.

    PubMed

    Falk, Ben; Williams, Tameeka; Aytekin, Murat; Moss, Cynthia F

    2011-05-01

    This study examined behavioral strategies for texture discrimination by echolocation in free-flying bats. Big brown bats, Eptesicus fuscus, were trained to discriminate a smooth 16 mm diameter object (S+) from a size-matched textured object (S-), both of which were tethered in random locations in a flight room. The bat's three-dimensional flight path was reconstructed using stereo images from high-speed video recordings, and the bat's sonar vocalizations were recorded for each trial and analyzed off-line. A microphone array permitted reconstruction of the sonar beam pattern, allowing us to study the bat's directional gaze and inspection of the objects. Bats learned the discrimination, but performance varied with S-. In acoustic studies of the objects, the S+ and S- stimuli were ensonified with frequency-modulated sonar pulses. Mean intensity differences between S+ and S- were within 4 dB. Performance data, combined with analyses of echo recordings, suggest that the big brown bat listens to changes in sound spectra from echo to echo to discriminate between objects. Bats adapted their sonar calls as they inspected the stimuli, and their sonar behavior resembled that of animals foraging for insects. Analysis of sonar beam-directing behavior in certain trials clearly showed that the bat sequentially inspected S+ and S-.

  14. Determining Direction of Arrival at a Y-Shaped Antenna Array

    NASA Technical Reports Server (NTRS)

    Starr, Stan

    2003-01-01

    An algorithm computes the direction of arrival (both azimuth and elevation angles) of a lightning-induced electromagnetic signal from differences among the times of arrival of the signal at four antennas in a Y-shaped array on the ground. In the original intended application of the algorithm, the baselines of the array are about 90 m long and the array is part of a lightning-detection-and-ranging (LDAR) system. The algorithm and its underlying equations can also be used to compute directions of arrival of impulsive phenomena other than lightning on arrays of sensors other than radio antennas: for example, of an acoustic pulse arriving at an array of microphones.

  15. 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…

  16. Wall-Pressure-Array Measurements Beneath a Separating/Reattaching Flow Region

    NASA Astrophysics Data System (ADS)

    Hudy, Laura; Naguib, Ahmed; Humphreys, William; Bartram, Scott

    2000-11-01

    The surface-pressure signature of the structure within a separated flow region was investigated using a microphone array. The experimental set-up consisted of a splitter plate instrumented with 80 flush-mounted Panasonic electret microphones behind a fence attached perpendicular to the plate. Additionally, static pressure taps were positioned on the top and the bottom of the splitter plate and were used to align the model in the NASA Langley Subsonic Basic Research Tunnel. Data were acquired for two Reynolds numbers of 8000 and 10500, based on the fence height. A spatio-temporal analysis was conducted on the measurements in the time as well as the frequency domain. Results revealed the dominant flow modes in the separating shear layer and their convective characteristics. Furthermore, the relationship between the shear layer modes and the low-frequency oscillation of the reattachment zone was examined.

  17. Microlens arrays

    NASA Astrophysics Data System (ADS)

    Hutley, Michael C.; Stevens, Richard F.; Daly, Daniel J.

    1992-04-01

    Microlenses have been with us for a long time as indeed the very word lens reminds us. Many early lenses,including those made by Hooke and Leeuwenhoek in the 17th century were small and resembled lentils. Many languages use the same word for both (French tilentillelt and German "Linse") and the connection is only obscure in English because we use the French word for the vegetable and the German for the optic. Many of the applications for arrays of inicrolenses are also well established. Lippmann's work on integral photography at the turn of the century required lens arrays and stimulated an interest that is very much alive today. At one stage, lens arrays played an important part in high speed photography and various schemes have been put forward to take advantage of the compact imaging properties of combinations of lens arrays. The fact that many of these ingenious schemes have not been developed to their full potential has to a large degree been due to the absence of lens arrays of a suitable quality and cost.

  18. Ears of the Robot: Direction of Arrival Estimation Based on Pattern Recognition Using Robot-Mounted Microphones

    NASA Astrophysics Data System (ADS)

    Mochiki, Naoya; Ogawa, Tetsuji; Kobayashi, Tetsunori

    We propose a new type of direction-of-arrival estimation method for robot audition that is free from strict head related transfer function estimation. The proposed method is based on statistical pattern recognition that employs a ratio of power spectrum amplitudes occurring for a microphone pair as a feature vector. It does not require any phase information explicitly, which is frequently used in conventional techniques, because the phase information is unreliable for the case in which strong reflections and diffractions occur around the microphones. The feature vectors we adopted can treat these influences naturally. The effectiveness of the proposed method was shown from direction-of-arrival estimation tests for 19 kinds of directions: 92.4% of errors were reduced compared with the conventional phase-based method.

  19. Quantum cascade laser based standoff photoacoustic detection of explosives using ultra-sensitive microphone and sound reflector

    NASA Astrophysics Data System (ADS)

    Chen, Xing; Guo, Dingkai; Choa, Fow-Sen; Wang, Chen-Chia; Trivedi, Sudhir; Fan, Jenyu

    2013-01-01

    We report standoff detection of explosives using quantum cascade laser (QCL) and photoacoustic technique. In our experiment, a QCL with emission wavelength near 7.35 μm was used and operated at pulsed mode. The output light was focused on Trinitrotoluene (TNT) sample in its powder form. Photoacoustic signals were generated and detected by an ultra-sensitive low-noise microphone with one inch diameter. A detection distance up to 8 inches was obtained using the microphone alone. With the increasing detection distance the measured photoacoustic signal not only decayed in amplitude but also delayed in phase, which clearly verified the source location. To further increase the detection distance, a parabolic sound reflector was used for effective sound collection. With the help of the sound reflector, standoff photoacoustic detection of TNT with distance of 8 feet was demonstrated.

  20. Adaptive antennas

    NASA Astrophysics Data System (ADS)

    Barton, P.

    1987-04-01

    The basic principles of adaptive antennas are outlined in terms of the Wiener-Hopf expression for maximizing signal to noise ratio in an arbitrary noise environment; the analogy with generalized matched filter theory provides a useful aid to understanding. For many applications, there is insufficient information to achieve the above solution and thus non-optimum constrained null steering algorithms are also described, together with a summary of methods for preventing wanted signals being nulled by the adaptive system. The three generic approaches to adaptive weight control are discussed; correlation steepest descent, weight perturbation and direct solutions based on sample matrix conversion. The tradeoffs between hardware complexity and performance in terms of null depth and convergence rate are outlined. The sidelobe cancellor technique is described. Performance variation with jammer power and angular distribution is summarized and the key performance limitations identified. The configuration and performance characteristics of both multiple beam and phase scan array antennas are covered, with a brief discussion of performance factors.

  1. A study of blast waveforms detected simultaneously by a microphone and a laser probe during laser ablation

    NASA Astrophysics Data System (ADS)

    Diaci, J.; Možina, J.

    1992-10-01

    We examine blast waves generated in air during irradiation of absorbing samples with Nd: YAG laser pulses of fluences exceeding the ablation threshold. Blast waves were detected simultaneously by a wideband microphone and a laser beam deflection probe. By a comparative analysis of both signals in the time and frequency domain we investigate characteristic features of their nonlinear waveform evolution. To explain the observed phenomena we employ the weak shock solution of the point explosion model.

  2. [INVITED] A miniaturized optical fiber microphone with concentric nanorings grating and microsprings structured diaphragm

    NASA Astrophysics Data System (ADS)

    Wang, Hui; Xie, Zhenwei; Zhang, Mile; Cui, Hailin; He, Jingsuo; Feng, Shengfei; Wang, Xinke; Sun, Wenfeng; Ye, Jiasheng; Han, Peng; Zhang, Yan

    2016-04-01

    A miniaturized optical fiber microphone (OFM) is created by fabricating a concentric nanorings grating and microsprings structured half spherical diaphragm on the end facet of a single-mode fiber (SMF). The diaphragm is fabricated via the method of two-photon 3D lithography. The thin nanorings grating patterned diaphragm is actually a resonant grating-waveguide. It exhibits high reflectivity when resonance is excited. A microlens is fabricated at the core of the fiber, which is used to diverge the output light to make it be normally incident onto the diaphragm, then reflected back to the fiber. The intensities of the reflected back light will be changed if the resonant conditions of the resonant grating-waveguide are broken due to the sound pressure induced geometrical changes of the configuration. This makes such device be an acoustic sensor. The microsprings are designed to improve the sensitivity to the sound pressure. Acoustic inspections show that this OFM can detect the weak sound in air with frequency band from 400 to 2000 Hz.

  3. Using Concha Electrodes to Measure Cochlear Microphonic Waveforms and Auditory Brainstem Responses

    PubMed Central

    Zhang, Ming

    2010-01-01

    During electrocochleography, that is, ECochG or ECoG, a recording electrode can be placed in the ear canal lateral to the tympanic membrane. We designed a concha electrode to record both sinusoidal waveforms of cochlear microphonics (CMs) and auditory brainstem responses (ABRs). The amplitudes of CM waveforms and Wave I or compound action potentials (CAPs) recorded at the concha were greater than those recorded at the mastoid but slightly lower than those recorded at the ear canal. Wave V amplitudes recorded at the concha were greater than those recorded at the ear canal but lower than those recorded at the mastoid. There was not a significant difference between the amplitudes recorded at the concha and at the ear canal. For CM and Wave I or CAP, the latency recorded at the concha was longer than at the canal but shorter than at the mastoid; for Wave V, the reverse was true. However, these differences were not statistically significant and may be due to the distance to response generators. Aside from the advantages that the regular ECoG has over otoacoustic emission (OAE) testing, the concha electrode was also easier and safer to place and may be suitable for children, newborn screening, participants with canal conditions, and remote clinics which could have concerns with the availability and cost of a canal electrode. Using concha electrodes, we also experienced fewer postauricular artifacts than when using a mastoid electrode. PMID:21131635

  4. Separating Turbofan Engine Noise Sources Using Auto and Cross Spectra from Four Microphones

    NASA Technical Reports Server (NTRS)

    Miles, Jeffrey Hilton

    2008-01-01

    The study of core noise from turbofan engines has become more important as noise from other sources such as the fan and jet were reduced. A multiple-microphone and acoustic-source modeling method to separate correlated and uncorrelated sources is discussed. The auto- and cross spectra in the frequency range below 1000 Hz are fitted with a noise propagation model based on a source couplet consisting of a single incoherent monopole source with a single coherent monopole source or a source triplet consisting of a single incoherent monopole source with two coherent monopole point sources. Examples are presented using data from a Pratt& Whitney PW4098 turbofan engine. The method separates the low-frequency jet noise from the core noise at the nozzle exit. It is shown that at low power settings, the core noise is a major contributor to the noise. Even at higher power settings, it can be more important than jet noise. However, at low frequencies, uncorrelated broadband noise and jet noise become the important factors as the engine power setting is increased.

  5. Cochlear microphonics recordable at the non-shielded bedside using a new tubal transducer.

    PubMed

    Nishida, H; Komatsuzaki, A; Noguchi, Y

    1997-01-01

    A new tubal transducer (NC-3) for measuring cochlear microphonics (CM) in extratympanic electrocochleography (ECochG) was developed by improving the common hearing aid earphone. Using a human forearm as a dummy ear, the artifact contamination generated from the NC-3 tubal transducer was tested and the possibility of measuring the CM at a non-shielded bedside was studied. An HN-5 electrode was fixed to a subject's forearm, and a sound stimulus of 90 dBnHL was delivered through the tube of the NC-3. When the earphone of the transducer was placed at a right-angle to the electrode on either a vertical or horizontal plane and the electrode was placed in direct contact with the tip of the tube, contamination from electromagnetic induction and CM-like mechanical vibration were prevented. Using the HN-5 electrode and NC-3, extratympanic ECochG-CM was recorded from normal-hearing subjects in both a shielded soundproof room and a non-shielded ordinary, quiet room. No differences were found between CMs measured in the two rooms. These results suggest that the NC-3 overcomes the shortcomings of a loudspeaker system and allows CM to be recorded accurately at non-shielded bedsides.

  6. The influence of inner hair cell loss on the instantaneous frequency of the cochlear microphonic.

    PubMed

    Chertoff, Mark E; Amani-Taleshi, Darius; Guo, Yuqing; Burkard, Robert

    2002-12-01

    The cochlear microphonic (CM) is produced by a change in standing currents during the motion of the cochlear partition. The motion of the partition and associated hair cell transduction processes are nonlinear and are reflected in the variation of the instantaneous frequency (IF) of the CM. Although the CM is dominated from receptor currents from outer hair cells (OHCs), receptor currents from inner hair cells (IHCs) may contribute to the fluctuation in the IF. In this paper we examine the influence of IHCs on the variation of the IF of the CM. A 75 mg/kg intraperitoneal (i.p.) dose of carboplatin reduced the IHC population by approximately 40%. The reduction in IHCs did not substantially affect the amplitude of the CM. The amplitude of the IF, however, was reduced at high signal levels (90 and 100 dB peak SPL). A phenomenological model of the CM indicated that the contribution of IHC receptor currents to the IF was small and that changes in OHC transducer characteristics may have a greater impact on the IF. PMID:12433400

  7. Measurement of Vertical Temperature Distribution Using a Single Pair of Loudspeaker and Microphone with Acoustic Reflection

    NASA Astrophysics Data System (ADS)

    Saito, Ikumi; Mizutani, Koichi; Wakatsuki, Naoto; Kawabe, Satoshi

    2009-07-01

    It is important to maintain an adequate indoor temperature for comfortable working conditions, improvement of the rate of production of farm goods grown in greenhouses, and for saving energy. Thus, it is necessary to measure the temperature distribution to realize efficient air-conditioning systems. However, we have to use many conventional instruments to measure the temperature distribution. We proposed a measurement system for vertical temperature distribution using a single pair of loudspeaker (SP) and microphone (MIC), and acoustic reflectors. This system consists of SP, MIC, and multiple acoustic reflectors, and it can be used to determine the temperature distribution from the mean temperature of the area bounded by two reflectors. In experiments, the vertical temperature distribution was measured using five sound probes in a large facility every 20 s for 24 h. From the results of this experiment, it was verified that this system can be used to measure the vertical temperature distribution from the mean temperature of each area bounded by two reflectors. This system could be used to measure the change in the temperature distribution over time. We constructed a simple system to measure the vertical temperature distribution.

  8. Infrastructure-Less Indoor Localization Using the Microphone, Magnetometer and Light Sensor of a Smartphone

    PubMed Central

    Galván-Tejada, Carlos E.; García-Vázquez, Juan Pablo; Galván-Tejada, Jorge I.; Delgado-Contreras, J. Rubén; Brena, Ramon F.

    2015-01-01

    In this paper, we present the development of an infrastructure-less indoor location system (ILS), which relies on the use of a microphone, a magnetometer and a light sensor of a smartphone, all three of which are essentially passive sensors, relying on signals available practically in any building in the world, no matter how developed the region is. In our work, we merge the information from those sensors to estimate the user’s location in an indoor environment. A multivariate model is applied to find the user’s location, and we evaluate the quality of the resulting model in terms of sensitivity and specificity. Our experiments were carried out in an office environment during summer and winter, to take into account changes in light patterns, as well as changes in the Earth’s magnetic field irregularities. The experimental results clearly show the benefits of using the information fusion of multiple sensors when contrasted with the use of a single source of information. PMID:26295237

  9. 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.

  10. High density arrays of micromirrors

    SciTech Connect

    Folta, J. M.; Decker, J. Y.; Kolman, J.; Lee, C.; Brase, J. M.

    1999-02-01

    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.

  11. Global Arrays

    2006-02-23

    The Global Arrays (GA) toolkit provides an efficient and portable “shared-memory” programming interface for distributed-memory computers. Each process in a MIMD parallel program can asynchronously access logical blocks of physically distributed dense multi-dimensional arrays, without need for explicit cooperation by other processes. Unlike other shared-memory environments, the GA model exposes to the programmer the non-uniform memory access (NUMA) characteristics of the high performance computers and acknowledges that access to a remote portion of the sharedmore » data is slower than to the local portion. The locality information for the shared data is available, and a direct access to the local portions of shared data is provided. Global Arrays have been designed to complement rather than substitute for the message-passing programming model. The programmer is free to use both the shared-memory and message-passing paradigms in the same program, and to take advantage of existing message-passing software libraries. Global Arrays are compatible with the Message Passing Interface (MPI).« less

  12. Pacific Array

    NASA Astrophysics Data System (ADS)

    Kawakatsu, H.; Takeo, A.; Isse, T.; Nishida, K.; Shiobara, H.; Suetsugu, D.

    2014-12-01

    Based on our recent results on broadband ocean bottom seismometry, we propose a next generation large-scale array experiment in the ocean. Recent advances in ocean bottom broadband seismometry (e.g., Suetsugu & Shiobara, 2014, Annual Review EPS), together with advances in the seismic analysis methodology, have now enabled us to resolve the regional 1-D structure of the entire lithosphere/asthenosphere system, including seismic anisotropy (both radial and azimuthal), with deployments of ~10-15 broadband ocean bottom seismometers (BBOBSs) (namely "ocean-bottom broadband dispersion survey"; Takeo et al., 2013, JGR; Kawakatsu et al., 2013, AGU; Takeo, 2014, Ph.D. Thesis; Takeo et al., 2014, JpGU). Having ~15 BBOBSs as an array unit for 2-year deployment, and repeating such deployments in a leap-frog way (an array of arrays) for a decade or so would enable us to cover a large portion of the Pacific basin. Such efforts, not only by giving regional constraints on the 1-D structure, but also by sharing waveform data for global scale waveform tomography, would drastically increase our knowledge of how plate tectonics works on this planet, as well as how it worked for the past 150 million years. International collaborations might be sought.

  13. Use of an accelerometer and a microphone as gas detectors in the online quantitative detection of hydrogen released from ammonia borane by gas chromatography.

    PubMed

    He, Yi-San; Chen, Kuan-Fu; Lin, Chien-Hung; Lin, Min-Tsung; Chen, Chien-Chung; Lin, Cheng-Huang

    2013-03-19

    The use of an accelerometer as a gas detector in gas chromatography (GC) is described for the first time. A milli-whistle was connected to the outlet of the GC capillary. When the eluted and GC carrier gases pass through the capillary and milli-whistle, a sound is produced. After a fast Fourier transform (FFT), the sound wave generated from the milli-whistle is picked up by a microphone and the resulting vibration of the milli-whistle body can be recorded by an accelerometer. The release of hydrogen gas, as the result of thermal energy, from ammonia borane (NH3BH3), which has been suggested as a storage medium for hydrogen, was selected as the model sample. The findings show that the frequencies generated, either by sound or by the vibration from the whistle body, were identical. The concentration levels of the released hydrogen gas can be determined online, based on the frequency changes. Ammonia borane was placed in a brass reservoir, heated continually, and the released hydrogen gas was directly injected into the GC inlet at 0.5 min intervals, using a home-built electromagnetic pulse injector. The concentration of hydrogen for each injection can be calculated immediately. When the ammonia borane was encapsulated within a polycarbonate (PC) microtube array membrane, the temperature required for the release of hydrogen can be decreased, which would make such a material more convenient for use. The findings indicate that 1.0 mg of ammonia borane can produce hydrogen in the range of 1.0-1.25 mL, in the temperature range of 85-115 °C.

  14. 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.

  15. The indirect binary n-cube array

    NASA Technical Reports Server (NTRS)

    Pease, M. C.

    1977-01-01

    The array is built from a large number (hundreds or thousands) of microprocessors or microcomputers linked through a switching network into an indirect binary n-cube array. Control is two level, the array operating synchronously, or in lock step, at the higher level, and with the broadcast commands being locally interpreted into rewritable microinstruction streams in the microprocessors and in the switch control units. The key to the design is the switching array. By properly programming it, the array can be made into a wide variety of virtual arrays which are well adapted to a wide range of applications. It is believed that the flexibility of the switching array can be used to obtain fault avoidance, which appears necessary in any highly parallel design.

  16. 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.

  17. Automatic Detection of Whole Night Snoring Events Using Non-Contact Microphone

    PubMed Central

    Dafna, Eliran; Tarasiuk, Ariel; Zigel, Yaniv

    2013-01-01

    Objective Although awareness of sleep disorders is increasing, limited information is available on whole night detection of snoring. Our study aimed to develop and validate a robust, high performance, and sensitive whole-night snore detector based on non-contact technology. Design Sounds during polysomnography (PSG) were recorded using a directional condenser microphone placed 1 m above the bed. An AdaBoost classifier was trained and validated on manually labeled snoring and non-snoring acoustic events. Patients Sixty-seven subjects (age 52.5±13.5 years, BMI 30.8±4.7 kg/m2, m/f 40/27) referred for PSG for obstructive sleep apnea diagnoses were prospectively and consecutively recruited. Twenty-five subjects were used for the design study; the validation study was blindly performed on the remaining forty-two subjects. Measurements and Results To train the proposed sound detector, >76,600 acoustic episodes collected in the design study were manually classified by three scorers into snore and non-snore episodes (e.g., bedding noise, coughing, environmental). A feature selection process was applied to select the most discriminative features extracted from time and spectral domains. The average snore/non-snore detection rate (accuracy) for the design group was 98.4% based on a ten-fold cross-validation technique. When tested on the validation group, the average detection rate was 98.2% with sensitivity of 98.0% (snore as a snore) and specificity of 98.3% (noise as noise). Conclusions Audio-based features extracted from time and spectral domains can accurately discriminate between snore and non-snore acoustic events. This audio analysis approach enables detection and analysis of snoring sounds from a full night in order to produce quantified measures for objective follow-up of patients. PMID:24391903

  18. Effects of Stimulus Intensity on Low-Frequency Toneburst Cochlear Microphonic Waveforms

    PubMed Central

    Zhang, Ming

    2013-01-01

    This study investigates changes in amplitude and delays in low-frequency toneburst cochlear microphonic (CM) waveforms recorded at the ear canal in response to different stimulus intensities. Ten volunteers aged 20-30 were recruited. Low-frequency CM waveforms at 500 Hz in response to a 14-ms toneburst were recorded from an ear canal electrode using electrocochleography techniques. The data was statistically analyzed in order to confirm whether the differences were significant in the effects of stimulus intensity on the amplitudes and delays of the low-frequency CM waveforms. Electromagnetic interference artifacts can jeopardize CM measurements but such artifacts can be avoided. The CM waveforms can be recorded at the ear canal in response to a toneburst which is longer than that used in ABR measurements. The CM waveforms thus recorded are robust, and the amplitude of CM waveforms is intensity-dependent. In contrast, the delay of CM waveforms is intensity-independent, which is different from neural responses as their delay or latency is intensity-dependent. These findings may be useful for development of the application of CM measurement as a supplementary approach to otoacoustic emission (OAE) measurement in the clinic which is severely affected by background acoustic noise. The development of the application in the assessment of low-frequency cochlear function may become possible if a further series of studies can verify the feasibility, but it is not meant to be a substitute for audiometry or OAE measurements. The measurement of detection threshold of CM waveform responses using growth function approach may become possible in the clinic. The intensity-independent nature of CMs with regards to delay measurements may also become an impacting factor for differential diagnoses and for designing new research studies. PMID:26557341

  19. Microdischarge arrays

    NASA Astrophysics Data System (ADS)

    Shi, Wenhui

    Microhollow cathode discharges (MHCDs) are DC or pulsed gas discharges between two electrodes, separated by a dielectric, and containing a concentric hole. The diameter of the hole, in this hollow cathode configuration, is in the hundred-micrometer range. MHCDs satisfy the two conditions necessary for an efficient excimer radiation sources: (1) high energy electrons which are required to provide a high concentration of excited or ionized rare gas atoms; (2) high pressure operation which favors excimer formation (a three-body process). Flat panel excimer sources require parallel operation of MHCDs. Based on the current-voltage characteristics of MHCD discharges, which have positive slopes in the low current (Townsend) mode and in the abnormal glow mode, stable arrays of MHCD discharges in argon and xenon could be generated in these current ranges without ballasting each MHCD separately. In the Townsend range, these arrays could be operated up to pressures of 400 Torr. In the abnormal glow mode, discharge arrays were found to be stable up to atmospheric pressure. By using semi-insulating silicon as the anode material, the stable operation of MHCD arrays could be extended to the current range with constant voltage (normal glow) and also that with negative differential conductance (hollow cathode discharge region). Experiments with a cathode geometry without microholes, i.e. excluding the hollow cathode phase, revealed that stable operation of discharges over an extended area were possible. The discharge structure in this configuration reduces to only the cathode fall and negative glow, with the negative glow plasma serving to conduct the discharge current radially to the circular anode. With decreasing current, a transition from homogenous plasma to self-organized plasma filaments is observed. Array formation was not only studied with discharges in parallel, but also with MHCD discharges in series. By using a sandwich electrode configuration, a tandem discharge was

  20. Evaluation of the hearing protector in a real work situation using the field-microphone-in-real-ear method.

    PubMed

    Rocha, Clayton Henrique; Longo, Isadora Altero; Moreira, Renata Rodrigues; Samelli, Alessandra Giannella

    2016-04-01

    Purpose To evaluate the effectiveness of the attenuation of a hearing protector (HP) in a real work situation using the field-microphone-in-real-ear method (f-MIRE). Methods Eighteen individuals of both genders (mean age of 47.17±8 years) participated in this study. In the workplace, the personal attenuation level of the HP was assessed using the f-MIRE method, followed by orientation about the importance of using the HP, cleaning and storing the device, and training for effective placement. Results The analyses showed a significant statistic attenuation for all of the collected data (total noise, by frequency band and dose) when the noise levels in the lapel microphone and the probe microphone were compared. In the comparison of the attenuation values provided by the manufacturer and those found in this study, we observed higher values for the manufacturer in all frequency bands. No difference was observed for the noise levels in the different activities and times evaluated. Conclusion The findings of this study enabled us to know the personal level of attenuation of the HP during a real work situation, which was within the limits of tolerance. It was also possible to collect information about the environmental noise to which these workers are exposed. We noticed situations where this level exceeded the safety values, and therefore it is recommended the use of the HP. It is important that more studies are conducted using the f-MIRE method, because it may be an ally to assess the effectiveness of the HP attenuation in the workplace.

  1. Evaluation of the hearing protector in a real work situation using the field-microphone-in-real-ear method.

    PubMed

    Rocha, Clayton Henrique; Longo, Isadora Altero; Moreira, Renata Rodrigues; Samelli, Alessandra Giannella

    2016-04-01

    Purpose To evaluate the effectiveness of the attenuation of a hearing protector (HP) in a real work situation using the field-microphone-in-real-ear method (f-MIRE). Methods Eighteen individuals of both genders (mean age of 47.17±8 years) participated in this study. In the workplace, the personal attenuation level of the HP was assessed using the f-MIRE method, followed by orientation about the importance of using the HP, cleaning and storing the device, and training for effective placement. Results The analyses showed a significant statistic attenuation for all of the collected data (total noise, by frequency band and dose) when the noise levels in the lapel microphone and the probe microphone were compared. In the comparison of the attenuation values provided by the manufacturer and those found in this study, we observed higher values for the manufacturer in all frequency bands. No difference was observed for the noise levels in the different activities and times evaluated. Conclusion The findings of this study enabled us to know the personal level of attenuation of the HP during a real work situation, which was within the limits of tolerance. It was also possible to collect information about the environmental noise to which these workers are exposed. We noticed situations where this level exceeded the safety values, and therefore it is recommended the use of the HP. It is important that more studies are conducted using the f-MIRE method, because it may be an ally to assess the effectiveness of the HP attenuation in the workplace. PMID:27191871

  2. An Evidence-Based Systematic Review of Directional Microphones and Digital Noise Reduction Hearing Aids in School-Age Children With Hearing Loss

    PubMed Central

    McCreery, Ryan W.; Venediktov, Rebecca A.; Coleman, Jaumeiko J.; Leech, Hillary M.

    2013-01-01

    Purpose The purpose of this evidence-based systematic review was to evaluate the efficacy of digital noise reduction and directional microphones for outcome measures of audibility, speech recognition, speech and language, and self- or parent-report in pediatric hearing aid users. Method The authors searched 26 databases for experimental studies published after 1980 addressing one or more clinical questions and meeting all inclusion criteria. The authors evaluated studies for methodological quality and reported or calculated p values and effect sizes when possible. Results A systematic search of the literature resulted in the inclusion of 4 digital noise reduction and 7 directional microphone studies (in 9 journal articles) that addressed speech recognition, speech and language, and/or self-or parent-report outcomes. No digital noise reduction or directional microphone studies addressed audibility outcomes. Conclusions On the basis of a moderate level of evidence, digital noise reduction was not found to improve or degrade speech understanding. Additional research is needed before conclusions can be drawn regarding the impact of digital noise reduction on important speech, language, hearing, and satisfaction outcomes. Moderate evidence also indicates that directional microphones resulted in improved speech recognition in controlled optimal settings; however, additional research is needed to determine the effectiveness of directional microphones in actual everyday listening environments. PMID:22858614

  3. 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.

  4. Global Arrays

    SciTech Connect

    Krishnamoorthy, Sriram; Daily, Jeffrey A.; Vishnu, Abhinav; Palmer, Bruce J.

    2015-11-01

    Global Arrays (GA) is a distributed-memory programming model that allows for shared-memory-style programming combined with one-sided communication, to create a set of tools that combine high performance with ease-of-use. GA exposes a relatively straightforward programming abstraction, while supporting fully-distributed data structures, locality of reference, and high-performance communication. GA was originally formulated in the early 1990’s to provide a communication layer for the Northwest Chemistry (NWChem) suite of chemistry modeling codes that was being developed concurrently.

  5. 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.

  6. 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. PMID:19406714

  7. 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.

  8. An acoustic-array based structural health monitoring technique for wind turbine blades

    NASA Astrophysics Data System (ADS)

    Aizawa, Kai; Poozesh, Peyman; Niezrecki, Christopher; Baqersad, Javad; Inalpolat, Murat; Heilmann, Gunnar

    2015-04-01

    This paper proposes a non-contact measurement technique for health monitoring of wind turbine blades using acoustic beamforming techniques. The technique works by mounting an audio speaker inside a wind turbine blade and observing the sound radiated from the blade to identify damage within the structure. The main hypothesis for the structural damage detection is that the structural damage (cracks, edge splits, holes etc.) on the surface of a composite wind turbine blade results in changes in the sound radiation characteristics of the structure. Preliminary measurements were carried out on two separate test specimens, namely a composite box and a section of a wind turbine blade to validate the methodology. The rectangular shaped composite box and the turbine blade contained holes with different dimensions and line cracks. An acoustic microphone array with 62 microphones was used to measure the sound radiation from both structures when the speaker was located inside the box and also inside the blade segment. A phased array beamforming technique and CLEAN-based subtraction of point spread function from a reference (CLSPR) were employed to locate the different damage types on both the composite box and the wind turbine blade. The same experiment was repeated by using a commercially available 48-channel acoustic ring array to compare the test results. It was shown that both the acoustic beamforming and the CLSPR techniques can be used to identify the damage in the test structures with sufficiently high fidelity.

  9. 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.

  10. The Australian Square Kilometre Array Pathfinder: Performance of the Boolardy Engineering Test Array

    NASA Astrophysics Data System (ADS)

    McConnell, D.; Allison, J. R.; Bannister, K.; Bell, M. E.; Bignall, H. E.; Chippendale, A. P.; Edwards, P. G.; Harvey-Smith, L.; Hegarty, S.; Heywood, I.; Hotan, A. W.; Indermuehle, B. T.; Lenc, E.; Marvil, J.; Popping, A.; Raja, W.; Reynolds, J. E.; Sault, R. J.; Serra, P.; Voronkov, M. A.; Whiting, M.; Amy, S. W.; Axtens, P.; Ball, L.; Bateman, T. J.; Bock, D. C.-J.; Bolton, R.; Brodrick, D.; Brothers, M.; Brown, A. J.; Bunton, J. D.; Cheng, W.; Cornwell, T.; DeBoer, D.; Feain, I.; Gough, R.; Gupta, N.; Guzman, J. C.; Hampson, G. A.; Hay, S.; Hayman, D. B.; Hoyle, S.; Humphreys, B.; Jacka, C.; Jackson, C. A.; Jackson, S.; Jeganathan, K.; Joseph, J.; Koribalski, B. S.; Leach, M.; Lensson, E. S.; MacLeod, A.; Mackay, S.; Marquarding, M.; McClure-Griffiths, N. M.; Mirtschin, P.; Mitchell, D.; Neuhold, S.; Ng, A.; Norris, R.; Pearce, S.; Qiao, R. Y.; Schinckel, A. E. T.; Shields, M.; Shimwell, T. W.; Storey, M.; Troup, E.; Turner, B.; Tuthill, J.; Tzioumis, A.; Wark, R. M.; Westmeier, T.; Wilson, C.; Wilson, T.

    2016-09-01

    We describe the performance of the Boolardy Engineering Test Array, the prototype for the Australian Square Kilometre Array Pathfinder telescope. Boolardy Engineering Test Array is the first aperture synthesis radio telescope to use phased array feed technology, giving it the ability to electronically form up to nine dual-polarisation beams. We report the methods developed for forming and measuring the beams, and the adaptations that have been made to the traditional calibration and imaging procedures in order to allow BETA to function as a multi-beam aperture synthesis telescope. We describe the commissioning of the instrument and present details of Boolardy Engineering Test Array's performance: sensitivity, beam characteristics, polarimetric properties, and image quality. We summarise the astronomical science that it has produced and draw lessons from operating Boolardy Engineering Test Array that will be relevant to the commissioning and operation of the final Australian Square Kilometre Array Path telescope.

  11. A broadband, capacitive, surface-micromachined, omnidirectional microphone with more than 200 kHz bandwidth.

    PubMed

    Kuntzman, Michael L; Hall, Neal A

    2014-06-01

    A surface micromachined microphone is presented with 230 kHz bandwidth. The structure uses a 2.25 μm thick, 315 μm radius polysilicon diaphragm suspended above an 11 μm gap to form a variable parallel-plate capacitance. The back cavity of the microphone consists of the 11 μm thick air volume immediately behind the moving diaphragm and also an extended lateral cavity with a radius of 504 μm. The dynamic frequency response of the sensor in response to electrostatic signals is presented using laser Doppler vibrometry and indicates a system compliance of 0.4 nm/Pa in the flat-band of the response. The sensor is configured for acoustic signal detection using a charge amplifier, and signal-to-noise ratio measurements and simulations are presented. A resolution of 0.80 mPa/√Hz (32 dB sound pressure level in a 1 Hz bin) is achieved in the flat-band portion of the response extending from 10 kHz to 230 kHz. The proposed sensor design is motivated by defense and intelligence gathering applications that require broadband, airborne signal detection.

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

    NASA Astrophysics Data System (ADS)

    Zumberge, M. A.; Walker, K. T.; Dewolf, S.; Berger, J.; Hedlin, M. A.

    2009-12-01

    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. Spatial separation of sensors with array processing provides only limited signal-to-noise improvement. Mechanical wind filters, like rosette pipe arrays, also help to reduce wind noise, but the rosette infrasound response depends on the apparent speed and frequency of the signal that propagates across the ports as well as the rosette size. This response places an upper limit of about 70 m on the diameter of rosettes; larger rosettes, while better at wind noise reduction, attenuate desirable infrasound signals arriving from all directions. 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. We have previously shown that a single 90 m OFIS (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 and tested computationally efficient beamforming techniques that permit the rapid 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. Recently, several improvements to the instrumentation and methodology have been achieved. We have made an important modification to our interferometric technique that makes the interferometer

  13. Phased-Array Measurements of Single Flow Hot Jets

    NASA Technical Reports Server (NTRS)

    Bridges, James; Lee, Sang Soo

    2005-01-01

    A 16 microphone phased-array system has been successfully applied to measure jet noise source distributions. In this study, a round convergent nozzle was tested at various hot and cold flow conditions: acoustic Mach numbers are between 0.35 and 1.6 and static temperature ratios are varied from cold to 2.7. The classical beamforming method was applied on narrowband frequencies. From the measured source distributions locations of peak strength were tracked and found to be very consistent between adjacent narrowband frequencies. In low speed heated and unheated jets, the peak source locations vary smoothly from the nozzle exit to downstream as the frequency is decreased. When the static temperature ratio was kept constant, the peak source position moved downstream with increasing acoustic Mach number for the Strouhal numbers smaller than about 1.5. It was also noted that the peak source locations of low frequencies occur farther downstream than the end of potential core.

  14. Ordered arrays of near-field optical probes

    NASA Astrophysics Data System (ADS)

    Sojic, Neso; Chovin, Arnaud; Garrigue, Patrick; Manek-Honninger, Inka; Servant, Laurent

    2005-06-01

    Ordered arrays of nanometer-sized optical probes with electrochemiluminescent properties were developed on the distal face of imaging fiber bundles. The fabrication steps are adapted from SNOM probes and nanoelectrodes methodologies and allow to produce high-density arrays of opto-electrochemical probes which retain the initial architecture of the bundle. Apertureless probe arrays and also nanoaperture arrays have thus been prepared. The angular distribution of the far-field intensity transmitted through such nanostructured arrays depends both on their respective architectures and on the characteristic dimensions of the nanoprobes. The subwavelength aperture arrays show a diffracting behavior which is a function of the optical aperture size. The far-field analysis demonstrates their potential application as a parallel near-field optical array in both apertureless and aperture configurations. In addition, each optical nanoaperture is surrounded by a ring-shaped gold nanoelectrode. The electrochemical response of the array is sigmoidal in shape indicating that the nanoelectrodes forming the array are diffusively independent. In other words, each nanoelectrode of the array probes electrochemically a different micro-environment. We show also that the nanoaperture array can be used as an electrochemiluminescent nanosensor array for NADH. Eventually, the arrays keep the imaging properties at both nanometer and micrometer scales. Indeed, each nanoprobe can explore optically a near-field region, whereas the global array allows imaging simultaneously a large micrometric area. This optical array format plays therefore a bridging role by interrelating optical and electrochemiluminescent information obtained concomitantly at the nanometer and micrometer scales.

  15. 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.

  16. Staring arrays - The future lightweight imagers

    NASA Astrophysics Data System (ADS)

    Dennis, P. N. J.; Dann, R. J.

    1985-01-01

    High performance thermal imagers, such as the common modules, are now readily available. These systems generally employ a scanning mechanism to generate the two-dimensional display which makes their adaptation to cheap, lightweight, small imagers difficult. However, with the advent of two-dimensional close packed arrays of infrared detectors the development of such a system is now becoming feasible. A small imager using cadium mercury telluride detectors has been produced commercially. The system has been designed to be adaptable to use both 3-5-micrometer and 8-14-micrometer arrays, and to study various electronic correction mechanisms.

  17. Cochlear microphonic evidence for mechanical propagation of distortion products (f2-f1) and (2f1-f2)

    NASA Astrophysics Data System (ADS)

    Gibian, G. L.

    1980-03-01

    Cochlear microphonic (CM) data were obtained from the second and third turns of the chinchilla cochlea. Fluid-filled glass micropipettes were used to record from scala media, and nichrome wire electrodes were used to record differentially between scala vestibuli and scala tympani. Validity of our results is supported in part by the sensitivity and sharp tuning of the CM and, in the case of the scala media recordings, by the presence of a normal DC endolymphatic potential. It was observed, with sound pressure levels (SPL) as low as 25 dB, that these distortion products in CM display tuning similar to the single-tone response. The tuning similarities observed in the present CM study are consistent with previous neural studies. From these tuning similarities, it is concluded that our CM data reflect the presence of mechanically propagated distortion products at low SPLs.

  18. 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

  19. 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.

  20. Wall-Pressure-Array Measurements Beneath a Separating/Reattaching Flow Region

    NASA Technical Reports Server (NTRS)

    Hudy, Laura M.; Naguib, Ahmed M.; Humphreys, William M., Jr.; Bartram, Scott M.

    2002-01-01

    A database of wall-pressure array measurements was compiled for studying the space-time character of the surface-pressure field within a separating/reattaching flow region. The experimental setup consisted of a long splitter plate instrumented with an array of 80 flush-mounted microphones located within the wake of a fence. Data were acquired for a Reynolds number of 7885, based on the fence height. Two distinctive regions, defined based on their location relative to the position of the mean reattachment point (x(sub r)) of the shear layer, emerged from this investigation. Upstream, from the fence to 1/4x(sub r), the surface-pressure signature was dominated by large time scale disturbances and an upstream convecting velocity of 0.21U(sub infinity). Beyond 1/4x(sub r), turbulent structures with small time scales and a downstream convection velocity of 0.57U(sub infinity) generated most of the pressure fluctuations. There was evidence that these structures began to form around 1/4x(sub r) and grew in strength and size with downstream distance before reattaching on the plate. Only the time-averaged results from the microphones have been examined hitherto and will be presented.

  1. 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.

    2001-05-01

    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.

  2. 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.

  3. Integrated infrared array technology

    NASA Technical Reports Server (NTRS)

    Goebel, J. H.; Mccreight, C. R.

    1986-01-01

    An overview of integrated infrared (IR) array technology is presented. Although the array pixel formats are smaller, and the readout noise of IR arrays is larger, than the corresponding values achieved with optical charge-coupled-device silicon technology, substantial progress is being made in IR technology. Both existing IR arrays and those being developed are described. Examples of astronomical images are given which illustrate the potential of integrated IR arrays for scientific investigations.

  4. Solar array drive system

    NASA Technical Reports Server (NTRS)

    Berkopec, F. D.; Sturman, J. C.; Stanhouse, R. W.

    1976-01-01

    A solar array drive system consisting of a solar array drive mechanism and the corresponding solar array drive electronics is being developed. The principal feature of the solar array drive mechanism is its bidirectional capability which enables its use in mechanical redundancy. The solar array drive system is of a widely applicable design. This configuration will be tested to determine its acceptability for generic mission sets. Foremost of the testing to be performed is the testing for extended duration.

  5. 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.

  6. 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.

  7. 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…

  8. Multipurpose Communication Satellite solar array

    NASA Astrophysics Data System (ADS)

    Bastard, J. L.; Guyot, Ph.

    Through research and development studies, Aerospatiale has developed with the French National Space Agency (CNES) a new concept of a rigid Solar Array (GSR) which covers a range of power between 1 kW and 10 kW EOL 7 yr. This technology is used by Aerospatiale on the currently running programs:—TELECOM 1, Telecommunication satellite, spinned in transfer; —ARABSAT, Telecommunication satellite, 3-axis stabilized in transfer; —TELE-X and TV-SAT/TDF1, Direct Broadcasting satellites, 3-axis stabilized in transfer. This paper deals with the description and the electrical and mechanical performance of the Multipurpose Communication Satellite Solar Array (M.C.S.). For this satellite, which is expected to be launched in 1984, Aerospatiale has selected the following concept: primary power (1.4 kW EOL 7 yr) will be provided by two separate sun-oriented solar array wings equipped with AEG Telefunken BSR (Back Surface Reflector) solar cells. This solar generator is directly derived from the GSR technology. Owing to the three-axis stabilization of the satellite and the partial deployment of the solar array during the transfer phase it has been necessary to use a primary and a secondary hold-down devices. In the same way the position of the shunt on the solar generator induced an increase of the mass, because of the supplementary hold-down point and of the support structure. Now scribing Laser System leads to the optimization of the solar cells dimensions and allows a better fill factor of the panels and so an increase of the power performance. Beside the M.C.S concept which is fully adapted as regards its mission specifications, it was interesting to present different concepts of solar generators optimized not only as regards mass budget but also power budget. These concepts covering a large range of powers are fully adapted to telecommunication missions.

  9. Adaptive Management

    EPA Science Inventory

    Adaptive management is an approach to natural resource management that emphasizes learning through management where knowledge is incomplete, and when, despite inherent uncertainty, managers and policymakers must act. Unlike a traditional trial and error approach, adaptive managem...

  10. High-accuracy fiber optical microphone in a DBR fiber laser based on a nanothick silver diaphragm by self-mixing technique.

    PubMed

    Du, Zhengting; Lu, Liang; Zhang, Wenhua; Yang, Bo; Wu, Shuang; Zhao, Yunhe; Xu, Feng; Wang, Zhiping; Gui, Huaqiao; Liu, Jianguo; Yu, Benli

    2013-12-16

    A high-accuracy fiber optical microphone (FOM) is first applied by self-mixing technique in a DBR fiber laser based on a nanothick silver diaphragm. The nanothick silver diaphragm fabricated by the convenient and low cost electroless plating method is functioned as sensing diaphragm due to critically susceptible to the air vibration. Simultaneously, micro-vibration theory model of self-mixing interference fiber optical microphone is deduced based on quasi-analytical method. The dynamic property to frequencies and amplitudes are experimentally carried out to characterize the fabricated FOM and also the reproduced sound of news and music can clearly meet the ear of the people which shows the technique proposed in this paper guarantee steady, high signal-noise ratio operation and outstanding accuracy in the DBR fiber laser which is potential to medical and security applications such as real-time voice reproduction for throat and voiceprint verification. PMID:24514635

  11. High-accuracy fiber optical microphone in a DBR fiber laser based on a nanothick silver diaphragm by self-mixing technique.

    PubMed

    Du, Zhengting; Lu, Liang; Zhang, Wenhua; Yang, Bo; Wu, Shuang; Zhao, Yunhe; Xu, Feng; Wang, Zhiping; Gui, Huaqiao; Liu, Jianguo; Yu, Benli

    2013-12-16

    A high-accuracy fiber optical microphone (FOM) is first applied by self-mixing technique in a DBR fiber laser based on a nanothick silver diaphragm. The nanothick silver diaphragm fabricated by the convenient and low cost electroless plating method is functioned as sensing diaphragm due to critically susceptible to the air vibration. Simultaneously, micro-vibration theory model of self-mixing interference fiber optical microphone is deduced based on quasi-analytical method. The dynamic property to frequencies and amplitudes are experimentally carried out to characterize the fabricated FOM and also the reproduced sound of news and music can clearly meet the ear of the people which shows the technique proposed in this paper guarantee steady, high signal-noise ratio operation and outstanding accuracy in the DBR fiber laser which is potential to medical and security applications such as real-time voice reproduction for throat and voiceprint verification.

  12. Axiom turkey genotyping array

    Technology Transfer Automated Retrieval System (TEKTRAN)

    The Axiom®Turkey Genotyping Array interrogates 643,845 probesets on the array, covering 643,845 SNPs. The array development was led by Dr. Julie Long of the USDA-ARS Beltsville Agricultural Research Center under a public-private partnership with Hendrix Genetics, Aviagen, and Affymetrix. The Turk...

  13. Response Pattern Based on the Amplitude of Ear Canal Recorded Cochlear Microphonic Waveforms across Acoustic Frequencies in Normal Hearing Subjects

    PubMed Central

    2012-01-01

    Low-frequency otoacoustic emissions (OAEs) are often concealed by acoustic background noise such as those from a patient’s breathing and from the environment during recording in clinics. When using electrocochleaography (ECochG or ECoG), such as cochlear microphonics (CMs), acoustic background noise do not contaminate the recordings. Our objective is to study the response pattern of CM waveforms (CMWs) to explore an alternative approach in assessing cochlear functions. In response to a 14-msec tone burst across several acoustic frequencies, CMWs were recorded at the ear canal from ten normal hearing subjects. A relatively long tone burst has a relatively narrow frequency band. The CMW amplitudes among different frequencies were compared. The CMW amplitudes among different frequencies were compared. Two features were observed in the response pattern of CMWs: the amplitude of CMWs decreased with an increase of stimulus frequency of the tone bursts; and such a decrease occurred at a faster rate at lower frequencies than at higher frequencies. Five factors as potential mechanisms for these features are proposed. Clinical applications such as hearing screening are discussed. Therefore, the response pattern of CMWs suggests that they may be used as an alternative to OAEs in the assessment of cochlear functions in the clinic, especially at low frequencies. PMID:22696071

  14. Convergence of reference frequencies by multiple CF-FM bats (Rhinolophus ferrumequinum nippon) during paired flights evaluated with onboard microphones.

    PubMed

    Furusawa, Yuto; Hiryu, Shizuko; Kobayasi, Kohta I; Riquimaroux, Hiroshi

    2012-09-01

    The constant frequency component of the second harmonic (CF(2)) of echolocation sounds in Rhinolophus ferrumequinum nippon were measured using onboard telemetry microphones while the bats exhibited Doppler-shift compensation during flights with conspecifics. (1) The CF(2) frequency of pulses emitted by individual bats at rest (F (rest)) showed a long-term gradual decline by 0.22 kHz on average over a period of 3 months. The mean neighboring F (rest) (interindividual differences in F (rest) between neighboring bats when the bats were arranged in ascending order according to F (rest)) ranged from 0.08 to 0.11 kHz among 18 bats in a laboratory colony. (2) The standard deviation of observed echo CF(2) (reference frequency) for bats during paired flights ranged from 50 to 90 Hz, which was not significantly different from that during single flights. This finding suggests that during paired flights, bats exhibit Doppler-shift compensation with the same accuracy as when they fly alone. (3) In 60% (n = 29) of the cases, the difference in the reference frequency between two bats during paired flights significantly decreased compared to when the bats flew alone. However, only 15% of the cases (n = 7) showed a significant increase during paired flights. The difference in frequency between two bats did not increase even when the reference frequencies of the individuals were not statistically different during single flights.

  15. Two-microphone spatial filtering provides speech reception benefits for cochlear implant users in difficult acoustic environments

    PubMed Central

    Goldsworthy, Raymond L.; Delhorne, Lorraine A.; Desloge, Joseph G.; Braida, Louis D.

    2014-01-01

    This article introduces and provides an assessment of a spatial-filtering algorithm based on two closely-spaced (∼1 cm) microphones in a behind-the-ear shell. The evaluated spatial-filtering algorithm used fast (∼10 ms) temporal-spectral analysis to determine the location of incoming sounds and to enhance sounds arriving from straight ahead of the listener. Speech reception thresholds (SRTs) were measured for eight cochlear implant (CI) users using consonant and vowel materials under three processing conditions: An omni-directional response, a dipole-directional response, and the spatial-filtering algorithm. The background noise condition used three simultaneous time-reversed speech signals as interferers located at 90°, 180°, and 270°. Results indicated that the spatial-filtering algorithm can provide speech reception benefits of 5.8 to 10.7 dB SRT compared to an omni-directional response in a reverberant room with multiple noise sources. Given the observed SRT benefits, coupled with an efficient design, the proposed algorithm is promising as a CI noise-reduction solution. PMID:25096120

  16. A local active noise control system based on a virtual-microphone technique for railway sleeping vehicle applications

    NASA Astrophysics Data System (ADS)

    Diaz, J.; Egaña, J. M.; Viñolas, J.

    2006-11-01

    Low-frequency broadband noise generated on a railway vehicle by the wheel-rail interaction could be a big annoyance for passengers in sleeping cars. Low-frequency acoustic radiation is extremely difficult to attenuate by using passive devices. In this article, an active noise control (ANC) technique has been proposed for this purpose. A three-dimensional cabin was built in the laboratory to carry out the experiments. The proposed scheme is based on a Filtered-X Least Mean Square (FXLMS) control algorithm, particularised for a virtual-microphone technique. Control algorithms were designed with the Matlab-Simulink tool, and the Real Time Windows Target toolbox of Matlab was used to run in real time the ANC system. Referring to the results, different simulations and experimental performances were analysed to enlarge the silence zone around the passenger's ear zone and along the bed headboard. Attenuations of up to 20 and 15 dB(A) (re:20 μPa) were achieved at the passenger's ear in simulations and in experimental results, respectively.

  17. New Measurement Service for Determining Pressure Sensitivity of Type LS2aP Microphones by the Reciprocity Method

    PubMed Central

    Wagner, Randall P.; Nedzelnitsky, Victor; Fick, Steven E.

    2011-01-01

    A new National Institute of Standards and Technology (NIST) measurement service has been developed for determining the pressure sensitivities of American National Standards Institute and International Electrotechnical Commission type LS2aP laboratory standard microphones over the frequency range 31.5 Hz to 20 000 Hz. At most frequencies common to the new service and the old service, the values of the expanded uncertainties of the new service are one-half the corresponding values of the old service, or better. The new service uses an improved version of the system employed by NIST in the Consultative Committee for Acoustics, Ultrasound, and Vibration (CCAUV) key comparison CCAUV.A-K3. Measurements are performed using a long and a short air-filled plane-wave coupler. For each frequency in the range 31.5 Hz to 2000 Hz, the reported sensitivity level is the average of data from both couplers. For each frequency above 2000 Hz, the reported sensitivity level is determined with data from the short coupler only. For proof test data in the frequency range 31.5 Hz to 2000 Hz, the average absolute differences between data from the long and the short couplers are much smaller than the expanded uncertainties. PMID:26989598

  18. 3D-Printing of inverted pyramid suspending architecture for pyroelectric infrared detectors with inhibited microphonic effect

    NASA Astrophysics Data System (ADS)

    Xu, Qing; Zhao, Xiangyong; Li, Xiaobing; Deng, Hao; Yan, Hong; Yang, Linrong; Di, Wenning; Luo, Haosu; Neumann, Norbert

    2016-05-01

    A sensitive chip with ultralow dielectric loss based on Mn doped PMNT (71/29) has been proposed for high-end pyroelectric devices. The dielectric loss at 1 kHz is 0.005%, one order lower than the minimum value reported so far. The detective figure of merit (Fd) is up to 92.6 × 10-5 Pa-1/2 at 1 kHz and 53.5 × 10-5 Pa-1/2 at 10 Hz, respectively. In addition, an inverted pyramid suspending architecture for supporting the sensitive chip has been designed and manufactured by 3D printing technology. The combination of this sensitive chip and the proposed suspending architecture largely enhances the performance of the pyroelectric detectors. The responsivity and specific detectivity are 669,811 V/W and 3.32 × 109 cm Hz1/2/W at 10 Hz, respectively, 1.9 times and 1.5 times higher than those of the highest values in literature. Furthermore, the microphonic effect can be largely inhibited according to the theoretical and experimental analysis. This architecture will have promising applications in high-end and stable pyroelectric infrared detectors.

  19. Four-channel HF receiving antenna array simulator

    NASA Astrophysics Data System (ADS)

    Owens, K. C.

    1993-08-01

    This document discusses the design, fabrication, and operation of an antenna array simulator, which is an integral part of the NRaD adaptive array evaluation facility. A brief introduction is provided to explain how adaptive receiving antenna arrays can improve Navy high-frequency communications and how this developmental work is supported at NRaD. The main body of the text encompasses both the software algorithms created to support computerized remote control and antenna phase-tuning operation and a detailed hardware description of the Simulator circuitry, which is suitable as a guide for maintenance and troubleshooting tasks.

  20. Fireplace adapters

    SciTech Connect

    Hunt, R.L.

    1983-12-27

    An adapter is disclosed for use with a fireplace. The stove pipe of a stove standing in a room to be heated may be connected to the flue of the chimney so that products of combustion from the stove may be safely exhausted through the flue and outwardly of the chimney. The adapter may be easily installed within the fireplace by removing the damper plate and fitting the adapter to the damper frame. Each of a pair of bolts has a portion which hooks over a portion of the damper frame and a threaded end depending from the hook portion and extending through a hole in the adapter. Nuts are threaded on the bolts and are adapted to force the adapter into a tight fit with the adapter frame.

  1. Thermophotovoltaic Array Optimization

    SciTech Connect

    SBurger; E Brown; K Rahner; L Danielson; J Openlander; J Vell; D Siganporia

    2004-07-29

    A systematic approach to thermophotovoltaic (TPV) array design and fabrication was used to optimize the performance of a 192-cell TPV array. The systematic approach began with cell selection criteria that ranked cells and then matched cell characteristics to maximize power output. Following cell selection, optimization continued with an array packaging design and fabrication techniques that introduced negligible electrical interconnect resistance and minimal parasitic losses while maintaining original cell electrical performance. This paper describes the cell selection and packaging aspects of array optimization as applied to fabrication of a 192-cell array.

  2. 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.

  3. 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

  4. Adaptive SPECT

    PubMed Central

    Barrett, Harrison H.; Furenlid, Lars R.; Freed, Melanie; Hesterman, Jacob Y.; Kupinski, Matthew A.; Clarkson, Eric; Whitaker, Meredith K.

    2008-01-01

    Adaptive imaging systems alter their data-acquisition configuration or protocol in response to the image information received. An adaptive pinhole single-photon emission computed tomography (SPECT) system might acquire an initial scout image to obtain preliminary information about the radiotracer distribution and then adjust the configuration or sizes of the pinholes, the magnifications, or the projection angles in order to improve performance. This paper briefly describes two small-animal SPECT systems that allow this flexibility and then presents a framework for evaluating adaptive systems in general, and adaptive SPECT systems in particular. The evaluation is in terms of the performance of linear observers on detection or estimation tasks. Expressions are derived for the ideal linear (Hotelling) observer and the ideal linear (Wiener) estimator with adaptive imaging. Detailed expressions for the performance figures of merit are given, and possible adaptation rules are discussed. PMID:18541485

  5. Adaptive passive fathometer processing.

    PubMed

    Siderius, Martin; Song, Heechun; Gerstoft, Peter; Hodgkiss, William S; Hursky, Paul; Harrison, Chris

    2010-04-01

    Recently, a technique has been developed to image seabed layers using the ocean ambient noise field as the sound source. This so called passive fathometer technique exploits the naturally occurring acoustic sounds generated on the sea-surface, primarily from breaking waves. The method is based on the cross-correlation of noise from the ocean surface with its echo from the seabed, which recovers travel times to significant seabed reflectors. To limit averaging time and make this practical, beamforming is used with a vertical array of hydrophones to reduce interference from horizontally propagating noise. The initial development used conventional beamforming, but significant improvements have been realized using adaptive techniques. In this paper, adaptive methods for this process are described and applied to several data sets to demonstrate improvements possible as compared to conventional processing.

  6. Generation of remote adaptive torsional shear waves with an octagonal phased array to enhance displacements and reduce variability of shear wave speeds: comparison with quasi-plane shear wavefronts.

    PubMed

    Ouared, Abderrahmane; Montagnon, Emmanuel; Cloutier, Guy

    2015-10-21

    A method based on adaptive torsional shear waves (ATSW) is proposed to overcome the strong attenuation of shear waves generated by a radiation force in dynamic elastography. During the inward propagation of ATSW, the magnitude of displacements is enhanced due to the convergence of shear waves and constructive interferences. The proposed method consists in generating ATSW fields from the combination of quasi-plane shear wavefronts by considering a linear superposition of displacement maps. Adaptive torsional shear waves were experimentally generated in homogeneous and heterogeneous tissue mimicking phantoms, and compared to quasi-plane shear wave propagations. Results demonstrated that displacement magnitudes by ATSW could be up to 3 times higher than those obtained with quasi-plane shear waves, that the variability of shear wave speeds was reduced, and that the signal-to-noise ratio of displacements was improved. It was also observed that ATSW could cause mechanical inclusions to resonate in heterogeneous phantoms, which further increased the displacement contrast between the inclusion and the surrounding medium. This method opens a way for the development of new noninvasive tissue characterization strategies based on ATSW in the framework of our previously reported shear wave induced resonance elastography (SWIRE) method proposed for breast cancer diagnosis.

  7. Generation of remote adaptive torsional shear waves with an octagonal phased array to enhance displacements and reduce variability of shear wave speeds: comparison with quasi-plane shear wavefronts

    NASA Astrophysics Data System (ADS)

    Ouared, Abderrahmane; Montagnon, Emmanuel; Cloutier, Guy

    2015-10-01

    A method based on adaptive torsional shear waves (ATSW) is proposed to overcome the strong attenuation of shear waves generated by a radiation force in dynamic elastography. During the inward propagation of ATSW, the magnitude of displacements is enhanced due to the convergence of shear waves and constructive interferences. The proposed method consists in generating ATSW fields from the combination of quasi-plane shear wavefronts by considering a linear superposition of displacement maps. Adaptive torsional shear waves were experimentally generated in homogeneous and heterogeneous tissue mimicking phantoms, and compared to quasi-plane shear wave propagations. Results demonstrated that displacement magnitudes by ATSW could be up to 3 times higher than those obtained with quasi-plane shear waves, that the variability of shear wave speeds was reduced, and that the signal-to-noise ratio of displacements was improved. It was also observed that ATSW could cause mechanical inclusions to resonate in heterogeneous phantoms, which further increased the displacement contrast between the inclusion and the surrounding medium. This method opens a way for the development of new noninvasive tissue characterization strategies based on ATSW in the framework of our previously reported shear wave induced resonance elastography (SWIRE) method proposed for breast cancer diagnosis.

  8. Dynamic Associations in Nonlinear Computing Arrays

    NASA Astrophysics Data System (ADS)

    Huberman, B. A.; Hogg, T.

    1985-10-01

    We experimentally show that nonlinear parallel arrays can be made to compute with attractors. This leads to fast adaptive behavior in which dynamical associations can be made between different inputs which initially produce sharply distinct outputs. We first define a set of simple local procedures which allow a general computing structure to change its state in time so as to produce classical Pavlovian conditioning. We then examine the dynamics of coalescence and dissociation of attractors with a number of quantitative experiments. We also show how such arrays exhibit generalization and differentiation of inputs in their behavior.

  9. Adaptive Computing.

    ERIC Educational Resources Information Center

    Harrell, William

    1999-01-01

    Provides information on various adaptive technology resources available to people with disabilities. (Contains 19 references, an annotated list of 129 websites, and 12 additional print resources.) (JOW)

  10. Contour adaptation.

    PubMed

    Anstis, Stuart

    2013-01-01

    It is known that adaptation to a disk that flickers between black and white at 3-8 Hz on a gray surround renders invisible a congruent gray test disk viewed afterwards. This is contrast adaptation. We now report that adapting simply to the flickering circular outline of the disk can have the same effect. We call this "contour adaptation." This adaptation does not transfer interocularly, and apparently applies only to luminance, not color. One can adapt selectively to only some of the contours in a display, making only these contours temporarily invisible. For instance, a plaid comprises a vertical grating superimposed on a horizontal grating. If one first adapts to appropriate flickering vertical lines, the vertical components of the plaid disappears and it looks like a horizontal grating. Also, we simulated a Cornsweet (1970) edge, and we selectively adapted out the subjective and objective contours of a Kanisza (1976) subjective square. By temporarily removing edges, contour adaptation offers a new technique to study the role of visual edges, and it demonstrates how brightness information is concentrated in edges and propagates from them as it fills in surfaces.

  11. Adaptive multibeam antennas for spacelab. Phase A: Feasibility study

    NASA Technical Reports Server (NTRS)

    Allen, C. C.; Applebaum, S. P.; Popowsky, W. J.; Wouch, G.

    1976-01-01

    The feasibility was studied of using adaptive multibeam multi-frequency antennas on the spacelab, and to define the experiment configuration and program plan needed for a demonstration to prove the concept. Three applications missions were selected, and requirements were defined for an L band communications experiment, an L band radiometer experiment, and a Ku band communications experiment. Reflector, passive lens, and phased array antenna systems were considered, and the Adaptive Multibeam Phased Array (AMPA) was chosen. Array configuration and beamforming network tradeoffs resulted in a single 3m x 3m L band array with 576 elements for high radiometer beam efficiency. Separate 0.4m x 0.4 m arrays are used to transmit and receive at Ku band with either 576 elements or thinned apertures. Each array has two independently steerable 5 deg beams, which are adaptively controlled.

  12. Superconducting Bolometer Array Architectures

    NASA Technical Reports Server (NTRS)

    Benford, Dominic; Chervenak, Jay; Irwin, Kent; Moseley, S. Harvey; Shafer, Rick; Staguhn, Johannes; Wollack, Ed; Oegerle, William (Technical Monitor)

    2002-01-01

    The next generation of far-infrared and submillimeter instruments require large arrays of detectors containing thousands of elements. These arrays will necessarily be multiplexed, and superconducting bolometer arrays are the most promising present prospect for these detectors. We discuss our current research into superconducting bolometer array technologies, which has recently resulted in the first multiplexed detections of submillimeter light and the first multiplexed astronomical observations. Prototype arrays containing 512 pixels are in production using the Pop-Up Detector (PUD) architecture, which can be extended easily to 1000 pixel arrays. Planar arrays of close-packed bolometers are being developed for the GBT (Green Bank Telescope) and for future space missions. For certain applications, such as a slewed far-infrared sky survey, feedhorncoupling of a large sparsely-filled array of bolometers is desirable, and is being developed using photolithographic feedhorn arrays. Individual detectors have achieved a Noise Equivalent Power (NEP) of -10(exp 17) W/square root of Hz at 300mK, but several orders of magnitude improvement are required and can be reached with existing technology. The testing of such ultralow-background detectors will prove difficult, as this requires optical loading of below IfW. Antenna-coupled bolometer designs have advantages for large format array designs at low powers due to their mode selectivity.

  13. Electronic Switch Arrays for Managing Microbattery Arrays

    NASA Technical Reports Server (NTRS)

    Mojarradi, Mohammad; Alahmad, Mahmoud; Sukumar, Vinesh; Zghoul, Fadi; Buck, Kevin; Hess, Herbert; Li, Harry; Cox, David

    2008-01-01

    Integrated circuits have been invented for managing the charging and discharging of such advanced miniature energy-storage devices as planar arrays of microscopic energy-storage elements [typically, microscopic electrochemical cells (microbatteries) or microcapacitors]. The architecture of these circuits enables implementation of the following energy-management options: dynamic configuration of the elements of an array into a series or parallel combination of banks (subarrarys), each array comprising a series of parallel combination of elements; direct addressing of individual banks for charging/or discharging; and, disconnection of defective elements and corresponding reconfiguration of the rest of the array to utilize the remaining functional elements to obtain the desited voltage and current performance. An integrated circuit according to the invention consists partly of a planar array of field-effect transistors that function as switches for routing electric power among the energy-storage elements, the power source, and the load. To connect the energy-storage elements to the power source for charging, a specific subset of switches is closed; to connect the energy-storage elements to the load for discharging, a different specific set of switches is closed. Also included in the integrated circuit is circuitry for monitoring and controlling charging and discharging. The control and monitoring circuitry, the switching transistors, and interconnecting metal lines are laid out on the integrated-circuit chip in a pattern that registers with the array of energy-storage elements. There is a design option to either (1) fabricate the energy-storage elements in the corresponding locations on, and as an integral part of, this integrated circuit; or (2) following a flip-chip approach, fabricate the array of energy-storage elements on a separate integrated-circuit chip and then align and bond the two chips together.

  14. 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.

  15. An introduction to coil array design for parallel MRI.

    PubMed

    Ohliger, Michael A; Sodickson, Daniel K

    2006-05-01

    The basic principles of radiofrequency coil array design for parallel MRI are described from both theoretical and practical perspectives. Because parallel MRI techniques rely on coil array sensitivities to provide spatial information about the sample, a careful choice of array design is essential. The concepts of coil array spatial encoding are first discussed from four qualitative perspectives. These qualitative descriptions include using coil arrays to emulate spatial harmonics, choosing coils with selective sensitivities to aliased pixels, using coil sensitivities with broad k-space reception profiles, and relying on detector coils to provide a set of generalized projections of the sample. This qualitative discussion is followed by a quantitative analysis of coil arrays, which is discussed in terms of the baseline SNR of the received images as well as the noise amplifications (g-factor) in the reconstructed data. The complications encountered during the experimental evaluation of coil array SNR are discussed, and solutions are proposed. A series of specific array designs are reviewed, with an emphasis on the general design considerations that motivate each approach. Finally, a set of special topics is discussed, which reflect issues that have become important, especially as arrays are being designed for more high-performance applications of parallel MRI. These topics include concerns about the depth penetration of arrays composed of small elements, the use of adaptive arrays for systems with limited receiver channels, the management of inductive coupling between array elements, and special considerations required at high field strengths. The fundamental limits of spatial encoding using coil arrays are discussed, with a primary emphasis on how the determination of these limits impacts the design of optimized arrays. This review is intended to provide insight into how arrays are currently used for parallel MRI and to place into context the new innovations that are

  16. Designing linear systolic arrays

    SciTech Connect

    Kumar, V.K.P.; Tsai, Y.C. . Dept. of Electrical Engineering)

    1989-12-01

    The authors develop a simple mapping technique to design linear systolic arrays. The basic idea of the technique is to map the computations of a certain class of two-dimensional systolic arrays onto one-dimensional arrays. Using this technique, systolic algorithms are derived for problems such as matrix multiplication and transitive closure on linearly connected arrays of PEs with constant I/O bandwidth. Compared to known designs in the literature, the technique leads to modular systolic arrays with constant hardware in each PE, few control lines, lexicographic data input/output, and improved delay time. The unidirectional flow of control and data in this design assures implementation of the linear array in the known fault models of wafer scale integration.

  17. Carbon nanotube nanoelectrode arrays

    DOEpatents

    Ren, Zhifeng; Lin, Yuehe; Yantasee, Wassana; Liu, Guodong; Lu, Fang; Tu, Yi

    2008-11-18

    The present invention relates to microelectode arrays (MEAs), and more particularly to carbon nanotube nanoelectrode arrays (CNT-NEAs) for chemical and biological sensing, and methods of use. A nanoelectrode array includes a carbon nanotube material comprising an array of substantially linear carbon nanotubes each having a proximal end and a distal end, the proximal end of the carbon nanotubes are attached to a catalyst substrate material so as to form the array with a pre-determined site density, wherein the carbon nanotubes are aligned with respect to one another within the array; an electrically insulating layer on the surface of the carbon nanotube material, whereby the distal end of the carbon nanotubes extend beyond the electrically insulating layer; a second adhesive electrically insulating layer on the surface of the electrically insulating layer, whereby the distal end of the carbon nanotubes extend beyond the second adhesive electrically insulating layer; and a metal wire attached to the catalyst substrate material.

  18. Climate adaptation

    NASA Astrophysics Data System (ADS)

    Kinzig, Ann P.

    2015-03-01

    This paper is intended as a brief introduction to climate adaptation in a conference devoted otherwise to the physics of sustainable energy. Whereas mitigation involves measures to reduce the probability of a potential event, such as climate change, adaptation refers to actions that lessen the impact of climate change. Mitigation and adaptation differ in other ways as well. Adaptation does not necessarily have to be implemented immediately to be effective; it only needs to be in place before the threat arrives. Also, adaptation does not necessarily require global, coordinated action; many effective adaptation actions can be local. Some urban communities, because of land-use change and the urban heat-island effect, currently face changes similar to some expected under climate change, such as changes in water availability, heat-related morbidity, or changes in disease patterns. Concern over those impacts might motivate the implementation of measures that would also help in climate adaptation, despite skepticism among some policy makers about anthropogenic global warming. Studies of ancient civilizations in the southwestern US lends some insight into factors that may or may not be important to successful adaptation.

  19. Pacific Array (Transportable Broadband Ocean Floor Array)

    NASA Astrophysics Data System (ADS)

    Kawakatsu, Hitoshi; Ekstrom, Goran; Evans, Rob; Forsyth, Don; Gaherty, Jim; Kennett, Brian; Montagner, Jean-Paul; Utada, Hisashi

    2016-04-01

    Based on recent developments on broadband ocean bottom seismometry, we propose a next generation large-scale array experiment in the ocean. Recent advances in ocean bottom broadband seismometry1, together with advances in the seismic analysis methodology, have enabled us to resolve the regional 1-D structure of the entire lithosphere/asthenosphere system, including seismic anisotropy (azimuthal, and hopefully radial), with deployments of ~15 broadband ocean bottom seismometers (BBOBSs). Having ~15 BBOBSs as an array unit for a 2-year deployment, and repeating such deployments in a leap-frog way or concurrently (an array of arrays) for a decade or so would enable us to cover a large portion of the Pacific basin. Such efforts, not only by giving regional constraints on the 1-D structure beneath Pacific ocean, but also by sharing waveform data for global scale waveform tomography, would drastically increase our knowledge of how plate tectonics works on this planet, as well as how it worked for the past 150 million years. International collaborations is essential: if three countries/institutions participate this endeavor together, Pacific Array may be accomplished within five-or-so years.

  20. Phased-array radars

    NASA Astrophysics Data System (ADS)

    Brookner, E.

    1985-02-01

    The operating principles, technology, and applications of phased-array radars are reviewed and illustrated with diagrams and photographs. Consideration is given to the antenna elements, circuitry for time delays, phase shifters, pulse coding and compression, and hybrid radars combining phased arrays with lenses to alter the beam characteristics. The capabilities and typical hardware of phased arrays are shown using the US military systems COBRA DANE and PAVE PAWS as examples.

  1. Integrated avalanche photodiode arrays

    DOEpatents

    Harmon, Eric S.

    2015-07-07

    The present disclosure includes devices for detecting photons, including avalanche photon detectors, arrays of such detectors, and circuits including such arrays. In some aspects, the detectors and arrays include a virtual beveled edge mesa structure surrounded by resistive material damaged by ion implantation and having side wall profiles that taper inwardly towards the top of the mesa structures, or towards the direction from which the ion implantation occurred. Other aspects are directed to masking and multiple implantation and/or annealing steps. Furthermore, methods for fabricating and using such devices, circuits and arrays are disclosed.

  2. Focal plane array with modular pixel array components for scalability

    DOEpatents

    Kay, Randolph R; Campbell, David V; Shinde, Subhash L; Rienstra, Jeffrey L; Serkland, Darwin K; Holmes, Michael L

    2014-12-09

    A modular, scalable focal plane array is provided as an array of integrated circuit dice, wherein each die includes a given amount of modular pixel array circuitry. The array of dice effectively multiplies the amount of modular pixel array circuitry to produce a larger pixel array without increasing die size. Desired pixel pitch across the enlarged pixel array is preserved by forming die stacks with each pixel array circuitry die stacked on a separate die that contains the corresponding signal processing circuitry. Techniques for die stack interconnections and die stack placement are implemented to ensure that the desired pixel pitch is preserved across the enlarged pixel array.

  3. Reverse Phase Protein Arrays for Compound Profiling.

    PubMed

    Moerke, Nathan; Fallahi-Sichani, Mohammad

    2016-01-01

    Reverse phase protein arrays (RPPAs), also called reverse phase lysate arrays (RPLAs), involve immobilizing cell or tissue lysates, in small spots, onto solid supports which are then probed with primary antibodies specific for proteins or post-translational modifications of interest. RPPA assays are well suited for large-scale, high-throughput measurement of protein and PTM levels in cells and tissues. RPPAs are affordable and highly multiplexable, as a large number of arrays can readily be produced in parallel and then probed separately with distinct primary antibodies. This article describes a procedure for treating cells and preparing cell lysates, as well as a procedure for generating RPPAs using these lysates. A method for probing, imaging, and analyzing RPPAs is also described. These procedures are readily adaptable to a wide range of studies of cell signaling in response to drugs and other perturbations. © 2016 by John Wiley & Sons, Inc. PMID:27622568

  4. Relating hearing loss and executive functions to hearing aid users' preference for, and speech recognition with, different combinations of binaural noise reduction and microphone directionality

    PubMed Central

    Neher, Tobias

    2014-01-01

    Knowledge of how executive functions relate to preferred hearing aid (HA) processing is sparse and seemingly inconsistent with related knowledge for speech recognition outcomes. This study thus aimed to find out if (1) performance on a measure of reading span (RS) is related to preferred binaural noise reduction (NR) strength, (2) similar relations exist for two different, non-verbal measures of executive function, (3) pure-tone average hearing loss (PTA), signal-to-noise ratio (SNR), and microphone directionality (DIR) also influence preferred NR strength, and (4) preference and speech recognition outcomes are similar. Sixty elderly HA users took part. Six HA conditions consisting of omnidirectional or cardioid microphones followed by inactive, moderate, or strong binaural NR as well as linear amplification were tested. Outcome was assessed at fixed SNRs using headphone simulations of a frontal target talker in a busy cafeteria. Analyses showed positive effects of active NR and DIR on preference, and negative and positive effects of, respectively, strong NR and DIR on speech recognition. Also, while moderate NR was the most preferred NR setting overall, preference for strong NR increased with SNR. No relation between RS and preference was found. However, larger PTA was related to weaker preference for inactive NR and stronger preference for strong NR for both microphone modes. Equivalent (but weaker) relations between worse performance on one non-verbal measure of executive function and the HA conditions without DIR were found. For speech recognition, there were relations between HA condition, PTA, and RS, but their pattern differed from that for preference. Altogether, these results indicate that, while moderate NR works well in general, a notable proportion of HA users prefer stronger NR. Furthermore, PTA and executive functions can account for some of the variability in preference for, and speech recognition with, different binaural NR and DIR settings. PMID

  5. Predicting transient upset in gate arrays

    SciTech Connect

    Woodruff, R.L.; Nelson, D.A.; Scherr, S.

    1987-12-01

    A simulation program for predicting dose rate upset has been adapted from the Power Analysis for Integrated Circuits program (PANIC). The program provides detailed analysis on the V/sub CC/V/sub SS/ difference at any location within the array as well as the amount of photocurrent being collected, as a function of design. The simulation has been compared to experiment for a specific design and was found to correlate to within 20% at 5 volts.

  6. 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.

  7. Extension of DAMAS Phased Array Processing for Spatial Coherence Determination (DAMAS-C)

    NASA Technical Reports Server (NTRS)

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

    2006-01-01

    The present study reports a new development of the DAMAS microphone phased array processing methodology that allows the determination and separation of coherent and incoherent noise source distributions. In 2004, a Deconvolution Approach for the Mapping of Acoustic Sources (DAMAS) was developed which decoupled the array design and processing influence from the noise being measured, using a simple and robust algorithm. In 2005, three-dimensional applications of DAMAS were examined. DAMAS has been shown to render an unambiguous quantitative determination of acoustic source position and strength. However, an underlying premise of DAMAS, as well as that of classical array beamforming methodology, is that the noise regions under study are distributions of statistically independent sources. The present development, called DAMAS-C, extends the basic approach to include coherence definition between noise sources. The solutions incorporate cross-beamforming array measurements over the survey region. While the resulting inverse problem can be large and the iteration solution computationally demanding, it solves problems no other technique can approach. DAMAS-C is validated using noise source simulations and is applied to airframe flap noise test results.

  8. Observing Infrasound and Atmospheric Pressure with the NSF EarthScope USArray Transportable Array

    NASA Astrophysics Data System (ADS)

    Vernon, F. L.; Hedlin, M. A.; Busby, R. W.; Woodward, R.

    2010-12-01

    We are creating a real-time infrasound array whose sensing elements are co-located with the 400 seismic stations in the USArray Transportable Array component of the NSF EarthScope program. This continuously sampled array, of an unprecedented scale, will provide opportunities for groundbreaking and interdisciplinary research in atmospheric acoustics, atmospheric science, and seismology. Such an array will sample mean (absolute) values and fluctuations of the surface air pressure with nominal 70 km station spacing, with a dynamic range of about 7 orders of magnitude, and with a sampling frequency of up to 40 Hz. All samples will be synchronized to UTC. This dense network of infrasound sensors will permit us to study the nature of long-range infrasound propagation from regional to continental distances, and study the sources of infrasound signals, using actual acoustic data, free of concerns about seismic-to-acoustic coupling. All new TA stations deployed starting this fall will have a Quanterra Environmental Processor with internal VTI SCP1000 MEMS barometric pressure gauge, Setra 278 absolute microbarometer, and NCPA Infrasound Microphone. We will present data from field tests and from the newly deployed instrumentation.

  9. 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. PMID:25786949

  10. Active microstructured x-ray optical arrays

    NASA Astrophysics Data System (ADS)

    Michette, Alan G.; Pfauntsch, Slawka J.; Sahraei, Shahin; Shand, Matthew; Morrison, Graeme R.; Hart, David; Vojnovic, Boris; Stevenson, Tom; Parkes, William; Dunare, Camelia; Willingale, Richard; Feldman, Charlotte H.; Button, Tim W.; Zhang, Dou; Rodriguez-Sanmartin, Daniel; Wang, Hongchang; Smith, Andy D.

    2009-05-01

    The UK Smart X-Ray Optics consortium is developing novel reflective adaptive/active x-ray optics for small-scale laboratory applications, including studies of radiation-induced damage to biological material. The optics work on the same principle as polycapillaries, using configured arrays of channels etched into thin silicon, such that each x-ray photon reflects at most once off a channel wall. Using two arrays in succession provides two reflections and thus the Abbe sine condition can be approximately satisfied, reducing aberrations. Adaptivity is achieved by flexing one or both arrays using piezo actuation, which can provide further reduction of aberrations as well as controllable focal lengths. Modelling of such arrays for used on an x-ray microprobe, based on a microfocus source with an emitting region approximately 1μm in diameter, shows that a focused flux approximately two orders of magnitude greater than possible with a zone plate of comparable focal length is possible, assuming that the channel wall roughness is less than about 2nm.

  11. Solar array deployment mechanism

    NASA Astrophysics Data System (ADS)

    Calassa, Mark C.; Kackley, Russell

    1995-05-01

    This paper describes a Solar Array Deployment Mechanism (SADM) used to deploy a rigid solar array panel on a commercial spacecraft. The application required a deployment mechanism design that was not only lightweight, but also could be produced and installed at the lowest possible cost. This paper covers design, test, and analysis of a mechanism that meets these requirements.

  12. Solar array deployment mechanism

    NASA Technical Reports Server (NTRS)

    Calassa, Mark C.; Kackley, Russell

    1995-01-01

    This paper describes a Solar Array Deployment Mechanism (SADM) used to deploy a rigid solar array panel on a commercial spacecraft. The application required a deployment mechanism design that was not only lightweight, but also could be produced and installed at the lowest possible cost. This paper covers design, test, and analysis of a mechanism that meets these requirements.

  13. Array for detecting microbes

    DOEpatents

    Andersen, Gary L.; DeSantis, Todd D.

    2014-07-08

    The present embodiments relate to an array system for detecting and identifying biomolecules and organisms. More specifically, the present embodiments relate to an array system comprising a microarray configured to simultaneously detect a plurality of organisms in a sample at a high confidence level.

  14. ISS Solar Array Management

    NASA Technical Reports Server (NTRS)

    Williams, James P.; Martin, Keith D.; Thomas, Justin R.; Caro, Samuel

    2010-01-01

    The International Space Station (ISS) Solar Array Management (SAM) software toolset provides the capabilities necessary to operate a spacecraft with complex solar array constraints. It monitors spacecraft telemetry and provides interpretations of solar array constraint data in an intuitive manner. The toolset provides extensive situational awareness to ensure mission success by analyzing power generation needs, array motion constraints, and structural loading situations. The software suite consists of several components including samCS (constraint set selector), samShadyTimers (array shadowing timers), samWin (visualization GUI), samLock (array motion constraint computation), and samJet (attitude control system configuration selector). It provides high availability and uptime for extended and continuous mission support. It is able to support two-degrees-of-freedom (DOF) array positioning and supports up to ten simultaneous constraints with intuitive 1D and 2D decision support visualizations of constraint data. Display synchronization is enabled across a networked control center and multiple methods for constraint data interpolation are supported. Use of this software toolset increases flight safety, reduces mission support effort, optimizes solar array operation for achieving mission goals, and has run for weeks at a time without issues. The SAM toolset is currently used in ISS real-time mission operations.

  15. Computation of optimized arrays for 3-D electrical imaging surveys

    NASA Astrophysics Data System (ADS)

    Loke, M. H.; Wilkinson, P. B.; Uhlemann, S. S.; Chambers, J. E.; Oxby, L. S.

    2014-12-01

    3-D electrical resistivity surveys and inversion models are required to accurately resolve structures in areas with very complex geology where 2-D models might suffer from artefacts. Many 3-D surveys use a grid where the number of electrodes along one direction (x) is much greater than in the perpendicular direction (y). Frequently, due to limitations in the number of independent electrodes in the multi-electrode system, the surveys use a roll-along system with a small number of parallel survey lines aligned along the x-direction. The `Compare R' array optimization method previously used for 2-D surveys is adapted for such 3-D surveys. Offset versions of the inline arrays used in 2-D surveys are included in the number of possible arrays (the comprehensive data set) to improve the sensitivity to structures in between the lines. The array geometric factor and its relative error are used to filter out potentially unstable arrays in the construction of the comprehensive data set. Comparisons of the conventional (consisting of dipole-dipole and Wenner-Schlumberger arrays) and optimized arrays are made using a synthetic model and experimental measurements in a tank. The tests show that structures located between the lines are better resolved with the optimized arrays. The optimized arrays also have significantly better depth resolution compared to the conventional arrays.

  16. Toothbrush Adaptations.

    ERIC Educational Resources Information Center

    Exceptional Parent, 1987

    1987-01-01

    Suggestions are presented for helping disabled individuals learn to use or adapt toothbrushes for proper dental care. A directory lists dental health instructional materials available from various organizations. (CB)

  17. Microfabricated ion trap array

    DOEpatents

    Blain, Matthew G.; Fleming, James G.

    2006-12-26

    A microfabricated ion trap array, comprising a plurality of ion traps having an inner radius of order one micron, can be fabricated using surface micromachining techniques and materials known to the integrated circuits manufacturing and microelectromechanical systems industries. Micromachining methods enable batch fabrication, reduced manufacturing costs, dimensional and positional precision, and monolithic integration of massive arrays of ion traps with microscale ion generation and detection devices. Massive arraying enables the microscale ion traps to retain the resolution, sensitivity, and mass range advantages necessary for high chemical selectivity. The reduced electrode voltage enables integration of the microfabricated ion trap array with on-chip circuit-based rf operation and detection electronics (i.e., cell phone electronics). Therefore, the full performance advantages of the microfabricated ion trap array can be realized in truly field portable, handheld microanalysis systems.

  18. Photovoltaic array loss mechanisms

    NASA Technical Reports Server (NTRS)

    Gonzalez, Charles

    1986-01-01

    Loss mechanisms which come into play when solar cell modules are mounted in arrays are identified. Losses can occur either from a reduction in the array electrical performance or with nonoptimal extraction of power from the array. Electrical performance degradation is caused by electrical mismatch, transmission losses from cell surface soiling and steep angle of reflectance, and electrical losses from field wiring resistance and the voltage drop across blocking diodes. The second type of loss, concerned with the operating points of the array, can involve nonoptimal load impedance and limiting the operating envelope of the array to specific ranges of voltage and current. Each of the loss mechanisms are discussed and average energy losses expected from soiling, steep reflectance angles and circuit losses are calculated.

  19. High density pixel array

    NASA Technical Reports Server (NTRS)

    Wiener-Avnear, Eliezer (Inventor); McFall, James Earl (Inventor)

    2004-01-01

    A pixel array device is fabricated by a laser micro-milling method under strict process control conditions. The device has an array of pixels bonded together with an adhesive filling the grooves between adjacent pixels. The array is fabricated by moving a substrate relative to a laser beam of predetermined intensity at a controlled, constant velocity along a predetermined path defining a set of grooves between adjacent pixels so that a predetermined laser flux per unit area is applied to the material, and repeating the movement for a plurality of passes of the laser beam until the grooves are ablated to a desired depth. The substrate is of an ultrasonic transducer material in one example for fabrication of a 2D ultrasonic phase array transducer. A substrate of phosphor material is used to fabricate an X-ray focal plane array detector.

  20. Multibeam Phased Array Antennas

    NASA Technical Reports Server (NTRS)

    Popovic, Zoya; Romisch, Stefania; Rondineau, Sebastien

    2004-01-01

    In this study, a new architecture for Ka-band multi-beam arrays was developed and demonstrated experimentally. The goal of the investigation was to demonstrate a new architecture that has the potential of reducing the cost as compared to standard expensive phased array technology. The goals of this specific part of the project, as stated in the yearly statement of work in the original proposal are: 1. Investigate bounds on performance of multi-beam lens arrays in terms of beamwidths, volume (size), isolation between beams, number of simultaneous beams, etc. 2. Design a small-scale array to demonstrate the principle. The array will be designed for operation around 3OGHz (Ka-band), with two 10-degree beamwidth beams. 3. Investigate most appropriate way to accomplish fine-tuning of the beam pointing within 5 degrees around the main beam pointing angle.