Science.gov

Sample records for adaptive microphone array

  1. Multi-microphone adaptive array augmented with visual cueing.

    PubMed

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

    2012-01-01

    We present the development of an audiovisual array that enables hearing aid users to converse with multiple speakers in reverberant environments with significant speech babble noise where their hearing aids do not function well. The system concept consists of a smartphone, a smartphone accessory, and a smartphone software application. The smartphone accessory concept is a multi-microphone audiovisual array in a form factor that allows attachment to the back of the smartphone. The accessory will also contain a lower power radio by which it can transmit audio signals to compatible hearing aids. The smartphone software application concept will use the smartphone's built in camera to acquire images and perform real-time face detection using the built-in face detection support of the smartphone. The audiovisual beamforming algorithm uses the location of talking targets to improve the signal to noise ratio and consequently improve the user's speech intelligibility. Since the proposed array system leverages a handheld consumer electronic device, it will be portable and low cost. A PC based experimental system was developed to demonstrate the feasibility of an audiovisual multi-microphone array and these results are presented. PMID:23366063

  2. 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.

  3. Speech Enhancement Using Microphone Arrays.

    NASA Astrophysics Data System (ADS)

    Adugna, Eneyew

    Arrays of sensors have been employed effectively in communication systems for the directional transmission and reception of electromagnetic waves. Among the numerous benefits, this helps improve the signal-to-interference ratio (SIR) of the signal at the receiver. Arrays have since been used in related areas that employ propagating waves for the transmission of information. Several investigators have successfully adopted array principles to acoustics, sonar, seismic, and medical imaging. In speech applications the microphone is used as the sensor for acoustic data acquisition. The performance of subsequent speech processing algorithms--such as speech recognition or speaker recognition--relies heavily on the level of interference within the transduced or recorded speech signal. The normal practice is to use a single, hand-held or head-mounted, microphone. Under most environmental conditions, i.e., environments where other acoustic sources are also active, the speech signal from a single microphone is a superposition of acoustic signals present in the environment. Such cases represent a lower SIR value. To alleviate this problem an array of microphones--linear array, planar array, and 3-dimensional arrays--have been suggested and implemented. This work focuses on microphone arrays in room environments where reverberation is the main source of interference. The acoustic wave incident on the array from a point source is sampled and recorded by a linear array of sensors along with reflected waves. Array signal processing algorithms are developed and used to remove reverberations from the signal received by the array. Signals from other positions are considered as interference. Unlike most studies that deal with plane waves, we base our algorithm on spherical waves originating at a source point. This is especially true for room environments. The algorithm consists of two stages--a first stage to locate the source and a second stage to focus on the source. The first part

  4. Higher order differential-integral microphone arrays.

    PubMed

    Abhayapala, Thushara D; Gupta, Aastha

    2010-05-01

    This paper develops theory to design higher order directional microphone arrays. The proposed higher order designs have similar inter sensor spacings as traditional first and second order differential arrays. The Jacobi-Anger expansion is used to exploit the underlying structure of microphone signals from pairs of closely spaced sensors. Specifically, the difference and sum of these microphone signals are processed to design the novel directional array. PMID:21117719

  5. Robust speech coding using microphone arrays

    NASA Astrophysics Data System (ADS)

    Li, Zhao

    1998-09-01

    To achieve robustness and efficiency for voice communication in noise, the noise suppression and bandwidth compression processes are combined to form a joint process using input from an array of microphones. An adaptive beamforming technique with a set of robust linear constraints and a single quadratic inequality constraint is used to preserve desired signal and to cancel directional plus ambient noise in a small room environment. This robustly constrained array processor is found to be effective in limiting signal cancelation over a wide range of input SNRs (-10 dB to +10 dB). The resulting intelligibility gains (8-10 dB) provide significant improvement to subsequent CELP coding. In addition, the desired speech activity is detected by estimating Target-to-Jammer Ratios (TJR) using subband correlations between different microphone inputs or using signals within the Generalized Sidelobe Canceler directly. These two novel techniques of speech activity detection for coding are studied thoroughly in this dissertation. Each is subsequently incorporated with the adaptive array and a 4.8 kbps CELP coder to form a Variable Bit Kate (VBR) coder with noise canceling and Spatial Voice Activity Detection (SVAD) capabilities. This joint noise suppression and bandwidth compression system demonstrates large improvements in desired speech quality after coding, accurate desired speech activity detection in various types of interference, and a reduction in the information bits required to code the speech.

  6. Arrays of Miniature Microphones for Aeroacoustic Testing

    NASA Technical Reports Server (NTRS)

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

    2007-01-01

    A phased-array system comprised of custom-made and commercially available microelectromechanical system (MEMS) silicon microphones and custom ancillary hardware has been developed for use in aeroacoustic testing in hard-walled and acoustically treated wind tunnels. Recent advances in the areas of multi-channel signal processing and beam forming have driven the construction of phased arrays containing ever-greater numbers of microphones. Traditional obstacles to this trend have been posed by (1) the high costs of conventional condenser microphones, associated cabling, and support electronics and (2) the difficulty of mounting conventional microphones in the precise locations required for high-density arrays. The present development overcomes these obstacles. One of the hallmarks of the new system is a series of fabricated platforms on which multiple microphones can be mounted. These mounting platforms, consisting of flexible polyimide circuit-board material (see left side of figure), include all the necessary microphone power and signal interconnects. A single bus line connects all microphones to a common power supply, while the signal lines terminate in one or more data buses on the sides of the circuit board. To minimize cross talk between array channels, ground lines are interposed as shields between all the data bus signal lines. The MEMS microphones are electrically connected to the boards via solder pads that are built into the printed wiring. These flexible circuit boards share many characteristics with their traditional rigid counterparts, but can be manufactured much thinner, as small as 0.1 millimeter, and much lighter with boards weighing as much as 75 percent less than traditional rigid ones. For a typical hard-walled wind-tunnel installation, the flexible printed-circuit board is bonded to the tunnel wall and covered with a face sheet that contains precise cutouts for the microphones. Once the face sheet is mounted, a smooth surface is established over

  7. 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. PMID:26936583

  8. Using coprime microphone arrays for direction-of-arrival estimation

    NASA Astrophysics Data System (ADS)

    Nichols, John Paul

    Direction-of-arrival estimation using microphone arrays requires many sensors to reduce beam width in order to achieve precise location estimation in a noisy environment. Coprime linear microphone arrays allow for narrow beams with fewer sensors. Coprime sensing is a type of sparse sensing, meaning that the microphone elements are fewer and more spaced out than in a traditional array without sacrificing resolution, but requiring more post-processing. A coprime microphone array is made up of two overlapping uniform linear arrays with M and N sensors, where M and N are coprime. By applying spatial filtering with both arrays and combining their outputs, M+N sensors can yield MN directional bands. In this work, the coprime array theory is implemented experimentally for the first time with a microphone array to estimate the location of multiple uncorrelated sources in a noisy environment. Both simulated and measured results will be discussed.

  9. An acoustical array combining microphones and piezoelectric devices.

    PubMed

    Matsumoto, Mitsuharu; Hashimoto, Shuji

    2008-04-01

    This paper describes an acoustical array combining microphones and piezoelectric devices. Conventional microphone arrays have been widely utilized to realize noise reduction, sound separation and direction of arrival estimation system. However, when a conventional microphone array is mounted on a real system, such as a machine, vehicle or robot, the microphones are set extremely close to the system's actual body. In such cases, the noise from the system itself, such as motors, gears, and engines, namely internal noise, often becomes a troublesome problem. It is difficult to reduce internal noise utilizing a conventional microphone array because internal noise sources are extremely close to the microphones. As internal noise is not always stationary, statistically independent or sparse, most useful blind source separation approaches, such as independent component analysis and the sparseness approach, cannot be employed. Our aim is to reduce internal noise utilizing microphones and piezoelectric devices attached to the internal noise source. In this paper, a general description of the acoustical array is formulated and the characteristic features of microphones and piezoelectric devices in an acoustical array are given. An acoustical array combining microphones and piezoelectric devices is also described with some experimental results. PMID:18397019

  10. Rotor noise measurement using a directional microphone array

    NASA Technical Reports Server (NTRS)

    Marcolini, Michael A.; Brooks, Thomas F.

    1987-01-01

    A directional array of microphones was used to measure the noise from a 40 percent scale model rotor in a large aeroacoustic wind tunnel. The development and design of this directional array is described. A design goal was that the array focus on a constant sensing area over a broad frequency range. The implementation of the array design is presented, followed by sample results for several different rotor test conditions. The directional array spectral results are compared with predictions of broadband self noise, and with total rotor noise measurements obtained from individual microphones of the array. The directional array is demonstrated to be a useful tool in examining noise source distributions.

  11. 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.

  12. 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.

  13. 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

  14. 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

  15. 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

  16. 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.

  17. Beamforming with microphone arrays for directional sources.

    PubMed

    Bouchard, Christian; Havelock, David I; Bouchard, Martin

    2009-04-01

    Beamforming is done with an array of sensors to achieve a directional or spatially-specific response by using a model of the arriving wavefront. Real acoustic sources may deviate from the conventional plane wave or monopole model, causing decreased array gain or a total breakdown of beamforming. An alternative to beamforming with the conventional source model is presented which avoids this by using a more general source model. The proposed method defines a set of "sub-beamformers," each designed to respond to a different spatial mode of the source. The outputs of the individual sub-beamformers are combined in a weighted sum to give an overall output of better quality than that of a conventional (monopole) beamformer. It is shown that with appropriate weighting, the optimum array gain can be achieved. A simple method is demonstrated to estimate the weighted sum, based on the observed data. The variance and bias of the estimate in the presence of noise are evaluated. Simulation and experimentally measured results are shown for a simple directive source. In the experiment, the proposed method provides an array gain of about 11 dB while beamforming using a point source model achieves only -4 dB. PMID:19354386

  18. 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.

  19. 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?

  20. Non-Contact Surface Wave Scanning of Pavements Using a Rolling Microphone Array

    NASA Astrophysics Data System (ADS)

    Ryden, Nils; Lowe, Michael J. S.; Cawley, Peter

    2008-02-01

    We present experiments with a rolling multichannel microphone array where measurements can be taken continuously while moving. Leaky air-coupled surface waves are measured with ordinary non-directional audio microphones. Results show that microphones can be successfully used to produce a realistic phase velocity spectrum even while moving along the surface. The paper presents some theoretical background along with initial experimental results using the moving microphone array.

  1. Kalman filter-based microphone array signal processing using the equivalent source model

    NASA Astrophysics Data System (ADS)

    Bai, Mingsian R.; Chen, Ching-Cheng

    2012-10-01

    This paper demonstrates that microphone array signal processing can be implemented by using adaptive model-based filtering approaches. Nearfield and farfield sound propagation models are formulated into state-space forms in light of the Equivalent Source Method (ESM). In the model, the unknown source amplitudes of the virtual sources are adaptively estimated by using Kalman filters (KFs). The nearfield array aimed at noise source identification is based on a Multiple-Input-Multiple-Output (MIMO) state-space model with minimal realization, whereas the farfield array technique aimed at speech quality enhancement is based on a Single-Input-Multiple-Output (SIMO) state-space model. Performance of the nearfield array is evaluated in terms of relative error of the velocity reconstructed on the actual source surface. Numerical simulations for the nearfield array were conducted with a baffled planar piston source. From the error metric, the proposed KF algorithm proved effective in identifying noise sources. Objective simulations and subjective experiments are undertaken to validate the proposed farfield arrays in comparison with two conventional methods. The results of objective tests indicated that the farfield arrays significantly enhanced the speech quality and word recognition rate. The results of subjective tests post-processed with the analysis of variance (ANOVA) and a post-hoc Fisher's least significant difference (LSD) test have shown great promise in the KF-based microphone array signal processing techniques.

  2. Microphone array power ratio for quality assessment of reverberated speech

    NASA Astrophysics Data System (ADS)

    Berkun, Reuven; Cohen, Israel

    2015-12-01

    Speech signals in enclosed environments are often distorted by reverberation and noise. In speech communication systems with several randomly distributed microphones, involving a dynamic speaker and unknown source location, it is of great interest to monitor the perceived quality at each microphone and select the signal with the best quality. Most of existing approaches for quality estimation require prior information or a clean reference signal, which is unfortunately seldom available. In this paper, a practical non-intrusive method for quality assessment of reverberated speech signals is proposed. Using a statistical model of the reverberation process, we examine the energies as measured by unidirectional elements in a microphone array. By measuring the power ratio, we obtain a measure for the amount of reverberation in the received acoustic signals. This measure is then utilized to derive a blind estimation of the direct-to-reverberation energy ratio in the room. The proposed approach attains a simple, reliable, and robust quality measure, shown here through persuasive simulation results.

  3. Simulation of fixed microphone arrays for directional hearing aids.

    PubMed

    Liu, C; Sideman, S

    1996-08-01

    Microphone arrays with fixed optimum weights are known to suppress the background noise and reverberation that severely reduce the effectiveness of conventional hearing aids. By means of a general technique for digital frequency-domain implementation of optimum broadband arrays that was developed recently [C. Liu and S. Sideman, J. Acoust. Soc. Am. 98, 241-247 (1995)], a practically promising system is proposed to realize the arrays with the well-known sensitivity-constrained superdirective beamforming weights, and with five identical omnidirectional, cardioid, hypercardioid, or dipole microphones, respectively, in the endfire or broadside configurations, which were theoretically proposed by Stadler and Rabinowitz [J. Acoust. Soc. Am. 94, 1332-1342 (1993)]. The digital broadband frequency-domain beamforming system allows the broadband superdirective beamforming weights to be faithfully and independently applied to each frequency component of the signal. The practical application of this technique is demonstrated through computer simulation of the system in anechoic situations. Furthermore, its performance in simulated reverberant environments is evaluated. PMID:8759945

  4. 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

  5. Studying Room Acoustics using a Monopole-Dipole Microphone Array

    NASA Technical Reports Server (NTRS)

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

    1997-01-01

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

  6. 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. PMID:26627782

  7. 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.

  8. 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

  9. 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.

  10. 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

  11. 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. PMID:26121610

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

    NASA Technical Reports Server (NTRS)

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

    1974-01-01

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

  13. 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.

  14. 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.

  15. 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.

  16. 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.

  17. Experimental validation of a coprime linear microphone array for high-resolution direction-of-arrival measurements.

    PubMed

    Xiang, Ning; Bush, Dane; Summers, Jason E

    2015-04-01

    Coprime linear microphone arrays allow for narrower beams with fewer sensors. A coprime microphone array consists of two staggered uniform linear subarrays with M and N microphones, where M and N are coprime with each other. By applying spatial filtering to both subarrays and combining their outputs, M+N-1 microphones yield M⋅N directional bands. In this work, the coprime sampling theory is implemented in the form of a linear microphone array of 16 elements with coprime numbers of 9 and 8. This coprime microphone array is experimentally tested to validate the coprime array theory. Both predicted and measured results are discussed. Experimental results confirm that narrow beampatterns as predicted by the coprime sampling theory can be obtained by the coprime microphone array. PMID:25920875

  18. Improved BTE hearing-aid directivity using a directional microphone array

    NASA Astrophysics Data System (ADS)

    Jones, Douglas L.; Lockwood, Michael E.; Lansing, Charissa R.; Feng, Albert S.

    2001-05-01

    Extraction of speech in noise is of great importance to hearing-impaired listeners. Directional microphones are incorporated in some hearing aids to improve noise rejection through increased directivity. A symmetric cardioid response is created either with a single directional microphone, or using beamforming with two omnidirectional microphones. The head-related transfer function (HRTF), however, introduces an asymmetry that cannot be exploited by a linear array of omnidirectional microphones. A new BTE array consisting of a gradient directional microphone with nulls in the front-back vertical plane and two omnidirectional microphones exploits the asymmetry of the HRTF to obtain almost 2 dB better directivity than the best cardioid. HRTFs measured on KEMAR with this array were transformed to the frequency domain, where directivity-maximizing coefficients in each band were derived. The Articulation-Index (AI) weighted directivity gain of this optimal three-element directional array was 6.4 dB greater than a single omnidirectional microphone on the BTE, whereas the directivity gain of the HRTF-optimized two-omni beamformer was 4.6 dB, and the optimal free-field cardioid placed on the head yielded 4.4 dB. [Work partially supported by NIH (NIDCD) under Grant No. 1 R01 DC005762-01A1.

  19. Adaptive noise suppression for a dual-microphone hearing aid.

    PubMed

    Wouters, Jan; Berghe, Jeff Vanden; Maj, Jean-Baptiste

    2002-10-01

    An adaptive beamformer for behind-the-ear dual-microphone hearing aids has been optimized for speech intelligibility enhancement in the presence of disturbing sounds or noise. The noise reduction approach is based on the scheme presented by Vanden Berghe and Wouters (1998). A real-time implementation of the signal processing is realized in Audallion, a wearable, small digital signal processing (DSP) platform. After physical evaluation, speech-in-noise intelligibility tests have been carried out on three normally-hearing and two hearing-impaired subjects. A significant speech reception threshold improvement of 11.3 dB was obtained in a moderately reverberant environment for one jammer sound source (steady speech-weighted noise or multi-talker babble) in a direction of 90 degrees relative to the direction of the speech. PMID:12403608

  20. 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}.

  1. 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.

  2. 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

  3. 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.

  4. 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

  5. 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. PMID:21117762

  6. 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.

  7. 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.

  8. 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.

  9. 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.

  10. 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) PMID:7714267

  11. 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

  12. 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.

  13. 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

  14. Acoustic analysis by spherical microphone array processing of room impulse responses.

    PubMed

    Khaykin, Dima; Rafaely, Boaz

    2012-07-01

    Spherical microphone arrays have been recently used for room acoustics analysis, to detect the direction-of-arrival of early room reflections, and compute directional room impulse responses and other spatial room acoustics parameters. Previous works presented methods for room acoustics analysis using spherical arrays that are based on beamforming, e.g., delay-and-sum, regular beamforming, and Dolph-Chebyshev beamforming. Although beamforming methods provide useful directional selectivity, optimal array processing methods can provide enhanced performance. However, these algorithms require an array cross-spectrum matrix with a full rank, while array data based on room impulse responses may not satisfy this condition due to the single frame data. This paper presents a smoothing technique for the cross-spectrum matrix in the frequency domain, designed for spherical microphone arrays, that can solve the problem of low rank when using room impulse response data, therefore facilitating the use of optimal array processing methods. Frequency smoothing is shown to be performed effectively using spherical arrays, due to the decoupling of frequency and angular components in the spherical harmonics domain. Experimental study with data measured in a real auditorium illustrates the performance of optimal array processing methods such as MUSIC and MVDR compared to beamforming. PMID:22779475

  15. 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. PMID:20329837

  16. 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.

  17. 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.

  18. Micromachined microphone array on a chip for turbulent boundary layer measurements

    NASA Astrophysics Data System (ADS)

    Krause, Joshua Steven

    A surface micromachined microphone array on a single chip has been successfully designed, fabricated, characterized, and tested for aeroacoustic purposes. The microphone was designed to have venting through the diaphragm, 64 elements (8x8) on the chip, and used a capacitive transduction scheme. The microphone was fabricated using the MEMSCAP PolyMUMPs process (a foundry polysilicon surface micromachining process) along with facilities at Tufts Micro and Nano Fabrication Facility (TMNF) where a Parylene-C passivation layer deposition and release of the microstructures were performed. The devices are packaged with low profile interconnects, presenting a maximum of 100 mum of surface topology. The design of an individual microphone was completed through the use of a lumped element model (LEM) to determine the theoretical performance of the microphone. Off-chip electronics were created to allow the microphone array outputs to be redirected to one of two channels, allowing dynamic reconfiguration of the effective transducer shape in software and provide 80 dB off isolation. The characterization was completed through the use of laser Doppler vibrometry (LDV), acoustic plane wave tube and free-field calibration, and electrical noise floor testing in a Faraday cage. Measured microphone sensitivity is 0.15 mV/Pa for an individual microphone and 8.7 mV/Pa for the entire array, in close agreement with model predictions. The microphones and electronics operate over the 200--40 000 Hz band. The dynamic range extends from 60 dB SPL in a 1 Hz band to greater than 150 dB SPL. Element variability was +/-0.05 mV/Pa in sensitivity with an array yield of 95%. Wind tunnel testing at flow rates of up to 205.8 m/s indicates that the devices continue to operate in flow without damage, and can be successfully reconfigured on the fly. Care has been taken to systematically remove contaminating signals (acoustic, vibration, and noise floor) from the wind tunnel data to determine actual

  19. 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.

  20. 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. PMID:24603636

  1. Speech Enhancement Using a Square Microphone Array in the Presence of Directional and Diffuse Noise

    NASA Astrophysics Data System (ADS)

    Ogawa, Tetsuji; Takada, Shintaro; Akagiri, Kenzo; Kobayashi, Tetsunori

    We propose a new speech enhancement method suitable for mobile devices used in the presence of various types of noise. In order to achieve high-performance speech recognition and auditory perception in mobile devices, various types of noise have to be removed under the constraints of a space-saving microphone arrangement and few computational resources. The proposed method can reduce both the directional noise and the diffuse noise under the abovementioned constraints for mobile devices by employing a square microphone array and conducting low-computational-cost processing that consists of multiple null beamforming, minimum power channel selection, and Wiener filtering. The effectiveness of the proposed method is experimentally verified in terms of speech recognition accuracy and speech quality when both the directional noise and the diffuse noise are observed simultaneously; this method reduces the number of word errors and improves the log-spectral distances as compared to conventional methods.

  2. Room acoustics investigations of beamforming performance using coprime linear microphone arrays

    NASA Astrophysics Data System (ADS)

    Bush, Dane

    Linear microphone arrays are powerful tools for determining the direction of a sound source. Traditionally, uniform linear arrays (ULA) have inter-element spacing of half of the wavelength in question. This produces the narrowest possible beam without introducing grating lobes -- a form of aliasing governed by the spatial Nyquist theorem. Grating lobes are often undesirable because they make direction of arrival indistinguishable among their passband angles. Exploiting coprime number theory however, an array can be arranged sparsely with fewer total elements, exceeding the aforementioned spatial sampling limit separation. Two sparse ULA sub-arrays with coprime number of elements, when nested properly, each produce narrow grating lobes that overlap with one another exactly in just one direction. By combining the sub-array outputs it is possible to retain the shared beam while mostly canceling the other superfluous grating lobes. In this work beam patterns are simulated for a range of single frequencies, as well as for arbitrary bands of frequencies. Three coprime microphone arrays are built with different lengths and sub-array spacings. Two different techniques are explored for sub-array data processing and combination. Experimental beam patterns are shown to correspond with simulated results even at frequencies other than the array's design frequency. Beam width and side lobe locations are shown to correspond to the derived values. Side lobes in the directional pattern are mitigated by increasing bandwidth of analyzed signals. Accurate single-source direction of arrival (DOA) estimation is shown to be possible in free field and reverberant conditions. DOA estimation is also implemented for two simultaneous noise sources in the free field condition. Room reflections can be resolved in the reverberant condition, provided adequate reduction of side lobes.

  3. 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

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

    PubMed

    Bush, Dane; Xiang, Ning

    2015-07-01

    Coprime arrays represent a form of sparse sensing which can achieve narrow beams using relatively few elements, exceeding the spatial Nyquist sampling limit. The purpose of this paper is to expand on and experimentally validate coprime array theory in an acoustic implementation. Two nested sparse uniform linear subarrays with coprime number of elements ( M and N) each produce grating lobes that overlap with one another completely in just one direction. When the subarray outputs are combined it is possible to retain the shared beam while mostly canceling the other superfluous grating lobes. In this way a small number of microphones ( N+M-1) creates a narrow beam at higher frequencies, comparable to a densely populated uniform linear array of MN microphones. In this work beampatterns are simulated for a range of single frequencies, as well as bands of frequencies. Narrowband experimental beampatterns are shown to correspond with simulated results even at frequencies other than the arrays design frequency. Narrowband side lobe locations are shown to correspond to the theoretical values. Side lobes in the directional pattern are mitigated by increasing bandwidth of analyzed signals. Direction of arrival estimation is also implemented for two simultaneous noise sources in a free field condition. PMID:26233043

  5. Novel human-robot interface integrating real-time visual tracking and microphone-array signal processing

    NASA Astrophysics Data System (ADS)

    Mizoguchi, Hiroshi; Shigehara, Takaomi; Goto, Yoshiyasu; Hidai, Ken-ichi; Mishima, Taketoshi

    1998-10-01

    This paper proposes a novel human robot interface that is an integration of real time visual tracking and microphone array signal processing. The proposed interface is intended to be used as a speech signal input method for human collaborative robot. Utilizing it, the robot can clearly listen human master's voice remotely as if a wireless microphone were put just in front of the master. A novel technique to form `acoustic focus' at human face is developed. To track and locate the face dynamically, real time face tracking and stereo vision are utilized. To make the acoustic focus at the face, microphones array is utilized. Setting gain and delay of each microphone properly enables to form acoustic focus at desired location. The gain and delay are determined based upon the location of the face. Results of preliminary experiments and simulations demonstrate feasibility of the proposed idea.

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

    PubMed

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

    2015-04-01

    Ground reflections cause problems when estimating the direction of arrival of aircraft noise. In traditional methods, based on the time differences between the microphones of a compact array, they may cause a significant loss of accuracy in the vertical direction. This study evaluates the use of first-order directional microphones, instead of omnidirectional, with the aim of reducing the amplitude of the reflected sound. Such a modification allows the problem to be treated as in free field conditions. Although further tests are needed for a complete evaluation of the method, the experimental results presented here show that under the particular conditions tested the vertical angle error is reduced ∼10° for both jet and propeller aircraft by selecting an appropriate directivity pattern. It is also shown that the final level of error depends on the vertical angle of arrival of the sound, and that the estimates of the horizontal angle of arrival are not influenced by the directivity pattern of the microphones nor by the reflective properties of the ground. PMID:25920843

  7. 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

  8. 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.

  9. 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

  10. 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.

  11. Automatic estimation of position and orientation of an acoustic source by a microphone array network.

    PubMed

    Nakano, Alberto Yoshihiro; Nakagawa, Seiichi; Yamamoto, Kazumasa

    2009-12-01

    A method which automatically provides the position and orientation of a directional acoustic source in an enclosed environment is proposed. In this method, different combinations of the estimated parameters from the received signals and the microphone positions of each array are used as inputs to the artificial neural network (ANN). The estimated parameters are composed of time delay estimates (TDEs), source position estimates, distance estimates, and energy features. The outputs of the ANN are the source orientation (one out of four possible orientations shifted by 90 degrees and either the best array which is defined as the nearest to the source) or the source position in two dimensional/three dimensional (2D/3D) space. This paper studies the position and orientation estimation performances of the ANN for different input/output combinations (and different numbers of hidden units). The best combination of parameters (TDEs and microphone positions) yields 21.8% reduction in the average position error compared to the following baselines and a correct orientation ratio greater than 99%. Position localization baselines consist of a time delay of arrival based method with an average position error of 34.1 cm and the steered response power with phase transform method with an average position error of 29.8 cm in 3D space. PMID:20000922

  12. Measurement and simulation of surface roughness noise using phased microphone arrays

    NASA Astrophysics Data System (ADS)

    Liu, Y.; Dowling, A. P.; Shin, H.-C.

    2008-07-01

    A turbulent boundary-layer flow over a rough wall generates a dipole sound field as the near-field hydrodynamic disturbances in the turbulent boundary-layer scatter into radiated sound at small surface irregularities. In this paper, phased microphone arrays are applied to the measurement and simulation of surface roughness noise. The radiated sound from two rough plates and one smooth plate in an open jet is measured at three streamwise locations, and the beamforming source maps demonstrate the dipole directivity. Higher source strengths can be observed on the rough plates which also enhance the trailing-edge noise. A prediction scheme in previous theoretical work is used to describe the strength of a distribution of incoherent dipoles and to simulate the sound detected by the microphone array. Source maps of measurement and simulation exhibit satisfactory similarities in both source pattern and source strength, which confirms the dipole nature and the predicted magnitude of roughness noise. However, the simulations underestimate the streamwise gradient of the source strengths and overestimate the source strengths at the highest frequency.

  13. 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.

  14. 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.

  15. 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

  16. A maximum likelihood direction of arrival estimation method for open-sphere microphone arrays in the spherical harmonic domain.

    PubMed

    Hu, Yuxiang; Lu, Jing; Qiu, Xiaojun

    2015-08-01

    Open-sphere microphone arrays are preferred over rigid-sphere arrays when minimal interaction between array and the measured sound field is required. However, open-sphere arrays suffer from poor robustness at null frequencies of the spherical Bessel function. This letter proposes a maximum likelihood method for direction of arrival estimation in the spherical harmonic domain, which avoids the division of the spherical Bessel function and can be used at arbitrary frequencies. Furthermore, the method can be easily extended to wideband implementation. Simulation and experiment results demonstrate the superiority of the proposed method over the commonly used methods in open-sphere configurations. PMID:26328695

  17. Using hearing aid adaptive directional microphones to enhance cochlear implant performance.

    PubMed

    Chung, King; Zeng, Fan-Gang

    2009-04-01

    The goal of this study was to investigate whether adaptive microphone directionality could enhance cochlear implant performance. Speech stimuli were created by fitting a digital hearing aid with programmable omnidirectional (OM), fixed directional (FDM), or adaptive directional (ADM) microphones to KEMAR, and recording the hearing aid output in three noise conditions. The first condition simulated a diffused field with noise sources from five stationary locations, whereas the second and third condition represented one or three non-stationary locations in the back hemifield of KEMAR. Speech was always presented to 0 degrees azimuth and the overall signal-to-noise ratio (SNR) was +5 dB in the sound field. Eighteen postlingually deafened cochlear implant users listened to the recorded test materials via the direct audio input of their speech processors. Their speech recognition ability and overall sound quality preferences were assessed and the correlation between the amount of noise reduction and the improvement in speech recognition were calculated. The results indicated that ADM yielded significantly better speech recognition scores and overall sound quality preference than FDM and OM in all three noise conditions and the improvement in speech recognition scores was highly correlated with the amount of noise reduction. Factors influencing the noise level are discussed. PMID:19450437

  18. 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.

  19. 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

    2014-01-01

    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

  20. 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

  1. Adaptive arrays for satellite communications

    NASA Technical Reports Server (NTRS)

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

    1984-01-01

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

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

    NASA Astrophysics Data System (ADS)

    Gover, Bradford Noel

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

  3. Simulation of multi-microphone hearing aids in multiple interference environments.

    PubMed

    Hoffman, M W; Stewart, R W

    1996-08-01

    In this study, the advantages of Digital Signal Processing (DSP) hardware for hearing aids are investigated in the context of multiple microphone arrays. One key question in multiple microphone DSP system design remains the allocation of processing resources between the number of microphones and the number of adjustable tap weights applied to each microphone. This study addresses the appropriate distribution of these resources for currently implementable adaptive DSP systems. Comparisons are made by computer simulation that includes acoustic headshadow, reverberation effects and non-ideal microphone array hardware in a wide variety of environments. Variations in the number of interfering sources, the amount of reverberation and the microphone array configuration leads to several important conclusions. Performance improvements provided by the processors are reported as the broadband unweighted signal-to-babble ratio of pre-emphasized speech and speech-shaped babble. Results are demonstrated for both fixed and robust adaptive systems. PMID:8879690

  4. 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.

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

    NASA Astrophysics Data System (ADS)

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

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

  6. 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.

  7. 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

  8. 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.

  9. 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.

  10. 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.

  11. 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.

  12. Passive Acoustic Source Localization at a Low Sampling Rate Based on a Five-Element Cross Microphone Array

    PubMed Central

    Kan, Yue; Wang, Pengfei; Zha, Fusheng; Li, Mantian; Gao, Wa; Song, Baoyu

    2015-01-01

    Accurate acoustic source localization at a low sampling rate (less than 10 kHz) is still a challenging problem for small portable systems, especially for a multitasking micro-embedded system. A modification of the generalized cross-correlation (GCC) method with the up-sampling (US) theory is proposed and defined as the US-GCC method, which can improve the accuracy of the time delay of arrival (TDOA) and source location at a low sampling rate. In this work, through the US operation, an input signal with a certain sampling rate can be converted into another signal with a higher frequency. Furthermore, the optimal interpolation factor for the US operation is derived according to localization computation time and the standard deviation (SD) of target location estimations. On the one hand, simulation results show that absolute errors of the source locations based on the US-GCC method with an interpolation factor of 15 are approximately from 1/15- to 1/12-times those based on the GCC method, when the initial same sampling rates of both methods are 8 kHz. On the other hand, a simple and small portable passive acoustic source localization platform composed of a five-element cross microphone array has been designed and set up in this paper. The experiments on the established platform, which accurately locates a three-dimensional (3D) near-field target at a low sampling rate demonstrate that the proposed method is workable. PMID:26057042

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

    PubMed

    Wei, Qingkai; Huang, Xun; Peers, Edward

    2013-06-01

    An imaging method of acoustic spinning modes propagating within a circular duct simply with surface pressure information is introduced in this paper. The proposed method is developed in a theoretical way and is demonstrated by a numerical simulation case. Nowadays, the measurements within a duct have to be conducted using in-duct microphone array, which is unable to provide information of complete acoustic solutions across the test section. The proposed method can estimate immeasurable information by forming a so-called observer. The fundamental idea behind the testing method was originally developed in control theory for ordinary differential equations. Spinning mode propagation, however, is formulated in partial differential equations. A finite difference technique is used to reduce the associated partial differential equations to a classical form in control. The observer method can thereafter be applied straightforwardly. The algorithm is recursive and, thus, could be operated in real-time. A numerical simulation for a straight circular duct is conducted. The acoustic solutions on the test section can be reconstructed with good agreement to analytical solutions. The results suggest the potential and applications of the proposed method. PMID:23742352

  14. A Bayesian direction-of-arrival model for an undetermined number of sources using a two-microphone array.

    PubMed

    Escolano, Jose; Xiang, Ning; Perez-Lorenzo, Jose M; Cobos, Maximo; Lopez, Jose J

    2014-02-01

    Sound source localization using a two-microphone array is an active area of research, with considerable potential for use with video conferencing, mobile devices, and robotics. Based on the observed time-differences of arrival between sound signals, a probability distribution of the location of the sources is considered to estimate the actual source positions. However, these algorithms assume a given number of sound sources. This paper describes an updated research account on the solution presented in Escolano et al. [J. Acoust. Am. Soc. 132(3), 1257-1260 (2012)], where nested sampling is used to explore a probability distribution of the source position using a Laplacian mixture model, which allows both the number and position of speech sources to be inferred. This paper presents different experimental setups and scenarios to demonstrate the viability of the proposed method, which is compared with some of the most popular sampling methods, demonstrating that nested sampling is an accurate tool for speech localization. PMID:25234883

  15. Inverse problem with beamforming regularization matrix applied to sound source localization in closed wind-tunnel using microphone array

    NASA Astrophysics Data System (ADS)

    Padois, Thomas; Gauthier, Philippe-Aubert; Berry, Alain

    2014-12-01

    Microphone arrays have become a standard technique to localize and quantify source in aeroacoustics. The simplest approach is the beamforming that provides noise source maps with large main lobe and strong side lobes at low frequency. Since a decade, the focus is set on deconvolution techniques such as DAMAS or Clean-SC. While the source map is clearly improved, these methods require a large computation time. In this paper, we propose a sound source localization technique based on an inverse problem with beamforming regularization matrix called Hybrid Method. With synthetic data, we show that the side lobes are removed and the main lobe is narrower. Moreover, if the sound noise source map provided by this method is used as input in the DAMAS process, the number of DAMAS iterations is highly reduced. The Hybrid Method is applied to experimental data obtained in a closed wind-tunnel. In both cases of acoustic or aeroacoustic data, the source is correctly detected. The proposed Hybrid Method is found simple to implement and the computation time is low if the number of scan points is reasonable.

  16. 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.

  17. a Study of Microphone Arrays for the Location of Vibrational Sound Sources

    NASA Astrophysics Data System (ADS)

    Matzumoto, Andres Esteban Perez

    Available from UMI in association with The British Library. The original objective of the work was to develop an acoustic imaging technique, Nearfield Acoustic Holography (NAH), into a reasonably affordable practical system for in-situ applications in an industrial environment. In order to place NAH in the general context of source identification techniques, the thesis summarizes theoretical considerations about sound sources and sound fields, and the principles of different sound source location techniques used in practical situations. The development of NAH theory for planar arrays is the central point of this summary, and the different systems that apply this theory are discussed. Theoretical research and computer simulations show that, at the present state of the art, sound source reconstructions using NAH can only work in certain cases which have to be analyzed individually. The limitations of the theory are better understood when the source reconstruction is studied using the proper methodology for the inherent ill-posed inverse problem. Such as study allowed us to improve the theoretical framework, and to obtain stable source reconstructions. A reliable system for field measurements of different types of sources was not found to be feasible at present. However, it is shown that the inverse problem theory allows us to overcome some of the limitations of the current theory for field measurements. The difficulties encountered with NAH, and the study of the inverse problem, led us to attempt to develop an alternative, simpler, system which could minimize the inverse problem of source reconstruction. The system considered was a hemispherical array with analogue signal processing. Applying this principle transforms the source reconstruction problem into a re-modelling problem, reducing the inverse problem to the solution of a stable direct problem. Although the antenna could not be tested to its full potential, the initial results were not applicable for source

  18. 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.

  19. 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…

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

    NASA Technical Reports Server (NTRS)

    Cohn, R. B. (Inventor)

    1983-01-01

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

  1. 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.

  2. 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.

  3. Adaptive identification by systolic arrays. Master's thesis

    SciTech Connect

    Willis, P.A.

    1987-12-01

    This thesis is concerned with the implementation of an adaptive-identification algorithm using parallel processing and systolic arrays. In particular, discrete samples of input and output data of a system with uncertain characteristics are used to determine the parameters of its model. The identification algorithm is based on recursive least squares, QR decomposition, and block-processing techniques with covariance resetting. Along similar lines as previous approaches, the identification process is based on the use of Givens rotations. This approach uses the Cordic algorithm for improved numerical efficiency in performing the rotations. Additionally, floating-point and fixed-point arithmetic implementations are compared.

  4. 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.

  5. 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

  6. 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.

  7. 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.

  8. Phase Calibration of Microphones by Measurement in the Free-field

    NASA Technical Reports Server (NTRS)

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

    2006-01-01

    Over the past several years, significant effort has been expended at NASA Langley developing new Micro-Electro-Mechanical System (MEMS)-based microphone directional array instrumentation for high-frequency aeroacoustic measurements in wind tunnels. This new type of array construction solves two challenges which have limited the widespread use of large channel-count arrays, namely by providing a lower cost-per-channel and a simpler method for mounting microphones in wind tunnels and in field-deployable arrays. The current generation of array instrumentation is capable of extracting accurate noise source location and directivity on a variety of airframe components using sophisticated data reduction algorithms [1-2]. Commercially-available MEMS microphones are condenser-type devices and have some desirable characteristics when compared with conventional condenser-type microphones. The most important advantages of MEMS microphones are their size, price, and power consumption. However, the commercially-available units suffer from certain important shortcomings. Based on experiments with array prototypes, it was found that both the bandwidth and the sound pressure limit of the microphones should be increased significantly to improve the performance and flexibility of the microphone array [3]. It was also desired to modify the packaging to eliminate unwanted Helmholtz resonance s exhibited by the commercial devices. Thus, new requirements were defined as follows: Frequency response: 100 Hz to 100 KHz (+/-3dB) Upper sound pressure limit: Design 1: 130 dB SPL (THD less than 5%) Design 2: 150-160 dB SPL (THD less than 5%) Packaging: 3.73 x 6.13 x 1.3 mm can with laser-etched lid. In collaboration with Novusonic Acoustic Innovation, NASA modified a Knowles SiSonic MEMS design to meet these new requirements. Coupled with the design of the enhanced MEMS microphones was the development of a new calibration method for simultaneously obtaining the sensitivity and phase response of

  9. 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.

  10. 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.

  11. Adaptive calibration of a three-microphone system for acoustic waveguide characterization under time-varying conditions.

    PubMed

    van Walstijn, Maarten; de Sanctis, Giovanni

    2014-02-01

    The pressure and velocity field in a one-dimensional acoustic waveguide can be sensed in a non-intrusive manner using spatially distributed microphones. Experimental characterization with sensor arrangements of this type has many applications in measurement and control. This paper presents a method for measuring the acoustic variables in a duct under fluctuating propagation conditions with specific focus on in-system calibration and tracking of the system parameters of a three-microphone measurement configuration. The tractability of the non-linear optimization problem that results from taking a parametric approach is investigated alongside the influence of extraneous measurement noise on the parameter estimates. The validity and accuracy of the method are experimentally assessed in terms of the ability of the calibrated system to separate the propagating waves under controlled conditions. The tracking performance is tested through measurements with a time-varying mean flow, including an experiment conducted under propagation conditions similar to those in a wind instrument during playing. PMID:25234899

  12. Protection of the main maximum in adaptive antenna arrays

    NASA Astrophysics Data System (ADS)

    Pistolkors, A. A.

    1980-12-01

    An adaptive algorithm based on the solution of the problem of minimizing the noise at the output of an array when a constraint is imposed on the main maximum direction is discussed. The suppression depth for the cases of one and two interferences and the enhancement of the direction-finding capability and resolution of an adaptive array are investigated.

  13. Applications of minimum redundancy arrays in adaptive beamforming

    NASA Astrophysics Data System (ADS)

    Fattouche, M.; Nichols, S. T.; Jorgenson, M. B.

    1991-10-01

    It is shown, through analysis and simulation, that the use of a minimum redundancy array (MRA) in conjunction with an adaptive beamformer results in performance superior to that attained by a comparable system based on an array with uniformly spaced elements, or uniform array (UA) in terms of rejecting interferences located in close angular proximity to the look direction. Further, it is demonstrated that choosing the adaptive elements of a thinned adaptive array (TAA) based on a minimum spatial redundancy criterion, rather than spacing them uniformly, results in improved rejection of main lobe interferences, with negligible degradation in sidelobe interference rejection capabilities.

  14. Temperature-adaptive Circuits on Reconfigurable Analog Arrays

    NASA Technical Reports Server (NTRS)

    Stoica, Adrian; Zebulum, Ricardo S.; Keymeulen, Didier; Ramesham, Rajeshuni; Neff, Joseph; Katkoori, Srinivas

    2006-01-01

    This paper describes a new reconfigurable analog array (MA) architecture and integrated circuit (IC) used to map analog circuits that can adapt to extreme temperatures under programmable control. Algorithm-driven adaptation takes place on the RAA IC. The algorithms are implemented in a separate Field Programmable Gate Array (FPGA) IC, co-located with the RAA in the extreme temperature environment. The experiments demonstrate circuit adaptation over a wide temperature range, from extremely low temperature of -180 C to high 120 C.

  15. 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)

  16. Probe microphone measurements: 20 years of progress.

    PubMed

    Mueller, H G

    2001-06-01

    physicians, and 69% for audiologists in private practice. But more importantly, and a bit puzzling, was the finding that showed that nearly one half of the people who fit hearing aids and have access to this equipment, seldom or never use it. I doubt that the use rate of probe-microphone equipment has changed much in the last three years, and if anything, I suspect it has gone down. Why do I say that? As programmable hearing aids have become the standard fitting in many clinics, it is tempting to become enamoured with the simulated gain curves on the fitting screen, somehow believing that this is what really is happening in the real ear. Additionally, some dispensers have been told that you can't do reliable probe-mic testing with modern hearing aids-this of course is not true, and we'll address this issue in the Frequently Asked Questions portion of this paper. The infrequent use of probe-mic testing among dispensers is discouraging, and let's hope that probe-mic equipment does not suffer the fate of the rowing machine stored in your garage. A lot has changed over the years with the equipment itself, and there are also expanded clinical applications and procedures. We have new manufacturers, procedures, acronyms and noises. We have test procedures that allow us to accurately predict the output of a hearing aid in an infant's ear. We now have digital hearing aids, which provide us the opportunity to conduct real-ear measures of the effects of digital noise reduction, speech enhancement, adaptive feedback, expansion, and all the other features. Directional microphone hearing aids have grown in popularity and what better way to assess the real-ear directivity than with probe-mic measures? The array of assistive listening devices has expanded, and so has the role of the real-ear assessment of these products. And finally, with today's PC -based systems, we can program our hearing aids and simultaneously observe the resulting real-ear effects on the same fitting screen, or even

  17. Effects of additional interfering signals on adaptive array performance

    NASA Technical Reports Server (NTRS)

    Moses, Randolph L.

    1989-01-01

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

  18. 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".

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

    PubMed Central

    Chung, King

    2004-01-01

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

  20. 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.

  1. 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.

  2. 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.

  3. Dynamic Pressure Difference Microphones

    NASA Astrophysics Data System (ADS)

    Werner, E.

    A microphone with a diaphragm that is exposed to the sound field only on one side responds essentially to the sound pressure. This quantity is a scalar, and thus, pressure microphones are essentially omnidirectional (see Chapter 66). However, directional microphones are useful because they make it possible to focus on a source and suppress background noise from other directions. Most directional microphones respond to the gradient of the sound pressure, to combinations of the sound pressure and its gradient, or to combinations of higher order spatial derivatives of the sound pressure.

  4. Implementation of LSCMA adaptive array terminal for mobile satellite communications

    NASA Astrophysics Data System (ADS)

    Zhou, Shun; Wang, Huali; Xu, Zhijun

    2007-11-01

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

  5. Shack-Hartmann wavefront sensor with adaptive holographic lenslet array

    NASA Astrophysics Data System (ADS)

    Podanchuk, Dmytro V.; Dan'ko, Volodymyr P.; Goloborodko, Andrey A.; Sutyagina, Natalia S.

    2009-10-01

    The method of the dynamic range expansion of the Shack-Hartmann wavefront sensor is discussed. It's based on the use of nonlinear dual focus holographic lenslet arrays with the aberration precompensation. The data concerning the optical setup and the technique of adaptive lenslet array producing based on nonlinear holographic recording phenomenon are represented. On the example of spherical wavefronts it is shown, that the use of three lenslet arrays with different amount of the aberration precompensation allows expanding approximately in five times the dynamic range of the sensor four times greater with preserving the specified sensitivity in comparison with the corresponding refractive lenslet array.

  6. 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.

  7. Adaptive laser array-receivers for acoustic waves detection

    NASA Astrophysics Data System (ADS)

    Tuovinen, Hemmo; Murray, Todd W.; Krishnaswamy, Sridhar

    2000-05-01

    Interferometric detection systems typically use a single focused laser point receiver for the detection of acoustic waves. In some cases, where optical damage of the structure is of concern, it may be advantageous to distribute the detection laser energy over an area. This can be done, for example, by using a point-array or a line-array probe. Other advantages of an array receiver include directional sensitivity and frequency selectivity. It is important to notice that laser-array reception is possible only with self-referential interferometers. In this paper adaptive array interferometric detection schemes, which are based on wave mixing in photorefractive bismuth silicate crystal, are described. An adaptive narrow-band laser array receiver of surface acoustic waves is demonstrated. The interferometer is also configured as a linearly frequency modulated (chirped) array receiver. The chirped receiver, when excited with a similarly chirped ultrasonic source, allows pulse compression of the ultrasonic signal thus maintaining high temporal resolution. The signal-to-noise ratio for the different array detection schemes are determined and compared. Several applications of laser-array reception are presented.

  8. 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.

  9. 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.

  10. 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

  11. 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.

  12. Motion compensation for adaptive horizontal line array processing

    NASA Astrophysics Data System (ADS)

    Yang, T. C.

    2003-01-01

    Large aperture horizontal line arrays have small resolution cells and can be used to separate a target signal from an interference signal by array beamforming. High-resolution adaptive array processing can be used to place a null at the interference signal so that the array gain can be much higher than that of conventional beamforming. But these nice features are significantly degraded by the source motion, which reduces the time period under which the environment can be considered stationary from the array processing point of view. For adaptive array processing, a large number of data samples are generally required to minimize the variance of the cross-spectral density, or the covariance matrix, between the array elements. For a moving source and interference, the penalty of integrating over a large number of samples is the spread of signal and interference energy to more than one or two eigenvalues. The signal and interference are no longer clearly identified by the eigenvectors and, consequently, the ability to suppress the interference suffers. We show in this paper that the effect of source motion can be compensated for the (signal) beam covariance matrix, thus allowing integration over a large number of data samples without loss in the signal beam power. We employ an equivalent of a rotating coordinate frame to track the signal bearing change and use the waveguide invariant theory to compensate the signal range change by frequency shifting.

  13. 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.

  14. 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.

  15. 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.

  16. 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.

  17. 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. PMID:27203554

  18. 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.

  19. 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.

  20. 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.

  1. 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.

  2. Howells-Applebaum adaptive superresolution array for accelerated scanning

    NASA Astrophysics Data System (ADS)

    Ohmiya, Manabu; Ogawa, Yasutaka; Itoh, Kiyohiko

    1988-12-01

    An approach is proposed that offers an acclerated scanning rate for a Howells-Applebaum adaptive superresolution array (H-A SRA). Analytical considerations clarify the causes of performance degradation of the H-A SRA at a high scanning rate. Then a suitable steering signal and implementation of an H-A weight control loop (H-A loop) for accelerated scanning are introduced. The weight solution determined by this method is shown to coincide approximately with the optimum Wiener one under some specific signal conditions and antenna parameters. Computer simulations show that the H-A SRA gives much better scanning performance than the conventional array. The system is readily implemented by improving the circuit inserting the steering signal in the H-A loop.

  3. An adaptive array antenna for mobile satellite communications

    NASA Technical Reports Server (NTRS)

    Milne, Robert

    1988-01-01

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

  4. Applications of trimode waveguide feeds in adaptive virtual array antennas

    NASA Astrophysics Data System (ADS)

    Allahgholi Pour, Z.; Shafai, Lotfollah

    2015-03-01

    This paper presents the formation of an adaptive virtual array antenna in a symmetric parabolic reflector antenna illuminated by trimode circular waveguide feeds with different mode alignments. The modes of interest are the TE11, TE21, and TM01 type modes. The terms TE and TM stand for the transverse electric and transverse magnetic modes, respectively. By appropriately exciting these modes and varying the mode orientations inside the primary feed, the effective source of radiation is displaced on the reflector aperture, while the resulting secondary patterns remain axial. Different antenna parameters such as gain, cross polarization, and phase center locations are investigated. It is demonstrated that the extra third mode facilitates the formation of symmetric virtual array antennas with reasonable cross polarization discriminations at the diagonal plane.

  5. 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

  6. 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.

  7. 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.

  8. 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.

  9. Adaptive Transthoracic Refocusing of Dual-Mode Ultrasound Arrays

    PubMed Central

    Casper, Andrew J.; Wan, Yayun; Ebbini, Emad S.

    2010-01-01

    We present experimental validation results of an adaptive, image-based refocusing algorithm of dual-mode ultrasound arrays (DMUAs) in the presence of strongly scattering objects. This study is motivated by the need to develop noninvasive techniques for therapeutic targeting of tumors seated in organs where the therapeutic beam is partially obstructed by the ribcage, e.g., liver and kidney. We have developed an algorithm that takes advantage of the imaging capabilities of DMUAs to identify the ribs and the intercostals within the path of the therapeutic beam to produce a specified power deposition at the target while minimizing the exposure at the rib locations. This image-based refocusing algorithm takes advantage of the inherent registration between the imaging and therapeutic coordinate systems of DMUAs in the estimation of array directivity vectors at the target and rib locations. These directivity vectors are then used in solving a constrained optimization problem allowing for adaptive refocusing, directing the acoustical energy through the intercostals, and avoiding the rib locations. The experimental validation study utilized a 1-MHz, 64-element DMUA in focusing through a block of tissue-mimicking phantom [0.5 dB/(cm·MHz)] with embedded Plexiglas ribs. Single transmit focus (STF) images obtained with the DMUA were used for image-guided selection of the critical and target points to be used for adaptive refocusing. Experimental results show that the echogenicity of the ribs in STF images provide feedback on the reduction of power deposition at rib locations. This was confirmed by direct comparison of measured temperature rise and integrated backscatter at the rib locations. Direct temperature measurements also confirm the improved power deposition at the target and the reduction in power deposition at the rib locations. Finally, we have compared the quality of the image-based adaptive refocusing algorithm with a phase-conjugation solution obtained by direct

  10. Rotating Microphone Rake Measures Spinning Acoustic Modes

    NASA Technical Reports Server (NTRS)

    Konno, Kevin E.; Hausmann, Clifford R.

    1996-01-01

    Rotating rake of pressure transducers developed for use in experimental studies of sources and propagation of noise generated by subsonic fan engines. Pressure transducers used as microphones to measure acoustic modes generated by, and spin with, fans. Versatility of control software used in rake-drive system enables measurements of acoustic modes on wide range of test-engine configurations. Rake-drive hardware easily adapted to different engines because not mechanically coupled to engine under test.

  11. 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.

  12. 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

  13. 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

  14. Theoretical response of condenser microphones

    NASA Technical Reports Server (NTRS)

    Zuckerwar, A. J.

    1978-01-01

    Modifications to prior theory yield expressions for the frequency response and equivalent lumped elements of a condenser microphone in terms of its fundamental geometrical and material properties. Results of the analysis show excellent agreement with experimental data taken on B&K pressure microphone types 4134 and 4146.

  15. 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

  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. 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).

  18. Source detection and high-resolution localization using microphone arrays for UGS: results of the NATO TG25 experiment measurements (Bourges, October 2002)

    NASA Astrophysics Data System (ADS)

    Hengy, Sebastien; Naz, Pierre; Gounon, Patrick

    2003-09-01

    This paper presents different ways to process acoustic data in order to localize targets.Beamforming and the MUSIC high resolution method have been tested for different propagation conditions during a NATO experimental campaign. This campaign,organized by DG /DCE/ETBS,has involved 6 countries in October 2002 in Bourges, France). Different localization methods were used to get the position of moving sources on a 4 kilometres circuit.The I.S.L. (French-German research institute of Saint Louis)has deployed a network of arrays nearby the circuit to test those localization techniques in different propagation conditions (day/night,early morning,...).Variance and mean error of the localization are compared for the different techniques used.

  19. Adaptive beamforming of a towed array during maneuvering

    NASA Astrophysics Data System (ADS)

    Gong, Zaixiao; Lin, Peng; Guo, Yonggang; Zhang, Renhe; Li, Fenghua

    2012-11-01

    During maneuvering, the performance of Minimum Variance Distortion-less Response (MVDR) beamforming for a towed hydrophone array will greatly degrade due to shape error. Under the assumption that the shape of a towed array changes in a known way during the observation interval, an improved MVDR method is proposed. A static array with average shape during the observation interval is taken as a reference array shape. The phase difference of the cross spectral density matrix (CSDM) between the time-varying array and the reference array is compensated on each azimuth. A coherent CSDM accumulation can then be achieved. Experimental results show that the improved MVDR method can yield better performance than conventional MVDR with a time-varying array. This helps to resolve the problems of left-right target ambiguity and weak signal detection for time-varying arrays.

  20. Microphones' directivity for the localization of sound sources

    NASA Astrophysics Data System (ADS)

    Rizzo, Piervincenzo; Tajari, Mahdi; Spada, Antonino

    2011-06-01

    In a recent paper [P. Rizzo, G. Bordoni, A. Marzani, and J. Vipperman, "Localization of Sound Sources by Means of Unidirectional Microphones, Meas. Sci. Tech., 20, 055202 (12pp), 2009] the proof-of-concept of an approach for the localization of acoustic sources was presented. The method relies on the use of unidirectional microphones and amplitude-based signals' features to extract information about the direction of the incoming sound. By intersecting the directions identified by a pair of microphones, the position of the emitting source can be identified. In this paper we expand the work presented previously by assessing the effectiveness of the approach for the localization of an acoustic source in an indoor setting. As the method relies on the accurate knowledge of the microphones directivity, analytical expression of the acoustic sensors polar pattern were derived by testing them in an anechoic chamber. Then an experiment was conducted in an empty laboratory by using an array of three unidirectional microphones. The ability to locate the position of a commercial speaker placed at different positions in the room is discussed. The objective of this study is to propose a valid alternative to the common application of spaced arrays and therefore to introduce a new generation of reduced size sound detectors and localizers. The ability of the proposed methodology to locate the position of a commercial speaker placed at different positions in the room was evaluated and compared to the accuracy provided by a conventional time delay estimate algorithm.

  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. A new approach to tackle noise issue in miniature directional microphones: bio-inspired mechanical coupling

    NASA Astrophysics Data System (ADS)

    Liu, Haijun; Yu, Miao

    2010-04-01

    When using microphone array for sound source localization, the most fundamental step is to estimate the time difference of arrival (TDOA) between different microphones. Since TDOA is proportional to the microphone separation, the localization performance degrades with decreasing size relative to the sound wavelength. To address the size constraint of conventional directional microphones, a new approach is sought by utilizing the mechanical coupling mechanism found in the superacute ears of the parasitic fly Ormia ochracea. Previously, we have presented a novel bio-inspired directional microphone consisting of two circular clamped membranes structurally coupled by a center pivoted bridge, and demonstrated both theoretically and experimentally that the fly ear mechanism is replicable in a man-made structure. The emphasis of this article is on theoretical analysis of the thermal noise floor of the bio-inspired directional microphones. Using an equivalent two degrees-of-freedom model, the mechanical-thermal noise limit of the structurally coupled microphone is estimated and compared with those obtained for a single omni-directional microphone and a conventional microphone pair. Parametric studies are also conducted to investigate the effects of key normalized parameters on the noise floor and the signal-to-noise ratio (SNR).

  3. Issues critical to the application of adaptive array antennas to missile seekers

    NASA Astrophysics Data System (ADS)

    Trapp, R. L.; Ronnenburg, C. H.

    1983-09-01

    Missile seekers will confront complex and hostile signal environments that can inhibit severely their ability to intercept threatening targets. Dramatic target detection and homing performance improvement in main beam and sidelobe jamming is realizable with a seeker antenna that can optimally adapt, in real time, its response to the signal environment. Adaptive array antennas can be designed to optimize the signal-to-interference-plus-noise ratio by forming pattern nulls directed toward sources of interference while simultaneously maximizing gain in the desired signal direction. Physical and operational missile constraints place severe requirements on an adaptive array. Nevertheless, there are several array configurations and adaptive processors that can satisfy these constraints in the next decade. Technology is a dominant limitation to adaptive array performance in a missile seeker. Signal processors and array implementations using state-of-the-art technology are required. Critical experimentation and representative simulations are needed to establish error effects, preferred adaptive array implementations, detailed requirements, and relative cost estimates. Although an adaptive missile seeker antenna is physically realizable in the next decade, the tradeoffs between cost, complexity, and performance will determine its utility and practicality.

  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. Injection monitoring with seismic arrays and adaptive noise cancellation

    SciTech Connect

    Harben, P.E.; Harris, D.B.; Jarpe, S.P.

    1991-01-01

    Although the application of seismic methods, active and passive, to monitor in-situ reservoir stimulation processes is not new, seismic arrays and array processing technology coupled with a new noise cancellation method has not been attempted. Successful application of seismic arrays to passively monitor in-situ reservoir stimulation processes depends on being able to sufficiently cancel the expected large amplitude background seismic noise typical of an oil or geothermal production environment so that small amplitude seismic signals occurring at depth can be detected and located. This report describes the results of a short field experiment conducted to test both the application of seismic arrays for in-situ reservoir stimulation monitoring and the active noise cancellation technique in a real reservoir production environment. Although successful application of these techniques to in-situ reservoir stimulation monitoring would have the greatest payoff in the oil industry, the proof-of-concept field experiment site was chosen to be the Geysers geothermal field in northern California. This site was chosen because of known high seismicity rates, a relatively shallow production depth, cooperation and some cost sharing the UNOCAL Oil Corporation, and the close proximity of the site to LLNL. The body of this report describes the Geysers field experimental configuration and then discusses the results of the seismic array processing and the results of the seismic noise cancellation followed by a brief conclusion. 2 refs., 11 figs.

  7. Ultra wideband photonic control of an adaptive phased array antenna

    NASA Astrophysics Data System (ADS)

    Cox, Joseph L.; Zmuda, Henry; Li, Jian; Sforza, Pasquale M.

    2006-05-01

    This paper presents a new concept for a photonic implementation of a time reversed RF antenna array beamforming system. The process does not require analog to digital conversion to implement and is therefore particularly suited for high bandwidth applications. Significantly, propagation distortion due to atmospheric effects, clutter, etc. is automatically accounted for with the time reversal process. The approach utilizes the reflection of an initial interrogation signal from off an extended target to precisely time match the radiating elements of the array so as to re-radiate signals precisely back to the target's location. The backscattered signal(s) from the desired location is captured by each antenna and used to modulate a pulsed laser. An electrooptic switch acts as a time gate to eliminate any unwanted signals such as those reflected from other targets whose range is different from that of the desired location resulting in a spatial null at that location. A chromatic dispersion processor is used to extract the exact array parameters of the received signal location. Hence, other than an approximate knowledge of the steering direction needed only to approximately establish the time gating, no knowledge of the target position is required, and hence no knowledge of the array element time delay is required. Target motion and/or array element jitter is automatically accounted for. This paper presents the preliminary study of the photonic processor, analytical justification, and simulated results. The technology has a broad range of applications including aerospace and defense and in medical imaging.

  8. 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.

  9. Adaptive passive fathometer processing using ambient noise received by vertical nested array

    NASA Astrophysics Data System (ADS)

    Kim, Junghun; Cho, Sungho; Choi, Jee Woong

    2015-07-01

    A passive fathometer technique utilizes surface-generated ambient noise received by a vertical line array as a sound source to estimate the depths of water-sediment interface and sub-bottom layers. Ambient noise was measured using a 24-channel, vertical nested line array consisting of four sub-arrays, in shallow water off the eastern coast of Korea. In this paper, nested array processing is applied to passive fathometer technique to improve the performance. Passive fathometer processing is performed for each sub-array, and the results are then combined to form a passive fathometer output for broadband ambient noise. Three types of beamforming technique, including conventional and two adaptive methods, are used in passive fathometer processing. The results are compared to the depths of water-sediment interface measured by an echo sounder. As a result, it is found that the adaptive methods have better performance than the conventional method.

  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. 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.

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

    PubMed

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

    2014-11-01

    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. PMID:25430255

  13. Adaptive array technique for differential-phase reflectometry in QUESTa)

    NASA Astrophysics Data System (ADS)

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

    2014-11-01

    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.

  14. 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

  15. Geophysical Inversion with Adaptive Array Processing of Ambient Noise

    NASA Astrophysics Data System (ADS)

    Traer, James

    2011-12-01

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

  16. 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.

  17. Resonant microphone based on laser beam deflection

    NASA Astrophysics Data System (ADS)

    Roark, Kevin; Diebold, Gerald J.

    2004-07-01

    A microphone consisting of a flexible membrane coupled to a Helmholtz resonator can be constructed to have a resonance at a specific frequency making it, unlike conventional broadband microphones, a frequency selective detector of sound. The present device uses a laser beam reflected from the membrane and directed onto a split photodiode to record the motion of the membrane. Since the microphone has a lightly damped resonance, both the thermal noise fluctuations in the displacement of the membrane from its equilibrium position and the response of the microphone to sound at the resonance frequency are large. The large amplitude of both the signal and the noise fluctuations means that effect of amplifier noise on the microphone's sensitivity is diminished relative to that in broadband microphones. Applications of the microphone include photoacoustic detection of gases employing low power lasers.

  18. A simulation study of jammer nulling trade-offs in a reactively steered adaptive array

    NASA Astrophysics Data System (ADS)

    Dinger, R. J.

    1985-02-01

    Antenna arrays that operate at frequencies up to 6 GHz or so on air-launched guided missiles are necessarily compact because of limited space. Research has been in progress since FY82 on a compact adaptive array that uses reactively loaded parasitic elements for pattern control. This report describes a study of this class of array whose purpose is to examine the trade-offs available among number of elements, element spacing, and number of nullable jammers. The research is part of a continuing effort to explore novel radio frequency radiating and receiving structures for application to airborne communications and radar system.

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

    NASA Technical Reports Server (NTRS)

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

    2006-01-01

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

  20. 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.

  1. 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.

  2. Effects of in-the-ear microphone directionality on sound direction identification.

    PubMed

    Chung, King; Neuman, Arlene C; Higgins, Michael

    2008-04-01

    As advanced signal processing algorithms have been proposed to enhance hearing protective device (HPD) performance, it is important to determine how directional microphones might affect the localization ability of users and whether they might cause safety hazards. The effect of in-the-ear microphone directivity was assessed by measuring sound source identification of speech in the horizontal plane. Recordings of speech in quiet and in noise were made with Knowles Electronic Manikin for Acoustic Research wearing bilateral in-the-ear hearing aids with microphones having adjustable directivity (omnidirectional, cardioid, hypercardioid, supercardioid). Signals were generated from 16 locations in a circular array. Sound direction identification performance of eight normal hearing listeners and eight hearing-impaired listeners revealed that directional microphones did not degrade localization performance and actually reduced the front-back and lateral localization errors made when listening through omnidirectional microphones. The summed rms speech level for the signals entering the two ears appear to serve as a cue for making front-back discriminations when using directional microphones in the experimental setting. The results of this study show that the use of matched directional microphones when worn bilaterally do not have a negative effect on the ability to localize speech in the horizontal plane and may thus be useful in HPD design. PMID:18397031

  3. 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.

  4. 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

  5. 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.

  6. The design and operation of infrasonic microphones

    SciTech Connect

    Mutschlecner, J.P.; Whitaker, R.W.

    1997-05-01

    This report is intended as a guide to the design and operation of infrasonic microphones. It will emphasize general principles and the effects of parameter choices upon performance but will not provide details of design for specific microphones. The report will consider primarily the mechanical aspects that control the acoustic properties; it will not discuss the features of electronic design, which vary greatly among microphones. Here the authors define infrasonic microphones as sensors capable of detecting pressure variations in the period range of about 0.1 s to 1000 s with changes from about 0.1 {mu}bar (microbar = 1 dyne cm{sup {minus}2}) to about 1 mbar.

  7. Application of adaptive optics to scintillation correction in phased array high-frequency radar

    NASA Astrophysics Data System (ADS)

    Theurer, Timothy E.; Bristow, William A.

    2015-06-01

    At high frequency, diffraction during ionospheric propagation can yield wavefronts whose amplitude and phase fluctuate over the physical dimensions of phased array radars such as those of the Super Dual Auroral Radar Network (SuperDARN). Distortion in the wavefront introduces amplitude and phase scintillation into the geometric beamformed signal while reducing radar performance in terms of angular resolution and achieved array gain. A scintillation correction algorithm based on adaptive optics techniques is presented. An experiment conducted using two SuperDARN radars is presented that quantifies the effect of wavefront distortion and demonstrates a reduction in observed scintillation and improvement in radar performance post scintillation correction.

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

    NASA Technical Reports Server (NTRS)

    Ward, James

    1987-01-01

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

  9. Two-way unidirectional condenser microphone

    NASA Astrophysics Data System (ADS)

    Morita, A.; Mizoguchi, A.

    1984-03-01

    A microphone for digital use is described which has a flat frequency-response over a wide range, from 20 Hz to 20 kHz; high sensitivity; and a unidirectional pattern over the whole frequency range. The principal ratings of the microphone are tabulated.

  10. Localization of sound sources by means of unidirectional microphones

    NASA Astrophysics Data System (ADS)

    Rizzo, Piervincenzo; Bordoni, Giacomo; Marzani, Alessandro; Vipperman, Jeffrey

    2009-05-01

    This paper describes the results of a new approach devoted to the localization of ground borne acoustic sources. It is demonstrated that an array made of at least three unidirectional microphones can be exploited to identify the position of the source. Sound features extracted either in the time domain or in the frequency domain are used to localize the direction of the incoming sound. This information is then fed into a semi-analytical algorithm aimed at identifying the source location. The novelty of the method presented here consists in the use of unidirectional microphones rather than omnidirectional microphones and in the ability to extract the sound direction by considering features like sound amplitude rather than the time of arrival. Experimental tests have been undertaken in a closed environment and have demonstrated the feasibility of the proposed approach. It is believed that this method may pave the road toward a new generation of reduced-size sound detectors and localizers, and future work is described in the conclusions.

  11. 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. PMID:22163859

  12. Ultrasound nondestructive evaluation (NDE) imaging with transducer arrays and adaptive processing.

    PubMed

    Li, Minghui; Hayward, Gordon

    2012-01-01

    This paper addresses the challenging problem of ultrasonic non-destructive evaluation (NDE) imaging with adaptive transducer arrays. In NDE applications, most materials like concrete, stainless steel and carbon-reinforced composites used extensively in industries and civil engineering exhibit heterogeneous internal structure. When inspected using ultrasound, the signals from defects are significantly corrupted by the echoes form randomly distributed scatterers, even defects that are much larger than these random reflectors are difficult to detect with the conventional delay-and-sum operation. We propose to apply adaptive beamforming to the received data samples to reduce the interference and clutter noise. Beamforming is to manipulate the array beam pattern by appropriately weighting the per-element delayed data samples prior to summing them. The adaptive weights are computed from the statistical analysis of the data samples. This delay-weight-and-sum process can be explained as applying a lateral spatial filter to the signals across the probe aperture. Simulations show that the clutter noise is reduced by more than 30 dB and the lateral resolution is enhanced simultaneously when adaptive beamforming is applied. In experiments inspecting a steel block with side-drilled holes, good quantitative agreement with simulation results is demonstrated. PMID:22368457

  13. 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%.

  14. 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.

  15. 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.

  16. 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.

  17. 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.

  18. Microphones for speech and speech recognition

    NASA Astrophysics Data System (ADS)

    West, James E.

    2004-10-01

    Automatic speech recognition (ASR) requires about a 15- to 20-dB signal-to-noise ratio (S/N) for high accuracy even for small vocabulary systems. This S/N is generally achievable using a telephone handset in normal office or home environments. In the early 1990s ATT and the regional telephone companies began using speaker-independent ASR to replace several operator services. The variable distortion in the carbon microphone was not transparent and resulted in reduced ASR accuracy. The linear electret condenser microphone, common in most modern telephones, improved handset performance both in sound quality and ASR accuracy. Hands-free ASR in quiet conditions is a bit more complex because of the increased distance between the microphone and the speech source. Cardioid directional microphones offer some improvement in noisy locations when the noise and desired signals are spatially separated, but this is not the general case and the resulting S/N is not adequate for seamless machine translation. Higher-order directional microphones, when properly oriented with respect to the talker and noise, have shown good improvement over omni-directional microphones. Some ASR results measured in simulated car noise will be presented.

  19. Understanding fly-ear inspired directional microphones

    NASA Astrophysics Data System (ADS)

    Liu, Haijun; Zhang, Xuming; Yu, Miao

    2009-03-01

    In this article, the equivalent two-degree-of-freedom (2-DOF) model for the hypersensitive ear of fly Ormia ocharacea is revisited. It is found that in addition to the mechanical coupling between the ears, the key to the remarkable directional hearing ability of the fly is the proper contributions of the rocking mode and bending mode of the ear structure. This can serve as the basis for the development of fly-ear inspired directional microphones. New insights are also provided to establish the connection between the mechanics of the fly ear and the prior biological experiments, which reveals that the fly ear is a nature-designed optimal structure that might have evolved to best perform its localization task at 5 kHz. Based on this understanding, a new design of the fly-ear inspired directional microphone is presented and a corresponding normalized continuum mechanics model is derived. Parametric studies are carried out to study the influence of the identified non-dimensional parameters on the microphone performance. Directional microphones are developed to verify the understanding and concept. This study provides a theoretical guidance to develop miniature bio-inspired directional microphones, and can impact many fronts that require miniature directional microphones.

  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. 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

  2. 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.

  3. Wide-bandwidth silicon nitride membrane microphones

    NASA Astrophysics Data System (ADS)

    Cunningham, Brian T.; Bernstein, Jonathan J.

    1997-09-01

    Small, low cost microphones with high sensitivity at frequencies greater than 20 KHz are desired for applications such as ultrasonic imaging and communication links. To minimize stray capacitance between the microphone and its amplifier circuit, process compatibility between the microphone and on-chip circuitry is also desired to facilitate integration. In this work, we have demonstrated micromachined microphones packaged with hybrid JFET amplifier circuitry with frequency response extending to 100 KHz, and voltage sensitivity of approximately 2.0 mV/Pa from 100 Hz to 10 KHz, and 16.5 mV/Pa at 30 KHz with a bias voltage of 8.0 V. The microphones are fabricated with membranes and fixed backplates made of low temperature plasma-enhanced chemical vapor deposited (PECVD) silicon nitride. Because the maximum temperature of the fabrication process is 300 degrees Celsius, microphones may be built on silicon wafers from any commercial CMOS foundry without affecting transistor characteristics, allowing integration with sophisticated amplifier circuitry. Low stress silicon nitride deposition was used to produce membranes up to 2.0 mm diameter and 0.5 micrometer thickness with plus or minus 0.10 micrometer flatness. The excellent planarity of both the diaphragm and the backplate, combined with a narrow sense gap (approximately 2 micrometers) results in high output capacitance (up to 6.0 pF). The high output capacitance results in noise spectral density which is approximately 3x lower than silicon diaphragms microphones previously fabricated by the authors. Diaphragms with corrugations were fabricated to relive tensile stress, to increase deflection per unit pressure and to increase deflection linearity with pressure.

  4. Extending a context model for microphone forensics

    NASA Astrophysics Data System (ADS)

    Kraetzer, Christian; Qian, Kun; Dittmann, Jana

    2012-03-01

    In this paper, we extend an existing context model for statistical pattern recognition based microphone forensics by: first, generating a generalized model for this process and second, using this general model to construct a complex new application scenario model for microphone forensic investigations on the detection of playback recordings (a.k.a. replays, re-recordings, double-recordings). Thereby, we build the theoretical basis for answering the question whether an audio recording was made to record a playback or natural sound. The results of our investigations on the research question of playback detection imply that it is possible with our approach on our evaluation set of six microphones. If the recorded sound is not modified prior to playback, we achieve in our tests 89.00% positive indications on the correct two microphones involved. If the sound is post-processed (here, by normalization) this figure decreases (in our normalization example to 36.00%, while another 50.67% of the tests still indicate two microphones, of which one has actually not been involved in the recording and playback recording process).

  5. Effects of microphone type on acoustic measures of voice.

    PubMed

    Parsa, V; Jamieson, D G; Pretty, B R

    2001-09-01

    Acoustic measures provide an objective means to describe pathological voices and are a routine component of the clinical voice examination. Because the voice sample is obtained using a microphone, microphone characteristics have the potential to influence the values of parameters obtained from a voice sample. This project examined how the choice of microphone affects key voice parameters and investigated how one might compensate for such microphone effects through filtering or by including additional parameters in the decision process. A database of 53 normal voice samples and 100 pathological voice samples was used in four experiments conducted in an anechoic chamber using four different microphones. One omnidirectional microphone and three cardioid microphones were used in these experiments. The original voice samples were presented to each microphone through a speaker located in an anechoic chamber, and the output of each microphone sampled to computer disk. Each microphone modified the frequency spectrum of the voice signal; this, in turn, affected the values of the voice parameters obtained. These microphone effects reduced the accuracy with which acoustic measures of voice could be used to discriminate pathological from normal voices. Discrimination performance improved when the microphone output was filtered to compensate for microphone frequency response. Performance also improved when spectral moment coefficient parameters were added to the vocal function parameters already in use. PMID:11575630

  6. 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.

  7. Microphone interlaboratory comparison in the Americas

    NASA Astrophysics Data System (ADS)

    Wong, George S. K.; Wu, Lixue

    2002-11-01

    The final results of a Sistema Interamericano de Metrologia (SIM) interlaboratory comparison on microphone calibration are presented. Initially the comparison involved NORAMET countries: USA, Canada, and Mexico. Later, the comparison was extended to include Argentina and Brazil, resulting in a SIM AUV.A-K1 microphone interlaboratory comparison. The National Metrology Institutes (NMIs) of the five American countries that participated were the Institute for National Measurement Standards (INMS--Canada), National Institute of Standards and Technology (NIST--USA), Centro Nacional de Metrologia (CENAM--Mexico), Instituto Nacional de Metrologia, Normalizacao e Qualidade Industrial (INMETRO--Brazil) and Unidad Tecnica Acustica, (INTI--Argentina). INMS, Canada was the pilot laboratory that provided the data for the final report. The maximum rms deviation for the two LS1P laboratory standard microphones measured by the above participants is 0.037 dB that may be considered as the key comparison reference value.

  8. 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.

  9. 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.

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

    PubMed

    Karch, Barry K; Hardie, Russell C

    2013-08-12

    Digital color cameras using a single detector array with a Bayer color filter array (CFA) require interpolation or demosaicing to estimate missing color information and provide full-color images. However, demosaicing does not specifically address fundamental undersampling and aliasing inherent in typical camera designs. Fast non-uniform interpolation based super-resolution (SR) is an attractive approach to reduce or eliminate aliasing and its relatively low computational load is amenable to real-time applications. The adaptive Wiener filter (AWF) SR algorithm was initially developed for grayscale imaging and has not previously been applied to color SR demosaicing. Here, we develop a novel fast SR method for CFA cameras that is based on the AWF SR algorithm and uses global channel-to-channel statistical models. We apply this new method as a stand-alone algorithm and also as an initialization image for a variational SR algorithm. This paper presents the theoretical development of the color AWF SR approach and applies it in performance comparisons to other SR techniques for both simulated and real data. PMID:23938797

  11. Performance of directional microphone hearing aids in everyday life.

    PubMed

    Cord, Mary T; Surr, Rauna K; Walden, Brian E; Olson, Laurel

    2002-06-01

    This study explored the use patterns and benefits of directional microphone technology in real-world situations experienced by patients who had been fitted with switchable omnidirectional/directional hearing aids. Telephone interviews and paper-and-pencil questionnaires were used to assess perceived performance with each microphone type in a variety of listening situations. Patients who used their hearing aids regularly and switched between the two microphone configurations reported using the directional mode, on average, about one-quarter of the time. From brief descriptions, patients could identify listening situations in which each microphone mode should provide superior performance. Further, they reported encountering listening situations in which an omnidirectional microphone should provide better performance more frequently than listening situations in which the directional microphones should be superior. Despite using the omnidirectional mode more often and encountering situations in which an omnidirectional microphone should provide superior performance more frequently, participants reported the same level of satisfaction with each microphone type. PMID:12141387

  12. A bidirectional microphone for the measurement of duct noise

    NASA Astrophysics Data System (ADS)

    La Fontaine, R. F.; Shepherd, I. C.; Cabelli, A.

    1985-08-01

    A bidirectional microphone which resolves acoustic plane waves in ducts into forward and backward propagating components is described. The microphone has a flat frequency response and finds applications in the analysis of duct noise and in the determination of reflection coefficients for various duct configurations. It can also be employed as a unidirectional microphone in active noise attenuators.

  13. 77 FR 64446 - Wireless Microphones Proceeding

    Federal Register 2010, 2011, 2012, 2013, 2014

    2012-10-22

    ... Docket No. 02-380, Second Memorandum Opinion and Order, 75 FR 75814, 25 FCC Rcd 18661 (2010) (TV White... Rulemaking, 75 FR 3622, 75 FR 3682, 25 FCC Rcd 643 (2010) (Wireless Microphones Order and Wireless... Memorandum Opinion and Order, 77 FR 29236, 27 FCC Rcd 3692 (2012). Background In the Wireless...

  14. 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.

  15. System Measures Thermal Noise In A Microphone

    NASA Technical Reports Server (NTRS)

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

    1994-01-01

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

  16. A Simple Laser Microphone for Classroom Demonstration

    NASA Astrophysics Data System (ADS)

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

    2006-12-01

    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 described.

  17. 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.

  18. 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…

  19. 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.

  20. Use of unidirectional microphones and signal processing for the localization of sound sources

    NASA Astrophysics Data System (ADS)

    Rizzo, Piervincenzo; Bordoni, Giacomo; Marzani, Alessandro

    2009-05-01

    Targeting people or objects by passive acoustic sensors is of relevant interest in several military and civil applications, spanning from surveillance and patrolling systems to teleconferencing and human-robot interaction. To date methods and patents focused solely on the use of beamforming algorithms to compute the time of arrival of sounds detected by using omnidirectional microphones (OM) sparsely deployed. This paper describes the preliminary results of a novel approach devoted to the localization of ground borne acoustic sources. It is demonstrated that an array made of at least three unidirectional microphones can be exploited to detect the position the source. Pulse features extracted either in the time domain or in the frequency domain are used to identify the direction of the incoming sound. This information is then fed into a semi-analytical algorithm devoted to the identification of the source location. The novelty of the method presented here consists on the use of unidirectional microphones rather than omnidirectional microphones and on the ability to extract the sound direction by considering features like the pulse amplitude rather than the pulse arrival time. It is believed that this method may pave the road toward a new generation of reduced size sound detectors and localizers.

  1. 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

  2. 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

  3. 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

  4. 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.

  5. Noise reduction using a multimicrophone array for automatic speech recognition on a handheld computer

    NASA Astrophysics Data System (ADS)

    Shaw, Scott T.; Larow, Andrew J.; Schoenborn, William E.; Rodriguez, Jason; Gibian, Gary L.

    2001-05-01

    A four-microphone array and signal-processing card have been integrated with a handheld computer such that the integrated device can be carried in and operated with one hand. Automatic speech recognition (ASR) was added to the USAMRMC/TATRCs Battlefield Medical Information System (BMIST) software using an approach that does not require modifying the original code, to produce a Speech-Capable Personal Digital Assistant (SCPDA). Noise reduction was added to allow operation in noisier environments, using the previously reported Hybrid Adaptive Beamformer (HAB) algorithm. Tests demonstrated benefits of the array over the HP/COMPAQ-iPAQ built-in shielded microphone for noise reduction and automatic speech recognition. In electroacoustic and human testing including voice control and voice annotation, the array provided substantial benefit over the built-in microphone. The benefit varied from about 5 dB (worst-case scenario, diffuse noise) to about 20 dB (best-case scenario, directional noise). Future work is expected to produce more rugged SCPDA prototypes for user evaluations, revise the design based on user feedback and real-world testing, and possibly to allow hands-free use by using ASR to replace the push-to-talk switch, providing feedback aurally and/or via a head-up display. [Work supported by the U.S. Army Medical Research and Materiel Command (USAMRMC), Contract No. DAMD17-02-C-0112.

  6. 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

  7. 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

  8. 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.

  9. Laser schlieren microphone for optoacoustic spectroscopy.

    PubMed

    Choi, J G; Diebold, G J

    1982-11-15

    This paper describes a laser schlieren microphone, where a low-power He-Ne laser beam is deflected by a reflecting diaphragm mounted on an optoacoustic Helmholtz resonator. The sinusoidal pressure variations in the resonator distort the surface of the diaphragm so that the reflected laser beam is alternately focused and defocused. The deflection is converted into an amplitude modulation of the beam by an iris located at a distance from the resonator and detected with a photodiode. The light beam can be modulated at a high frequency and the signal from the photodiode processed with a lock-in amplifier so that noise with a power spectral density proportional to the inverse of the frequency is significantly reduced in the final optoacoustic signal. A mathematical description of the laser schlieren microphone is given that shows the system to respond linearly to small signals. An experiment was done to determine the range of linear response of the microphone to large amplitude optoacoustic signals. PMID:20401014

  10. 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).

  11. Effect of microphone type and placement on voice perturbation measurements.

    PubMed

    Titze, I R; Winholtz, W S

    1993-12-01

    This study was conducted to explore the effects of microphone type (dynamic vs. condenser) and pattern (omnidirectional vs. cardioid) on the extraction of voice perturbation measures for sustained phonation. Also of interest were the effects of distance and angle between the source and the microphone. Four professional-grade and two consumer-grade microphones were selected for analysis. Synthesized phonation with different amplitude and frequency modulations at fundamental frequencies of 100 Hz and 300 Hz were presented over a loudspeaker. Human phonation was also included to test the validity of loudspeaker presentations. Three microphone distances (4 cm, 30 cm, 1 m) and three angles (0 degree, 45 degrees, 90 degrees) were used for microphone placement. Among the professional grade microphones, the cardioid condenser type had the smallest effect on perturbation measures. In general, condenser types gave better results than dynamic types. Microphones with an unbalanced output did not perform as well as those with balanced outputs. Microphone sensitivity and distance had the largest effect on perturbation measures, making it difficult to resolve normal vocal jitter at anything but a few centimeters from the mouth. Angle had little effect for short distances, but a greater effect for longer distances. These conclusions are preliminary because the sampling of microphones, distances, and signal types was very coarse. The study serves only to chart the course for future work. PMID:8114484

  12. Condenser Microphone Protective Grid Correction for High Frequency Measurements

    NASA Technical Reports Server (NTRS)

    Lee, Erik; Bennett, Reginald

    2010-01-01

    Use of a protective grid on small diameter microphones can prolong the lifetime of the unit, but the high frequency effects can complicate data interpretation. Analytical methods have been developed to correct for the grid effect at high frequencies. Specifically, the analysis pertains to quantifying the microphone protective grid response characteristics in the acoustic near field of a rocket plume noise source. A frequency response function computation using two microphones will be explained. Experimental and instrumentation setup details will be provided. The resulting frequency response function for a B&K 4944 condenser microphone protective grid will be presented, along with associated uncertainties

  13. Biomimetic optical directional microphone with structurally coupled diaphragms

    NASA Astrophysics Data System (ADS)

    Liu, H. J.; Yu, M.; Zhang, X. M.

    2008-12-01

    A biomimetic directional microphone based on structurally coupled diaphragms and a fiber-optic detection system is presented. The microphone design aims to mimic the fly Ormia Ochracea's ear structure and capture its performance. Experiments show that the designed microphone amplifies the interaural time difference (ITD) by 4.4 times and has a directional sensitivity of 6.5 μs/deg. An important finding is that one needs to utilize both the rocking and translational vibration modes to obtain the appropriate ITD amplification without sacrifice of directional sensitivity. This work can serve as a foundation for realizing fly-ear inspired miniature directional microphones.

  14. A silicon condenser microphone: Modelling and electronic circuitry

    NASA Astrophysics Data System (ADS)

    Vanderdonk, Armand Gijsbertus H.

    1992-01-01

    The operational mechanism of condenser microphones and some aspects concerning microphone terminology are described. A model of condenser microphones, concerning microphones with a highly tensioned diaphragm as well as microphones with a diaphragm without any initial stress, is presented. Expressions of the mechanical, the electrical, and the total sensitivity are derived. Some practical limitations concerning the influence of temperature changes during operation and the maximum allowable sound pressure are described. An optimization procedure with respect to the sensitivity, which takes into account the maximum allowable sound pressure, is presented. The noise contribution of the microphone preamplifier is considered. Amplifiers, consisting of a MOSFET and a bias resistor or bias diode, and amplifiers, consisting of a JFET and a bias resistor, are analyzed. The integration of a silicon condenser microphone with a PMOS source follower is addressed. Design criteria concerning the integration of the CMOS process and a specific microphone fabrication process are presented. The principle of electromechanical feedback is described. The electrical and mechanical problems and the limitations of feedback are discussed. The feedback was analyzed and tested for two types of actuator signals. The principle of operation of a DC voltage converter as a substitution of a low voltage electret is described. Experimental results of a converter realized in CMOS and the practical operation in combination with a silicon condenser microphone are presented.

  15. 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

  16. 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

  17. Wind noise at microphones within and across hearing aids at wind speeds below and above microphone saturation.

    PubMed

    Zakis, Justin A

    2011-06-01

    The variation of wind noise at hearing-aid microphones with wind speed, wind azimuth, and hearing-aid style was investigated. Comparisons were made across behind-the-ear (BTE) and completely-in-canal (CIC) devices, and between microphones within BTE devices. One CIC device and two BTE devices were placed on a Knowles Electronics Manikin for Acoustic Research. The smaller BTE device had vented plastic windshields around its microphone ports while the larger BTE device had none. The microphone output signals were digitally recorded in wind generated at 0, 3, 6, and 12 m/s at 8 wind azimuths. The microphone output signals were saturated at 12 m/s with wind-noise levels of up to 116 dB SPL at the microphone output. Wind-noise levels differed by up to 12 dB between microphones within the same BTE device, and across BTE devices by up to 6 or 8 dB for front or rear microphones, respectively. On average, wind-noise levels were lowest with the CIC device and highest at the rear microphone of the smaller BTE device. Engineering and clinical implications are discussed. PMID:21682412

  18. 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...

  19. 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...

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

    PubMed

    Oreinos, Chris; Buchholz, Jörg M

    2015-06-01

    Recently, an increased interest has been demonstrated in evaluating hearing aids (HAs) inside controlled, but at the same time, realistic sound environments. A promising candidate that employs loudspeakers for realizing such sound environments is the listener-centered method of higher-order ambisonics (HOA). Although the accuracy of HOA has been widely studied, it remains unclear to what extent the results can be generalized when (1) a listener wearing HAs that may feature multi-microphone directional algorithms is considered inside the reconstructed sound field and (2) reverberant scenes are recorded and reconstructed. For the purpose of objectively validating HOA for listening tests involving HAs, a framework was developed to simulate the entire path of sounds presented in a modeled room, recorded by a HOA microphone array, decoded to a loudspeaker array, and finally received at the ears and HA microphones of a dummy listener fitted with HAs. Reproduction errors at the ear signals and at the output of a cardioid HA microphone were analyzed for different anechoic and reverberant scenes. It was found that the diffuse reverberation reduces the considered time-averaged HOA reconstruction errors which, depending on the considered application, suggests that reverberation can increase the usable frequency range of a HOA system. PMID:26093433

  1. 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.

  2. 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.

  3. 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.

  4. Speech intelligibility enhancement using hearing-aid array processing.

    PubMed

    Saunders, G H; Kates, J M

    1997-09-01

    Microphone arrays can improve speech recognition in the noise for hearing-impaired listeners by suppressing interference coming from other than desired signal direction. In a previous paper [J. M. Kates and M. R. Weiss, J. Acoust. Soc. Am. 99, 3138-3148 (1996)], several array-processing techniques were evaluated in two rooms using the AI-weighted array gain as the performance metric. The array consisted of five omnidirectional microphones having uniform 2.5-cm spacing, oriented in the endfire direction. In this paper, the speech intelligibility for two of the array processing techniques, delay-and-sum beamforming and superdirective processing, is evaluated for a group of hearing-impaired subjects. Speech intelligibility was measured using the speech reception threshold (SRT) for spondees and speech intelligibility rating (SIR) for sentence materials. The array performance is compared with that for a single omnidirectional microphone and a single directional microphone having a cardioid response pattern. The SRT and SIR results show that the superdirective array processing was the most effective, followed by the cardioid microphone, the array using delay-and-sum beamforming, and the single omnidirectional microphone. The relative processing ratings do not appear to be strongly affected by the size of the room, and the SRT values determined using isolated spondees are similar to the SIR values produced from continuous discourse. PMID:9301060

  5. Evaluation of an adaptive beamforming method for hearing aids.

    PubMed

    Greenberg, J E; Zurek, P M

    1992-03-01

    In this paper evaluations of a two-microphone adaptive beamforming system for hearing aids are presented. The system, based on the constrained adaptive beamformer described by Griffiths and Jim [IEEE Trans. Antennas Propag. AP-30, 27-34 (1982)], adapts to preserve target signals from straight ahead and to minimize jammer signals arriving from other directions. Modifications of the basic Griffiths-Jim algorithm are proposed to alleviate problems of target cancellation and misadjustment that arise in the presence of strong target signals. The evaluations employ both computer simulations and a real-time hardware implementation and are restricted to the case of a single jammer. Performance is measured by the spectrally weighted gain in the target-to-jammer ratio in the steady state. Results show that in environments with relatively little reverberation: (1) the modifications allow good performance even with misaligned arrays and high input target-to-jammer ratios; and (2) performance is better with a broadside array with 7-cm spacing between microphones than with a 26-cm broadside or a 7-cm endfire configuration. Performance degrades in reverberant environments; at the critical distance of a room, improvement with a practical system is limited to a few dB. PMID:1564202

  6. 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.

  7. Testing and characterization of second-order differential microphones

    NASA Astrophysics Data System (ADS)

    Merlis, Joshua Howard

    The focus of this thesis is the testing and characterizing of directional microphones, designed based on the ear of the fly Ormia ochracea. The response of these microphones is modeled as a linear combination of the gradients of the sound field. A least squares approach is employed in order to determine the transfer functions between the response and these gradients. Knowledge of these complex transfer functions is crucial in understanding the nature and quality of the response of these microphones. Once determined, these transfer functions are used to simulate the plane wave response of differential microphones. This process is invaluable to acoustic research groups that do not have access to an anechoic chamber because the plane wave response is a standard by which acoustic devices are measured. This process was validated by comparing the true plane wave response of an industry standard differential microphone with its simulated plane wave response.

  8. In situ tuning of a MEMS microphone using electrodeposited nanostructures

    NASA Astrophysics Data System (ADS)

    Je, Sang-Soo; Harrison, Jere C.; Kozicki, Michael N.; Bakkaloglu, Bertan; Kiaei, Sayfe; Chae, Junseok

    2009-03-01

    This paper presents a new method for in situ tuning of acoustic sensitivity in micro-electro-mechanical-system (MEMS) microphones using silver metallic nano-electrodeposits. The nano-electrodeposits are electrochemically formed using an external dc bias under low power and at room temperature on an Ag-doped Ge30Se70 solid electrolyte film integrated with the microphone diaphragm. The growth/retraction mechanism generates mass/stress redistribution on the diaphragm and this effect is used to manipulate microphone sensitivity to incoming acoustic waves. Acoustic measurements with a reference microspeaker demonstrate that the microphone can achieve a tuning range of 0.6 dB (7.2%). This technique is useful for a variety of microdevice applications, including sensitivity matching for directional microphones (e.g., in hearing aids), post-package trimming and resonant frequency tuning.

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

    NASA Astrophysics Data System (ADS)

    Won, Yong Hyub

    2014-10-01

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

  10. 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

  11. 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.

  12. Biologically inspired MEMS based directional microphone

    NASA Astrophysics Data System (ADS)

    Touse, Michael; Harrison, Stephen; Catterlin, Jeffrey; Karunasiri, Gamani

    2009-11-01

    A novel MEMS microphone is presented which mimics the aural system of the Ormia ochracea fly and its extraordinary directional sensitivity. To overcome the minimal separation between its ears, a flexible hinge mechanically couples the fly's two tympanic membranes. By comparing the frequency response of these two structures, the interaural differences are amplified and sound source information is processed with unparalleled speed and accuracy. The presented device is 2mm x 1mm x 10μm SOI, hinged at the middle and attached to the substrate using two narrow legs, allowing both rocking and bending modes. Along the edges of the membrane, two sets of interdigitated comb fingers are connected to an Irvine Sensors capacitive readout chip to allow electronic measurement of the displacement. Also presented are results of extensive finite element modeling performed using COMSOL Multiphysics, which are in close agreement with experimental data.

  13. 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

  14. Array measurements adapted to the number of available sensors: Theoretical and practical approach for ESAC method

    NASA Astrophysics Data System (ADS)

    Galiana-Merino, J. J.; Rosa-Cintas, S.; Rosa-Herranz, J.; Garrido, J.; Peláez, J. A.; Martino, S.; Delgado, J.

    2016-05-01

    Array measurements of ambient noise have become a useful technique to estimate the surface wave dispersion curves and subsequently the subsurface elastic parameters that characterize the studied soil. One of the logistical handicaps associated with this kind of measurements is the requirement of several stations recording at the same time, which limits their applicability in the case of research groups without enough infrastructure resources. In this paper, we describe the theoretical basis of the ESAC method and we deduce how the number of stations needed to implement any array layout can be reduced to only two stations. In this way, we propose a new methodology to implement an N stations array layout by using only M stations (M < N), which will be recording in different positions of the original prearranged N stations geometry at different times. We also provide some practical guidelines to implement the proposed approach and we show different examples where the obtained results confirm the theoretical foundations. Thus, the study carried out reflects that we can use a minimum of 2 stations to deploy any array layout originally designed for higher number of sensors.

  15. 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

  16. 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

  17. Refined acoustic modeling and analysis of shotgun microphones.

    PubMed

    Bai, Mingsian R; Lo, Yi-Yang

    2013-04-01

    A shotgun microphone is a highly directional pickup device widely used in noisy environments. The key element that leads to its superior directivity is a tube with multiple slot openings along its length. One traditional way to model the directional response of a shotgun is to assume plane waves traveling in the tube as if it is in the free field. However, the frequency response and directivity predicted by this traveling wave model can differ drastically from practical measurements. In this paper, an in-depth electroacoustic analysis was conducted to examine the problem by considering the standing waves inside the tube with an analogous circuit containing phased pressure sources and T-networks of tube segments. A further refinement is to model the housing diffraction effect with the aid of the equivalent source method (ESM). The on-axis frequency response and directivity pattern predicted by the proposed model are in close agreement with the measurements. From the results, a peculiar bifurcation phenomenon of directivity pattern at the Helmholtz frequency was also noted. While the shotgun behaves like an endfire array above the Helmholtz frequency, it becomes a broadside array below the Helmholtz frequency. The standing wave effect can be mitigated by covering the slot openings with mesh screen, which was found to alter the shotgun response to be closer to that of the traveling wave model above a critical frequency predicted by the half-wavelength rule. A mode-switching model was developed to predict the directional responses of mesh-treated shotguns. PMID:23556574

  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. 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.

  1. 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.

  2. 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.

  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. Fast iterative adaptive nonuniformity correction with gradient minimization for infrared focal plane arrays

    NASA Astrophysics Data System (ADS)

    Zhao, Jufeng; Gao, Xiumin; Chen, Yueting; Feng, Huajun; Xu, Zhihai; Li, Qi

    2014-07-01

    A fast scene-based nonuniformity correction algorithm is proposed for fixed-pattern noise removal in infrared focal plane array imagery. Based on minimization of L0 gradient of the estimated irradiance, the correction function is optimized through correction parameters estimation via iterative optimization strategy. When applied to different real IR data, the proposed method provides enhanced results with good visual effect, making a good balance between nonuniformity correction and details preservation. Comparing with other excellent approaches, this algorithm can accurately estimate the irradiance rapidly with fewer ghosting artifacts.

  5. 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.

  6. 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.

  7. Computationally Efficient Locally Adaptive Demosaicing of Color Filter Array Images Using the Dual-Tree Complex Wavelet Packet Transform

    PubMed Central

    Aelterman, Jan; Goossens, Bart; De Vylder, Jonas; Pižurica, Aleksandra; Philips, Wilfried

    2013-01-01

    Most digital cameras use an array of alternating color filters to capture the varied colors in a scene with a single sensor chip. Reconstruction of a full color image from such a color mosaic is what constitutes demosaicing. In this paper, a technique is proposed that performs this demosaicing in a way that incurs a very low computational cost. This is done through a (dual-tree complex) wavelet interpretation of the demosaicing problem. By using a novel locally adaptive approach for demosaicing (complex) wavelet coefficients, we show that many of the common demosaicing artifacts can be avoided in an efficient way. Results demonstrate that the proposed method is competitive with respect to the current state of the art, but incurs a lower computational cost. The wavelet approach also allows for computationally effective denoising or deblurring approaches. PMID:23671575

  8. The virtual microphone technique in active sound field control systems

    NASA Astrophysics Data System (ADS)

    Lampropoulos, Iraklis E.; Shimizu, Yasushi

    2003-04-01

    Active Sound Field Control (AFC) has been proven very useful in reverberation enhancement applications in large rooms. However, feedback control is required in order to eliminate peaks in the frequency response of the system. The present research closely follows the studies of Shimizu in AFC, in which smoothing of the rooms transfer function is achieved by averaging the impulse responses of multiple microphones. ``The virtual or rotating microphone technique'' reduces the number of microphones in the aforementioned AFC technology, while still achieving the same acoustical effects in the room. After the impulse responses at previously specified pairs of microphone positions are measured, the ratio of transfer functions for every pair is calculated, thus yielding a constant K. Next, microphones are removed and their impulse responses are reproduced by processing the incoming signal of each pair through a convolver, where the computed K constants have been previously stored. Band limiting, windowing and time variance effects are critical factors, in order to reduce incoherence effects and yield reliable approximations of inverse filters and consequently calculations of K. The project is implemented in a church lacking low frequency reverberation for music and makes use of 2 physical and 2 virtual microphones.

  9. 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.

  10. 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.

  11. Reconfigurable mask for adaptive coded aperture imaging (ACAI) based on an addressable MOEMS microshutter array

    NASA Astrophysics Data System (ADS)

    McNie, Mark E.; Combes, David J.; Smith, Gilbert W.; Price, Nicola; Ridley, Kevin D.; Brunson, Kevin M.; Lewis, Keith L.; Slinger, Chris W.; Rogers, Stanley

    2007-09-01

    Coded aperture imaging has been used for astronomical applications for several years. Typical implementations use a fixed mask pattern and are designed to operate in the X-Ray or gamma ray bands. More recent applications have emerged in the visible and infra red bands for low cost lens-less imaging systems. System studies have shown that considerable advantages in image resolution may accrue from the use of multiple different images of the same scene - requiring a reconfigurable mask. We report on work to develop a novel, reconfigurable mask based on micro-opto-electro-mechanical systems (MOEMS) technology employing interference effects to modulate incident light in the mid-IR band (3-5μm). This is achieved by tuning a large array of asymmetric Fabry-Perot cavities by applying an electrostatic force to adjust the gap between a moveable upper polysilicon mirror plate supported on suspensions and underlying fixed (electrode) layers on a silicon substrate. A key advantage of the modulator technology developed is that it is transmissive and high speed (e.g. 100kHz) - allowing simpler imaging system configurations. It is also realised using a modified standard polysilicon surface micromachining process (i.e. MUMPS-like) that is widely available and hence should have a low production cost in volume. We have developed designs capable of operating across the entire mid-IR band with peak transmissions approaching 100% and high contrast. By using a pixelated array of small mirrors, a large area device comprising individually addressable elements may be realised that allows reconfiguring of the whole mask at speeds in excess of video frame rates.

  12. 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

  13. Eliciting the most prominent perceived differences between microphones.

    PubMed

    Pearce, Andy; Brookes, Tim; Dewhirst, Martin; Mason, Russell

    2016-05-01

    The attributes contributing to the differences perceived between microphones (when auditioning recordings made with those microphones) are not clear from previous research. Consideration of technical specifications and expert opinions indicated that recording five programme items with eight studio and two microelectromechanical system microphones could allow determination of the attributes related to the most prominent inter-microphone differences. Pairwise listening comparisons between the resulting 50 recordings, followed by multi-dimensional scaling analysis, revealed up to 5 salient dimensions per programme item; 17 corresponding pairs of recordings were selected exemplifying the differences across those dimensions. Direct elicitation and panel discussions on the 17 pairs identified a hierarchy of 40 perceptual attributes. An attribute contribution experiment on the 31 lowest-level attributes in the hierarchy allowed them to be ordered by degree of contribution and showed brightness, harshness, and clarity to always contribute highly to perceived inter-microphone differences. This work enables the future development of objective models to predict these important attributes. PMID:27250188

  14. Adaptation.

    PubMed

    Broom, Donald M

    2006-01-01

    The term adaptation is used in biology in three different ways. It may refer to changes which occur at the cell and organ level, or at the individual level, or at the level of gene action and evolutionary processes. Adaptation by cells, especially nerve cells helps in: communication within the body, the distinguishing of stimuli, the avoidance of overload and the conservation of energy. The time course and complexity of these mechanisms varies. Adaptive characters of organisms, including adaptive behaviours, increase fitness so this adaptation is evolutionary. The major part of this paper concerns adaptation by individuals and its relationships to welfare. In complex animals, feed forward control is widely used. Individuals predict problems and adapt by acting before the environmental effect is substantial. Much of adaptation involves brain control and animals have a set of needs, located in the brain and acting largely via motivational mechanisms, to regulate life. Needs may be for resources but are also for actions and stimuli which are part of the mechanism which has evolved to obtain the resources. Hence pigs do not just need food but need to be able to carry out actions like rooting in earth or manipulating materials which are part of foraging behaviour. The welfare of an individual is its state as regards its attempts to cope with its environment. This state includes various adaptive mechanisms including feelings and those which cope with disease. The part of welfare which is concerned with coping with pathology is health. Disease, which implies some significant effect of pathology, always results in poor welfare. Welfare varies over a range from very good, when adaptation is effective and there are feelings of pleasure or contentment, to very poor. A key point concerning the concept of individual adaptation in relation to welfare is that welfare may be good or poor while adaptation is occurring. Some adaptation is very easy and energetically cheap and

  15. 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

  16. 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

  17. Development of a micromachined piezoelectric microphone for aeroacoustics applications.

    PubMed

    Horowitz, Stephen; Nishida, Toshikazu; Cattafesta, Louis; Sheplak, Mark

    2007-12-01

    This paper describes the design, fabrication, and characterization of a bulk-micromachined piezoelectric microphone for aeroacoustic applications. Microphone design was accomplished through a combination of piezoelectric composite plate theory and lumped element modeling. The device consists of a 1.80-mm-diam, 3-microm-thick, silicon diaphragm with a 267-nm-thick ring of piezoelectric material placed near the boundary of the diaphragm to maximize sensitivity. The microphone was fabricated by combining a sol-gel lead zirconate-titanate deposition process on a silicon-on-insulator wafer with deep-reactive ion etching for the diaphragm release. Experimental characterization indicates a sensitivity of 1.66 microVPa, dynamic range greater than six orders of magnitude (35.7-169 dB, re 20 microPa), a capacitance of 10.8 nF, and a resonant frequency of 59.0 kHz. PMID:18247752

  18. Ultra-low-noise preamplifier for condenser microphones

    NASA Astrophysics Data System (ADS)

    Starecki, Tomasz

    2010-12-01

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

  19. 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.

  20. Ultra-low-noise preamplifier for condenser microphones.

    PubMed

    Starecki, Tomasz

    2010-12-01

    The paper presents the design of a low-noise preamplifier dedicated for condenser measurement microphones used in high sensitivity applications, in which amplifier noise is the main factor limiting sensitivity of the measurements. In measurement microphone preamplifiers, the dominant source of noise at lower frequencies is the bias resistance of the input stage. In the presented solution, resistors were connected to the input stage by means of switches. The switches are opened during measurements, which disconnects the resistors from the input stage and results in noise reduction. Closing the switches allows for fast charging of the microphone capacitance. At low frequencies the noise of the designed preamplifier is a few times lower in comparison to similar, commercially available instruments. PMID:21198039

  1. 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.

  2. 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.

  3. 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

  4. 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...

  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. 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)

  7. 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

  8. 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…

  9. 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…

  10. Microflown based sound pressure microphone suitable for harsh environments

    NASA Astrophysics Data System (ADS)

    Yntema, D. R.; de Bree, Hans-Elias

    2005-09-01

    There are several cases where a sound field reconstruction or prediction is required under harsh conditions such as high temperature, humidity or chemical attack. A regular pressure microphone will not last long under these conditions. Electret based pressure microphones stop working well above 70 degrees centigrade and other type of pressure microphones often operate with a built in amplifier that does not function above 120 degrees centigrade. The functionality of a MEMS based Microflown acoustic particle velocity sensor in air lies in the use of two heated platinum wires that are resistant to high temperatures and chemical attack. The wires are supported by silicon that has no other function than provide support. A pressure microphone is made based upon the Microflown principle by putting it in the opening of an enclosure. In this paper a silicon and platinum based sound probe for harsh environments is created, combining particle velocity and pressure measurements in a harsh environment. Use of this sensor is possible up to 250 degrees centigrade, in humid and under most chemical environments. The probe realization as well as calibration measurements are presented.

  11. 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

  12. 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.

  13. The Magellan Adaptive Secondary VisAO Camera: diffraction-limited broadband visible imaging and 20mas fiber array IFU

    NASA Astrophysics Data System (ADS)

    Kopon, Derek; Close, Laird M.; Males, Jared; Gasho, Victor; Follette, Katherine

    2010-07-01

    The Magellan Adaptive Secondary AO system, scheduled for first light in the fall of 2011, will be able to simultaneously perform diffraction limited AO science in both the mid-IR, using the BLINC/MIRAC4 10μm camera, and in the visible using our novel VisAO camera. The VisAO camera will be able to operate as either an imager, using a CCD47 with 8.5 mas pixels, or as an IFS, using a custom fiber array at the focal plane with 20 mas elements in its highest resolution mode. In imaging mode, the VisAO camera will have a full suite of filters, coronagraphic focal plane occulting spots, and SDI prism/filters. The imaging mode should provide ~20% mean Strehl diffraction-limited images over the band 0.5-1.0 μm. In IFS mode, the VisAO instrument will provide R~1,800 spectra over the band 0.6-1.05 μm. Our unprecedented 20 mas spatially resolved visible spectra would be the highest spatial resolution achieved to date, either from the ground or in space. We also present lab results from our recently fabricated advanced triplet Atmospheric Dispersion Corrector (ADC) and the design of our novel wide-field acquisition and active optics lens. The advanced ADC is designed to perform 58% better than conventional doublet ADCs and is one of the enabling technologies that will allow us to achieve broadband (0.5-1.0μm) diffraction limited imaging and wavefront sensing in the visible.

  14. 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

  15. 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

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

    PubMed

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

    2006-10-01

    This study examined the effect that signal processing strategies used in modern hearing aids, such as multi-channel WDRC, noise reduction, and directional microphones have on interaural difference cues and horizontal localization performance relative to linear, time-invariant amplification. Twelve participants were bilaterally fitted with BTE devices. Horizontal localization testing using a 360 degrees loudspeaker array and broadband pulsed pink noise was performed two weeks, and two months, post-fitting. The effect of noise reduction was measured with a constant noise present at 80 degrees azimuth. Data were analysed independently in the left/right and front/back dimension and showed that of the three signal processing strategies, directional microphones had the most significant effect on horizontal localization performance and over time. Specifically, a cardioid microphone could decrease front/back errors over time, whereas left/right errors increased when different microphones were fitted to left and right ears. Front/back confusions were generally prominent. Objective measurements of interaural differences on KEMAR explained significant shifts in left/right errors. In conclusion, there is scope for improving the sense of localization in hearing aid users. PMID:17062498

  17. SVD-based optimal filtering for noise reduction in dual microphone hearing aids: a real time implementation and perceptual evaluation.

    PubMed

    Maj, Jean-Baptiste; Royackers, Liesbeth; Moonen, Marc; Wouters, Jan

    2005-09-01

    In this paper, the first real-time implementation and perceptual evaluation of a singular value decomposition (SVD)-based optimal filtering technique for noise reduction in a dual microphone behind-the-ear (BTE) hearing aid is presented. This evaluation was carried out for a speech weighted noise and multitalker babble, for single and multiple jammer sound source scenarios. Two basic microphone configurations in the hearing aid were used. The SVD-based optimal filtering technique was compared against an adaptive beamformer, which is known to give significant improvements in speech intelligibility in noisy environment. The optimal filtering technique works without assumptions about a speaker position, unlike the two-stage adaptive beamformer. However this strategy needs a robust voice activity detector (VAD). A method to improve the performance of the VAD was presented and evaluated physically. By connecting the VAD to the output of the noise reduction algorithms, a good discrimination between the speech-and-noise periods and the noise-only periods of the signals was obtained. The perceptual experiments demonstrated that the SVD-based optimal filtering technique could perform as well as the adaptive beamformer in a single noise source scenario, i.e., the ideal scenario for the latter technique, and could outperform the adaptive beamformer in multiple noise source scenarios. PMID:16189969

  18. 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.

  19. 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.

  20. Design and feasibility studies for piezoresistive MEMs microphones

    NASA Astrophysics Data System (ADS)

    Varanda, Brenno R.

    Preliminary acoustic response formulation and finite element models for two distinct omnidirectional piezoresistive microelectromechanical microphone designs with infinite air back volume are investigated in this thesis. An ambitious goal of achieving a 1.0 mV/V˙Pa electrical sensitivity together with a frequency passband range of 0-15 KHz was set for both designs. The first design is a feasibility parameterization study sponsored by the Center for Advanced Microelectronics Manufacturing (CAMM) to construct a circular piezoresistive microphone from flexible roll-to-roll electronic manufacturing technologies. The second design was proposed by R. N. Miles in 2006 and consists of a rectangular miniature cantilever diaphragm (505im x 685im) manufactured from polysilicon micofabrication. The development of an accurate finite element model together with analytical tools to estimate the diaphragm's electrical sensitivity are discussed and compared to three manufactured prototypes, with a final prototype.

  1. Optical sensing in a directional hearing aid microphone

    NASA Astrophysics Data System (ADS)

    Zhou, Shuang

    This thesis describes the simulation and analysis of the use of optical sensing for a MEMS directional microphone. Diffraction gratings integrated with micro-electromechanical-systems (MEMS) offer an optical sensing scheme with high detection sensitivity, low noise level and compact device structure. An optical sensing method is applied in a hearing aid microphone to detect the movement of the diaphragm due to sound. Diffraction grating fingers are fabricated on both sides of the diaphragm with a gold mirror on top. Two photo detectors are placed on the substrate symmetrically to detect the positive and negative first order diffraction of 850 nm VCSEL light. A finite element analysis model is built in COMSOL to study the light distribution and energy loss. The signal output, predicted using an analytical model is shown to agree well with those obtained using the finite element model.

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

    ERIC Educational Resources Information Center

    American School Band Directors Association, Newark, OH.

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

  3. On Building Immersive Audio Applications Using Robust Adaptive Beamforming and Joint Audio-Video Source Localization

    NASA Astrophysics Data System (ADS)

    Beracoechea, J. A.; Torres-Guijarro, S.; García, L.; Casajús-Quirós, F. J.

    2006-12-01

    This paper deals with some of the different problems, strategies, and solutions of building true immersive audio systems oriented to future communication applications. The aim is to build a system where the acoustic field of a chamber is recorded using a microphone array and then is reconstructed or rendered again, in a different chamber using loudspeaker array-based techniques. Our proposal explores the possibility of using recent robust adaptive beamforming techniques for effectively estimating the original sources of the emitting room. A joint audio-video localization method needed in the estimation process as well as in the rendering engine is also presented. The estimated source signal and the source localization information drive a wave field synthesis engine that renders the acoustic field again at the receiving chamber. The system performance is tested using MUSHRA-based subjective tests.

  4. 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.

  5. A New Blind Adaptive Array Antenna Based on CMA Criteria for M-Ary/SS Signals Suitable for Software Defined Radio Architecture

    NASA Astrophysics Data System (ADS)

    Kozuma, Miho; Sasaki, Atsushi; Kamiya, Yukihiro; Fujii, Takeo; Umebayashi, Kenta; Suzuki, Yasuo

    M-ary/SS is a version of Direct Sequence/Spread Spectrum (DS/SS) aiming to improve the spectral efficiency employing orthogonal codes. However, due to the auto-correlation property of the orthogonal codes, it is impossible to detect the symbol timing by observing correlator outputs. Therefore, conventionally, a preamble has been inserted in M-ary/SS, signals. In this paper, we propose a new blind adaptive array antenna for M-ary/SS systems that combines signals over the space axis without any preambles. It is surely an innovative approach for M-ary/SS. The performance is investigated through computer simulations.

  6. 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

  7. A low-noise differential microphone inspired by the ears of the parasitoid fly Ormia ochracea

    PubMed Central

    Miles, R. N.; Su, Q.; Cui, W.; Shetye, M.; Degertekin, F. L.; Bicen, B.; Garcia, C.; Jones, S.; Hall, N.

    2009-01-01

    A miniature differential microphone is described having a low-noise floor. The sensitivity of a differential microphone suffers as the distance between the two pressure sensing locations decreases, resulting in an increase in the input sound pressure-referred noise floor. In the microphone described here, both the diaphragm thermal noise and the electronic noise are minimized by a combination of novel diaphragm design and the use of low-noise optical sensing that has been integrated into the microphone package. The differential microphone diaphragm measures 1×2 mm2 and is fabricated out of polycrystalline silicon. The diaphragm design is based on the coupled directionally sensitive ears of the fly Ormia ochracea. The sound pressure input-referred noise floor of this miniature differential microphone has been measured to be less than 36 dBA. PMID:19354377

  8. Experimental results for a photonic time reversal processor for the adaptive control of an ultra wideband phased array antenna

    NASA Astrophysics Data System (ADS)

    Zmuda, Henry; Fanto, Michael; McEwen, Thomas

    2008-04-01

    This paper describes a new concept for a photonic implementation of a time reversed RF antenna array beamforming system. The process does not require analog to digital conversion to implement and is therefore particularly suited for high bandwidth applications. Significantly, propagation distortion due to atmospheric effects, clutter, etc. is automatically accounted for with the time reversal process. The approach utilizes the reflection of an initial interrogation signal from off an extended target to precisely time match the radiating elements of the array so as to re-radiate signals precisely back to the target's location. The backscattered signal(s) from the desired location is captured by each antenna and used to modulate a pulsed laser. An electrooptic switch acts as a time gate to eliminate any unwanted signals such as those reflected from other targets whose range is different from that of the desired location resulting in a spatial null at that location. A chromatic dispersion processor is used to extract the exact array parameters of the received signal location. Hence, other than an approximate knowledge of the steering direction needed only to approximately establish the time gating, no knowledge of the target position is required, and hence no knowledge of the array element time delay is required. Target motion and/or array element jitter is automatically accounted for. Presented here are experimental results that demonstrate the ability of a photonic processor to perform the time-reversal operation on ultra-short electronic pulses.

  9. 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.

  10. On the potential of fixed arrays for hearing aids.

    PubMed

    Stadler, R W; Rabinowitz, W M

    1993-09-01

    Microphone arrays with fixed (time-invariant) weights are directed at enhancing a desired signal from one direction (straight ahead) while attenuating spatially distributed interference and reverberation. Using the theory of sensitivity-constrained optimal beamforming [Cox et al., IEEE Trans. Acoust. Speech Sig. Process. ASSP-34, 393-398 (1986)], free-field arrays of head-sized extents were studied. The key parameters affecting array design and performance are the set of transfer functions from the target direction to each array microphone [H(f)] and the intermicrophone cross-spectral densities for isotropic noise [Szz(f)]. Design variables included the orientation of the array, the number, and [as motivated by Soede, Ph.D. thesis, Delft University of Technology (1990)] the directionality of the microphones within the array, and the complexity and robustness of the required processing. Performance was characterized by the broadband intelligibility-weighted directivity (gain against isotropic noise) and noise sensitivity (reflecting the array's sensitivity to uncorrelated noise, as well as device tolerances). For broadside orientation, a variety of arrays based on cardioid and hypercardioid microphones gave very similar performance. They can provide directivities of 7-8 dB with easily implemented weights (simple scalars). For endfire orientation, as Soede (1990) recognized, similar directivities result with weights based on analog gains and pure time delays. However, with weightings chosen independently for each frequency, directivities up to approximately 11 dB may be obtained, although the increased noise sensitivities of these arrays require practical evaluation. Because of sound diffraction, placement of arrays onto the head potentially impacts both their design and performance. In-situ measurements of H(f) and Szz(f) as well as simplified theoretical models are suggested to explore the optimization of head-mounted arrays. PMID:8408974

  11. 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

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

    NASA Astrophysics Data System (ADS)

    Toda, Minoru; Thompson, Mitchell

    2005-09-01

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

  13. 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.

  14. 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.

  15. Use of proximity effect in hearing aid microphones to increase telephone intelligibility in noise

    NASA Astrophysics Data System (ADS)

    Wilson, Kathryn R.

    1988-07-01

    This thesis describes an experiment to test the use of the proximity effect to increase the intelligibility of telephone speech for hearing-aid wearers. NU-6 word lists were played through the equivalent of long-distance telephone lines with a standard Bell 500 handset, while Multi-Talker noise was played in the background at three different levels. The signals were picked up with one of three microphones placed by the ear of a dummy head: a first-order pressure-gradient microphone (bi-directional), a zero-order microphone (omni-directional), or one with order between zero and one (cardioid). The signal picked up by these microphones was recorded and played back to normal-hearing subjects through a modified hearing aid, while the Multi-Talker noise was played in the background. The pressure gradient microphones allowed significant better understanding of the telephone signal than did the pressure microphone and this difference was more pronounced at higher noise levels. The bidirectional and cardioid microphones did not provide significantly different scores at any noise level. It is argued that this similarity may be due to head effects reducing the pressure-gradient sensitivity of the microphones. The use of the proximity effect to enable hearing aids to pick up a telephone conversation while discriminating against background noise appears to be successful.

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

    PubMed

    Van Renterghem, Timothy; Thomas, Pieter; Dominguez, Frederico; Dauwe, Samuel; Touhafi, Abdellah; Dhoedt, Bart; Botteldooren, Dick

    2011-03-01

    The massive production of microphones for consumer electronics, and the shift from dedicated processing hardware to PC-based systems, opens the way to build affordable, extensive noise measurement networks. Applications include e.g. noise limit and urban soundscape monitoring, and validation of calculated noise maps. Microphones are the critical components of such a network. Therefore, in a first step, some basic characteristics of 8 microphones, distributed over a wide range of price classes, were measured in a standardized way in an anechoic chamber. In a next step, a thorough evaluation was made of the ability of these microphones to be used for environmental noise monitoring. This was done during a continuous, half-year lasting outdoor experiment, characterized by a wide variety of meteorological conditions. While some microphones failed during the course of this test, it was shown that it is possible to identify cheap microphones that highly correlate to the reference microphone during the full test period. When the deviations are expressed in total A-weighted (road traffic) noise levels, values of less than 1 dBA are obtained, in excess to the deviation amongst reference microphones themselves. PMID:21157618

  17. Quantifying Errors in Jet Noise Research Due to Microphone Support Reflection

    NASA Technical Reports Server (NTRS)

    Nallasamy, Nambi; Bridges, James

    2002-01-01

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

  18. Testing and characterization of a biologically-inspired first-order directional MEMS microphone

    NASA Astrophysics Data System (ADS)

    Antonelli, Daniel

    First-order directional microphones have a response that is proportional to the spatial gradient of sound pressure. The overall response, however, will also be influenced by the average sound pressure acting on the microphone diaphragm. For directional microphones to exhibit the desired first-order figure-8 directivity pattern, the response must be dominated by the pressure gradient rather than the pressure. A testing process has been developed to characterize the acoustic response of a biologically-inspired first-order directional MEMS microphone by separating the total measured response into the response due to the spatial average of the pressure and the response due to pressure gradient. Understanding how the pressure and pressure gradient of a sound field separately influence the overall behavior of this class of microphone is critical to assessing their performance. An experimental test setup and data processing algorithms have been developed which are shown to successfully achieve these goals.

  19. Aeroacoustic probe design for microphone to reduce flow-induced self-noise

    NASA Technical Reports Server (NTRS)

    Allew, Cristopher S.; Soderman, Paul T.

    1993-01-01

    A new aerodynamic microphone forebody design was tested in the National Full-scale Aerodynamics Complex (NFAC) at NASA Ames Research Center. An overview of the design theory is presented. The new microphone forebody was tested along with the industry standard Bruel and Kjaer (B&K) aerodynamic microphone forebody in the Ames 7- by 10-Foot Wind Tunnel, NFAC 40- by 80-Foot Wind Tunnel and Anechoic Chamber. The wind tunnel results show that high frequency tones, present with the B&K microphone forebody, are eliminated with the new probe. Broad band background noise levels at lower frequencies are also shown to decrease with the 1/4 inch version of the new probe. Finally, the Anechoic Chamber results show no difference in the directional characteristics between the new and B&K microphone forebodies below 80 kHz.

  20. Implementation of Joint Pre-FFT Adaptive Array Antenna and Post-FFT Space Diversity Combining for Mobile ISDB-T Receiver

    NASA Astrophysics Data System (ADS)

    Pham, Dang Hai; Gao, Jing; Tabata, Takanobu; Asato, Hirokazu; Hori, Satoshi; Wada, Tomohisha

    In our application targeted here, four on-glass antenna elements are set in an automobile to improve the reception quality of mobile ISDB-T receiver. With regard to the directional characteristics of each antenna, we propose and implement a joint Pre-FFT adaptive array antenna and Post-FFT space diversity combining (AAA-SDC) scheme for mobile ISDB-T receiver. By applying a joint hardware and software approach, a flexible platform is realized in which several system configuration schemes can be supported; the receiver can be reconfigured on the fly. Simulation results show that the AAA-SDC scheme drastically improves the performance of mobile ISDB-T receiver, especially in the region of large Doppler shift. The experimental results from a field test also confirm that the proposed AAA-SDC scheme successfully achieves an outstanding reception rate up to 100% while moving at the speed of 80km/h.

  1. 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. PMID:26627758

  2. A Two-Microphone Noise Reduction System for Cochlear Implant Users with Nearby Microphones—Part II: Performance Evaluation

    NASA Astrophysics Data System (ADS)

    Kompis, Martin; Bertram, Matthias; Senn, Pascal; Müller, Joachim; Pelizzone, Marco; Häusler, Rudolf

    2008-12-01

    Users of cochlear implants (auditory aids, which stimulate the auditory nerve electrically at the inner ear) often suffer from poor speech understanding in noise. We evaluate a small (intermicrophone distance 7 mm) and computationally inexpensive adaptive noise reduction system suitable for behind-the-ear cochlear implant speech processors. The system is evaluated in simulated and real, anechoic and reverberant environments. Results from simulations show improvements of 3.4 to 9.3 dB in signal to noise ratio for rooms with realistic reverberation and more than 18 dB under anechoic conditions. Speech understanding in noise is measured in 6 adult cochlear implant users in a reverberant room, showing average improvements of 7.9-9.6 dB, when compared to a single omnidirectional microphone or 1.3-5.6 dB, when compared to a simple directional two-microphone device. Subjective evaluation in a cafeteria at lunchtime shows a preference of the cochlear implant users for the evaluated device in terms of speech understanding and sound quality.

  3. 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

  4. Transversely Excited Multipass Photoacoustic Cell Using Electromechanical Film as Microphone

    PubMed Central

    Saarela, Jaakko; Sand, Johan; Sorvajärvi, Tapio; Manninen, Albert; Toivonen, Juha

    2010-01-01

    A novel multipass photoacoustic cell with five stacked electromechanical films as a microphone has been constructed, tested and characterized. The photoacoustic cell is an open rectangular structure with two steel plates facing each other. The longitudinal acoustic resonances are excited transversely in an optical multipass configuration. A detection limit of 22 ppb (10−9) was achieved for flowing NO2 in N2 at normal pressure by using the maximum of 70 laser beams between the resonator plates. The corresponding minimum detectable absorption and the normalized noise-equivalent absorption coefficients were 2.2 × 10−7 cm−1 and 3.2 × 10−9 cm−1WHz−1/2, respectively. PMID:22219662

  5. 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.

  6. 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.

  7. 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.

  8. Adaptive phase-locked fiber array with wavefront phase tip-tilt compensation using piezoelectric fiber positioners

    NASA Astrophysics Data System (ADS)

    Liu, Ling; Vorontsov, Mikhail A.; Polnau, Ernst; Weyrauch, Thomas; Beresnev, Leonid A.

    2007-09-01

    In this paper, we present the recent development of a conformal optical system with three adaptive phase-locked fiber elements. The coherent beam combining based on stochastic parallel gradient descent (SPGD) algorithm is investigated. We implement both phase-locking control and wavefront phase tip-tilt control in our conformal optical system. The phase-locking control is performed with fiber-coupled lithium niobate phase shifters which are modulated by an AVR micro-processor based SPGD controller. The perturbation rate of this SPGD controller is ~95,000 iterations per second. Phase-locking compensation bandwidth for phase distortion amplitude of 2π-radian phase shift is >100Hz. The tip-tilt control is realized with piezoelectric fiber positioners which are modulated by a computer-based software SPGD controller. The perturbation rate of the tip-tilt SPGD controller is up to ~950 iterations per second. The tip-tilt compensation bandwidth using fiber positioners is ~10Hz at 60-μrad. jitter swing angle.

  9. 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...

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

    NASA Astrophysics Data System (ADS)

    Bicen, Baris

    Measuring acoustic pressure gradients is critical in many applications such as directional microphones for hearing aids and sound intensity probes. This measurement is especially challenging with decreasing microphone size, which reduces the sensitivity due to small spacing between the pressure ports. Novel, micromachined biomimetic microphone diaphragms are shown to provide high sensitivity to pressure gradients on one side of the diaphragm with low thermal mechanical noise. These structures have a dominant mode shape with see-saw like motion in the audio band, responding to pressure gradients as well as spurious higher order modes sensitive to pressure. In this dissertation, integration of a diffraction based optical detection method with these novel diaphragm structures to implement a low noise optical pressure gradient microphone is described and experimental characterization results are presented, showing 36 dBA noise level with 1mm port spacing, nearly an order of magnitude better than the current gradient microphones. The optical detection scheme also provides electrostatic actuation capability from both sides of the diaphragm separately which can be used for active force feedback. A 4-port electromechanical equivalent circuit model of this microphone with optical readout is developed to predict the overall response of the device to different acoustic and electrostatic excitations. The model includes the damping due to complex motion of air around the microphone diaphragm, and it calculates the detected optical signal on each side of the diaphragm as a combination of two separate dominant vibration modes. This equivalent circuit model is verified by experiments and used to predict the microphone response with different force feedback schemes. Single sided force feedback is used for active damping to improve the linearity and the frequency response of the microphone. Furthermore, it is shown that using two sided force feedback one can significantly suppress

  11. 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.

  12. Bruel and Kjaer 4944 Microphone Grid Frequency Response Function System Identification

    NASA Technical Reports Server (NTRS)

    Bennett, Reginald; Lee, Erik

    2010-01-01

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

  13. 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.

  14. 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

  15. Low frequency wind noise contributions in measurement microphones.

    PubMed

    Raspet, Richard; Yu, Jiao; Webster, Jeremy

    2008-03-01

    In a previous paper [R. Raspet, et al., J. Acoust. Soc. Am. 119, 834-843 (2006)], a method was introduced to predict upper and lower bounds for wind noise measured in spherical wind-screens from the measured incident velocity spectra. That paper was restricted in that the predictions were only valid within the inertial range of the incident turbulence, and the data were from a measurement not specifically designed to test the predictions. This paper extends the previous predictions into the source region of the atmospheric wind turbulence, and compares the predictions to measurements made with a large range of wind-screen sizes. Predictions for the turbulence-turbulence interaction pressure spectrum as well as the stagnation pressure fluctuation spectrum are calculated from a form fit to the velocity fluctuation spectrum. While the predictions for turbulence-turbulence interaction agree well with measurements made within large (1.0 m) wind-screens, and the stagnation pressure predictions agree well with unscreened gridded microphone measurements, the mean shear-turbulence interaction spectra do not consistently appear in measurements. PMID:18345815

  16. Human Pulse Wave Measurement by MEMS Electret Condenser Microphone

    NASA Astrophysics Data System (ADS)

    Nomura, Shusaku; Hanasaka, Yasushi; Ishiguro, Tadashi; Ogawa, Hiroshi

    A micro Electret Condenser Microphone (ECM) fabricated by Micro Electro Mechanical System (MEMS) technology was employed as a novel apparatus for human pulse wave measurement. Since ECM frequency response characteristic, i.e. sensitivity, logically maintains a constant level at lower than the resonance frequency (stiffness control), the slightest pressure difference at around 1.0Hz generated by human pulse wave is expected to detect by MEMS-ECM. As a result of the verification of frequency response of MEMS-ECM, it was found that -20dB/dec of reduction in the sensitivity around 1.0Hz was engendered by a high input-impedance amplifier, i.e. the field effect transistor (FET), mounted near MEMS chip for amplifying tiny ECM signal. Therefore, MEMS-ECM is assumed to be equivalent with a differentiation circuit at around human pulse frequency. Introducing compensation circuit, human pulse wave was successfully obtained. In addition, the radial and ulnar artery tracing, and pulse wave velocity measurement at forearm were demonstrated; as illustrating a possible application of this micro device.

  17. Correlative changes of auditory nerve and microphonic potentials throughout sleep.

    PubMed

    Velluti, R; Pedemonte, M; García-Austt, E

    1989-05-01

    Gross cochlear potentials in response to alternating clicks and pure tone bursts were recorded in guinea-pigs with chronically implanted electrodes in the round window during sleep and the awake state. A significant increase in both averaged potentials, the compound auditory nerve action potential (cAP) and cochlear microphonics (CM) occurred in slow wave sleep (SWS) with a subsequent diminution in paradoxical sleep (PS) periods. The cAP, CM, amplitude and area averages were similar during quiet wakefulness and in PS. Moreover, as an episode of PS progressed, the recorded potentials continued to decrease. On the other hand, increased averaged values were again observed during a subsequent episode of SWS. An involvement of the efferent olivo-cochlear bundle is postulated, first, because it is the only known pathway connecting the CNS and the auditory periphery and, second, because several key pre-receptor variables (middle ear muscles and ossicles and sound-source ear relation) were either abolished or altered dramatically. PMID:2737966

  18. 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.

  19. Localization using ground- and air-based acoustic arrays

    NASA Astrophysics Data System (ADS)

    Goldman, Geoffrey H.; Reiff, Chris

    2011-06-01

    Techniques were developed to localize acoustic quasiperiodic signals using microphone arrays located on the ground and on an aerostat. The direction of arrival (DOA) was computed at each array and then the position of the source was estimated using algorithms based upon triangulation. Differential time delays between the microphones in a tetrahedral array were estimated in the frequency domain, and then DOA estimates were calculated using a weighted least squares approach. The location of the target was calculated by minimizing the weighted squared error of a cost function for different combinations of DOA estimates. The algorithms were tested offline using data collected by the U.S. Army Research Laboratory on an aircraft. The ground-truth position of the target was recorded using a GPS system as it maneuvered and compared to the results obtained from the localization algorithms. The algorithms performed well when estimating the x and y positions, but had difficulty obtaining consistently good z positions, or equivalently, height estimates.

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

    PubMed

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

    2006-10-01

    In order to provide relevant information to mobile users, such as workers engaging in the manual tasks of maintenance and assembly, a wearable computer requires information about the user's specific activities. This work focuses on the recognition of activities that are characterized by a hand motion and an accompanying sound. Suitable activities can be found in assembly and maintenance work. Here, we provide an initial exploration into the problem domain of continuous activity recognition using on-body sensing. We use a mock "wood workshop" assembly task to ground our investigation. We describe a method for the continuous recognition of activities (sawing, hammering, filing, drilling, grinding, sanding, opening a drawer, tightening a vise, and turning a screwdriver) using microphones and three-axis accelerometers mounted at two positions on the user's arms. Potentially "interesting" activities are segmented from continuous streams of data using an analysis of the sound intensity detected at the two different locations. Activity classification is then performed on these detected segments using linear discriminant analysis (LDA) on the sound channel and hidden Markov models (HMMs) on the acceleration data. Four different methods at classifier fusion are compared for improving these classifications. Using user-dependent training, we obtain continuous average recall and precision rates (for positive activities) of 78 percent and 74 percent, respectively. Using user-independent training (leave-one-out across five users), we obtain recall rates of 66 percent and precision rates of 63 percent. In isolation, these activities were recognized with accuracies of 98 percent, 87 percent, and 95 percent for the user-dependent, user-independent, and user-adapted cases, respectively. PMID:16986539

  1. 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.

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

    NASA Technical Reports Server (NTRS)

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

    1992-01-01

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

  3. 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.

  4. 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.

  5. Noise path identification using face-to-face and side-by-side microphone arrangements

    NASA Technical Reports Server (NTRS)

    Atwal, M.; Bernhard, R.

    1984-01-01

    In large complex structures, with several major sound transmission paths and high levels of background noise, it can be a complex task to locate and rank the contribution of an individual sound transmission path. The two microphone acoustic intensity techniques are investigated as a tool for path identification. Laboratory tests indicate that, if the intensity transmitted through a particular section of the fuselage is measured in the presence and absence of flanking paths using the face to face and side by side microphone arrangements, then no significant difference exists between the two measured intensities if the face to face microphone arrangement is used. However, if the side by side arrangement is used, then considerable difference exists between the two measured intensities.

  6. 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.

  7. Acoustic isolation vessel for measurement of the background noise in microphones

    NASA Technical Reports Server (NTRS)

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

    1993-01-01

    An acoustic isolation vessel has been developed to measure the background noise in microphones. The test microphone is installed in an inner vessel, which is suspended within an outer vessel, and the intervening air space is evacuated to a high vacuum. An analytical expression for the transmission coefficient is derived, based on a five-media model, and compared to experiment. At an isolation vacuum of 5 x 10 exp -6 Torr the experimental transmission coefficient was found to be lower than -155 dB at frequencies ranging from 40 to 1200 Hz. Measurements of the A-weighted noise levels of commercial condenser microphones of four different sizes show good agreement with published values.

  8. 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.

  9. 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.

  10. 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.

  11. 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%.

  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

    ... 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...

  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

    ... 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...

  14. 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...

  15. 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

    ... 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...

  16. 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

    ... 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...

  17. 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.

  18. A DETAILED GRAVITATIONAL LENS MODEL BASED ON SUBMILLIMETER ARRAY AND KECK ADAPTIVE OPTICS IMAGING OF A HERSCHEL-ATLAS SUBMILLIMETER GALAXY AT z = 4.243 {sup ,} {sup ,}

    SciTech Connect

    Bussmann, R. S.; Gurwell, M. A.; Fu Hai; Cooray, A.; Smith, D. J. B.; Bonfield, D.; Dunne, L.; Dye, S.; Eales, S.; Auld, R.; Baes, M.; Fritz, J.; Baker, A. J.; Cava, A.; Clements, D. L.; Dariush, A.; Coppin, K.; Dannerbauer, H.; De Zotti, G.; Hopwood, R.; and others

    2012-09-10

    We present high-spatial resolution imaging obtained with the Submillimeter Array (SMA) at 880 {mu}m and the Keck adaptive optics (AO) system at the K{sub S}-band of a gravitationally lensed submillimeter galaxy (SMG) at z = 4.243 discovered in the Herschel Astrophysical Terahertz Large Area Survey. The SMA data (angular resolution Almost-Equal-To 0.''6) resolve the dust emission into multiple lensed images, while the Keck AO K{sub S}-band data (angular resolution Almost-Equal-To 0.''1) resolve the lens into a pair of galaxies separated by 0.''3. We present an optical spectrum of the foreground lens obtained with the Gemini-South telescope that provides a lens redshift of z{sub lens} = 0.595 {+-} 0.005. We develop and apply a new lens modeling technique in the visibility plane that shows that the SMG is magnified by a factor of {mu} = 4.1 {+-} 0.2 and has an intrinsic infrared (IR) luminosity of L{sub IR} = (2.1 {+-} 0.2) Multiplication-Sign 10{sup 13} L{sub Sun }. We measure a half-light radius of the background source of r{sub s} = 4.4 {+-} 0.5 kpc which implies an IR luminosity surface density of {Sigma}{sub IR} (3.4 {+-} 0.9) Multiplication-Sign 10{sup 11} L{sub Sun} kpc{sup -2}, a value that is typical of z > 2 SMGs but significantly lower than IR luminous galaxies at z {approx} 0. The two lens galaxies are compact (r{sub lens} Almost-Equal-To 0.9 kpc) early-types with Einstein radii of {theta}{sub E1} 0.57 {+-} 0.01 and {theta}{sub E2} = 0.40 {+-} 0.01 that imply masses of M{sub lens1} = (7.4 {+-} 0.5) Multiplication-Sign 10{sup 10} M{sub Sun} and M{sub lens2} = (3.7 {+-} 0.3) Multiplication-Sign 10{sup 10} M{sub Sun }. The two lensing galaxies are likely about to undergo a dissipationless merger, and the mass and size of the resultant system should be similar to other early-type galaxies at z {approx} 0.6. This work highlights the importance of high spatial resolution imaging in developing models of strongly lensed galaxies discovered by Herschel.

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

    PubMed Central

    Wagner, Randall P.; Guthrie, William F.

    2015-01-01

    The devices calibrated most frequently by the acoustical measurement services at the National Institute of Standards and Technology (NIST) over the 50-year period from 1963 to 20121 were one-inch condenser microphones of three specific standard types: LS1Pn, LS1Po, and WS1P. Due to its long history of providing calibrations of such microphones to customers, NIST is in a unique position to analyze data concerning the long-term stability of these devices. This long history has enabled NIST to acquire and aggregate a substantial amount of repeat calibration data for a large number of microphones that belong to various other standards and calibration laboratories. In addition to determining microphone sensitivities at the time of calibration, it is important to have confidence that the microphones do not typically undergo significant drift as compared to the calibration uncertainty during the periods between calibrations. For each of the three microphone types, an average drift rate and approximate 95 % confidence interval were computed by two different statistical methods, and the results from the two methods were found to differ insignificantly in each case. These results apply to typical microphones of these types that are used in a suitable environment and handled with care. The average drift rate for Type LS1Pn microphones was −0.004 dB/year to 0.003 dB/year. The average drift rate for Type LS1Po microphones was −0.016 dB/year to 0.008 dB/year. The average drift rate for Type WS1P microphones was −0.004 dB/year to 0.018 dB/year. For each of these microphone types, the average drift rate is not significantly different from zero. This result is consistent with the performance expected of condenser microphones designed for use as transfer standards. In addition, the values that bound the confidence intervals are well within the limits specified for long-term stability in international standards. Even though these results show very good long-term stability

  20. Design of an Acoustic Array for Comparison with an Alternative Source Localization Method

    NASA Astrophysics Data System (ADS)

    Coombs, Deshawn; Lewalle, Jacques; Glauser, Mark; Wang, Guannan

    2013-11-01

    We report on the design, testing and construction of a conventional acoustic array, and document an alternate method of signal processing. The purpose of the new algorithm is to improve the spatial localization of acoustic sources. The reference results are obtained using the beamforming algorithm. The array design includes 60 microphones with a maximum aperture diameter of 39 inches. The arrays target frequency range is 500-5000 Hz. The new algorithm uses fewer microphones. We will show results with simulated signals and with jet noise experimental data. Details of the array calibration and representative data from measurements will be presented along with data post-processing procedures. Support from Syracuse University MAE department and LSAMP.

  1. Development of a directional hearing instrument based on array technology.

    PubMed

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

    1993-08-01

    A directional hearing aid might be beneficial in reducing background noise in relation to the desired speech signal. Conventional hearing aids with a directional cardioid microphone are insufficient because of the low directivity of cardioids. Research was done to develop microphone(s) with strong directional characteristics using array techniques. Particular emphasis was given to optimization and stability. Free-field simulations of several robust models show that a directivity index of 9 dB can be obtained at the higher frequencies. Simulations were verified with a laboratory model. The results of the measurements show a good agreement with the simulations. Based on simulations and measurements, two portable models were developed and tested with a KEMAR manikin. The KEMAR measurements show that the two models give an improvement of the signal-to-noise ratio of approximately 7.5 dB in a diffuse sound field. It may be concluded that the developed microphones have the capability to reach a significant improvement of speech intelligibility in noise under practical circumstances. PMID:8370885

  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. 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…

  4. 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…

  5. Effects of air cavity on fly-ear inspired directional microphones: a numerical study

    NASA Astrophysics Data System (ADS)

    Liu, Haijun; Yu, Miao

    2011-04-01

    The superacute ear of the parasitoid fly Ormia ochracea has inspired the development of a variety of novel miniature directional microphones for sound source localization, in which the effects of air cavity backing the eardrums are often neglected without validation. In the original testing on the fly ear, the integrity of the air space is shown not to be the key to the intertympnal coupling. However, it does not necessarily mean that the tympanum can be treated as in vacuo, and the effects of the air cavity backing the eardrums have yet to be fully understood. In this article, a normalized version of our previous model of air-backed circular membranes is derived to study the conditions under which the air cavity can be indeed neglected. This model is then used to study a fly-ear inspired directional microphone design with two clamped circular membranes mechanically coupled by a bridge. The performance of the directional microphone with air cavity is evaluated in comparison to its counterpart in vacuo. This article not only provides more insights into the fly ear phenomena, but builds a theoretical foundation on whether and how to take the air cavity into account in the design of pressure sensors and directional microphones in general.

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

    NASA Technical Reports Server (NTRS)

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

    2008-01-01

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

  7. Acoustic analysis of the interaction of choral arrangements, musical selection, and microphone location.

    PubMed

    Morris, Richard J; Mustafa, Ashley J; McCrea, Christopher R; Fowler, Linda P; Aspaas, Christopher

    2007-09-01

    Acoustic differences were evaluated among three choral arrangements and two choral textures recorded at three microphone locations. A choir was recorded when singing two musical selections of different choral texture, one homophonic and one polyphonic. Both musical selections were sung in three choral arrangements: block sectional, sectional-in-columns, and mixed. Microphones were placed at the level of the choristers, the conductor, and the audience. The recordings at each location were analyzed using long-term average spectrum (LTAS). The LTAS from the mixed arrangement exhibited more signal amplitude than the other arrangements in the range of 1000-3500Hz. When considering the musical selections, the chorus produced more signal amplitude in the region of 1800-2200Hz for the homophonic selection. In addition, the LTAS produced by the choir for the homophonic selection varied across the microphone locations. As for the microphone location, the LTAS of the signal detected directly in front of the chorus had a greater slope than the other two locations. Thus, the acoustic signal near the choristers differed from the signals near the conductor and in the audience. Conductors may be using acoustic information from the region of the second and third formants when they decide how to arrange a choir for a particular musical selection. PMID:16806816

  8. 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.

  9. High frequency calibration of MEMS microphones using spherical N-waves

    NASA Astrophysics Data System (ADS)

    Ollivier, S.; Desjouy, C.; Yuldashev, P. Y.; Koumela, A.; Salze, E.; Karzova, M.; Rufer, L.; Blanc-Benon, Ph.

    2015-10-01

    In the context of the scientific program SIMMIC supported by the French National Agency for Research (SIMI 9, ANR 2010 BLANC 0905 03), new wide band MEMS piezoresistive microphones have been designed and fabricated for weak shock wave measurements. The fabricated microphones have a high frequency resonance between 300 to 800 kHz depending on the membrane size. In order to characterize the frequency response of the fabricated sensors up to 1 MHz, new calibration methods based on an N-wave source were designed and tested. Short duration spherical N-waves can be generated by an electric spark source. To estimated a constant sensitivity coefficient, a known method is based on the estimation of the peak pressure from the lengthening of N-waves induced by non linear propagation. However, to obtain the sensitivity as a function of frequency, the output voltage must be compared to the incident pressure waveform, which must be accurately characterized. Taking advantage of recent works on the characterization of pressure N-waves generated by an electric spark source by means of optical methods, two calibration methods have been designed to obtain the frequency response. A method based on the comparison with pressure waveforms deduced from the analysis of schlieren images allowed to estimate the frequency response. A second method, based on a Mach-Zender optical interferometer, was found to be the best method to estimate the sensitivity of microphones up to 1 MHz. The methods were first tested by calibrating standard 1/8 inch condenser microphones. Then, frequency responses of different MEMS microphones prototypes were characterized to test different sensor designs. Results show that using a spark source and optical methods it is possible to calibrate sensors in the frequency range 10 kHz-1 MHz. The new calibration methods were used to improve the design of new high frequency MEMS pressure sensors.

  10. 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.

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

    PubMed Central

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

    2008-01-01

    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

  13. 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

  14. 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.

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

    PubMed

    Chung, King; Mongeau, Luc; McKibben, Nicholas

    2009-04-01

    Wind noise can be a significant problem for hearing instrument users. This study examined the polar characteristics of flow noise at outputs of two behind-the-ear digital hearing aids, and a microphone mounted on the surface of a cylinder at flow velocities ranging from a gentle breeze (4.5 m/s) to a strong gale (22.5 m/s) . The hearing aids were programed in an anechoic chamber, and tested in a quiet wind tunnel for flow noise recordings. Flow noise levels were estimated by normalizing the overall gain of the hearing aids to 0 dB. The results indicated that the two hearing aids had similar flow noise characteristics: The noise level was generally the lowest when the microphone faced upstream, higher when the microphone faced downstream, and the highest for frontal and rearward incidence angles. Directional microphones often generated higher flow noise level than omnidirectional microphones but they could reduce far-field background noise, resulting in a lower ambient noise level than omnidirectional microphones. Data for the academic microphone- on-cylinder configuration suggested that both turbulence and flow impingement might have contributed to the generation of flow noise in the hearing aids. Clinical and engineering design applications are discussed. PMID:19354400

  16. 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.

  17. Experimental Test-Bed for Intelligent Passive Array Research

    NASA Technical Reports Server (NTRS)

    Solano, Wanda M.; Torres, Miguel; David, Sunil; Isom, Adam; Cotto, Jose; Sharaiha, Samer

    2004-01-01

    This document describes the test-bed designed for the investigation of passive direction finding, recognition, and classification of speech and sound sources using sensor arrays. The test-bed forms the experimental basis of the Intelligent Small-Scale Spatial Direction Finder (ISS-SDF) project, aimed at furthering digital signal processing and intelligent sensor capabilities of sensor array technology in applications such as rocket engine diagnostics, sensor health prognostics, and structural anomaly detection. This form of intelligent sensor technology has potential for significant impact on NASA exploration, earth science and propulsion test capabilities. The test-bed consists of microphone arrays, power and signal distribution modules, web-based data acquisition, wireless Ethernet, modeling, simulation and visualization software tools. The Acoustic Sensor Array Modeler I (ASAM I) is used for studying steering capabilities of acoustic arrays and testing DSP techniques. Spatial sound distribution visualization is modeled using the Acoustic Sphere Analysis and Visualization (ASAV-I) tool.

  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. Magnetic arrays

    SciTech Connect

    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.

  20. 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

  1. Superdirective and gradient sensor arrays

    NASA Astrophysics Data System (ADS)

    Merklinger, Harold M.

    2003-10-01

    During the late 1960s and the 1970s, underwater acoustic investigators examined superdirective and gradient sensor systems in order to enhance submarine detection capabilities for surface ships and maritime aircraft. Simple gradient processing had already been used in both in-air acoustic systems (cardioid and super-cardioid microphones) as well as radio and radar applications. Superdirective techniques were known [R. L. Pritchard, J. Acoust. Soc. Am. 25, 879 (1953)] and sometimes exploited in radar systems. It was quickly demonstrated that simple gradient sensors and modest degrees of superdirective array processing were possible, although self-noise and the ability to calibrate hydrophones limited the processing gains achievable. Circular superdirective arrays were used extensively by the Defence Research Establishment Atlantic for noise directionality measurements in the frequency range 4 Hz to about 1 kHz and considered for naval ASW applications until the superiority of oil-filled conventional arrays became apparent. Nevertheless, the significant theoretical and practical development of spatial harmonic beamforming and direction finding was completed. Although much of this work was not considered classified, neither was it widely published. This presentation will review the concepts developed and progress made. Beamforming, noise mitigation and calibration issues are covered.

  2. Optimal design of an electret microphone metal-oxide-semiconductor field-effect transistor preamplifier.

    PubMed

    van der Donk, A G; Bergveld, P

    1992-04-01

    A theoretical noise analysis of the combination of a capacitive microphone and a preamplifier containing a metal-oxide-semiconductor field-effect transistor (MOSFET) and a high-value resistive bias element is given. It is found that the output signal-to-noise ratio for a source follower and for a common-source circuit is almost the same. It is also shown that the output noise can be reduced by making the microphone capacitance as well as the bias resistor as large as possible, and furthermore by keeping the parasitic gate capacitances as low as possible and finally by using an optimum value for the gate area of the MOSFET. The main noise source is the thermal noise of the gate leakage resistance of the MOSFET. It is also shown that short-channel MOSFETs produce more thermal channel noise than longer channel devices. PMID:1597614

  3. Reconstruction of the signal produced by a directional sound source from remote multi-microphone recordings.

    PubMed

    Guarato, Francesco; Hallam, John; Matsuo, Ikuo

    2011-09-01

    A mathematical method for reconstructing the signal produced by a directional sound source from knowledge of the same signal in the far field, i.e., microphone recordings, is developed. The key idea is to compute inverse filters that compensate for the directional filtering of the signal by the sound source directivity, using a least-square error optimization strategy. Previous work pointed out how the method strongly depends on arrival times of signal in the microphone recordings. Two strategies are used in this paper for calculating the time shifts that are afterward taken as inputs, together with source directivity, for the reconstruction. The method has been tested in a laboratory environment, where ground truth was available, with a Polaroid transducer as source. The reconstructions are similar with both strategies. The performance of the method also depends on source orientation. PMID:21895106

  4. In situ tuning of omnidirectional microelectromechanical-systems microphones to improve performance fit in hearing aids

    NASA Astrophysics Data System (ADS)

    Je, Sang-Soo; Kim, Jeonghwan; Harrison, Jere C.; Kozicki, Michael N.; Chae, Junseok

    2008-09-01

    Hearing aids are not a one-size-fits-all solution to hearing problems; they must be uniquely tuned for each wearer. There are currently no low-cost and/or effective methods for in situ tuning. This paper describes a microelectromechanical-systems (MEMS)-based dual omnidirectional microphone that can be tuned by growing metallic nanostructures. The nanostructures are grown on integrated solid electrolyte layers on a suspended parylene diaphragm using an external bias and tune the MEMS microphones in situ thereby limiting mismatch. In our tests, this tuning improved the directivity index from 3.5 (fair directionality) to 4.6 dB (excellent directionality) in normal (room temperature) operating environments.

  5. 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.

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

    NASA Technical Reports Server (NTRS)

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

    1978-01-01

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

  7. 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.

  8. 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. PMID:24907836

  9. 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.

  10. 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

  11. 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.

  12. 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.

  13. 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

  14. 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.

  15. 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.

  16. 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.

  17. 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

  18. 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.

  19. 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.

  20. 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.

  1. Accuracy of pointing a binaural listening array.

    PubMed

    Letowski, T R; Ricard, G L; Kalb, J T; Mermagen, T J; Amrein, K M

    1997-12-01

    We measured the accuracy with which sounds heard over a binaural, end-fire array could be located when the angular separation of the array's two arms was varied. Each individual arm contained nine cardioid electret microphones, the responses of which were combined to produce a unidirectional, band-limited pattern of sensitivity. We assessed the desirable angular separation of these arms by measuring the accuracy with which listeners could point to the source of a target sound presented against high-level background noise. We employed array separations of 30 degrees, 45 degrees, and 60 degrees, and signal-to-noise ratios of +5, -5, and -15 dB. Pointing accuracy was best for a separation of 60 degrees; this performance was indistinguishable from pointing during unaided listening conditions. In addition, the processing of the array was modeled to depict the information that was available for localization. The model indicates that highly directional binaural arrays can be expected to support accurate localization of sources of sound only near the axis of the array. Wider enhanced listening angles may be possible if the forward coverage of the sensor system is made less directional and more similar to that of human listeners. PMID:9473975

  2. 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.

  3. Adaptive feedback active noise control

    NASA Astrophysics Data System (ADS)

    Kuo, Sen M.; Vijayan, Dipa

    Feedforward active noise control (ANC) systems use a reference sensor that senses a reference input to the controller. This signal is assumed to be unaffected by the secondary source and is a good measure of the undesired noise to be cancelled by the system. The reference sensor may be acoustic (e.g., microphone) or non-acoustic (e.g., tachometer, optical transducer). An obvious problem when using acoustic sensors is that the reference signal may be corrupted by the canceling signal generated by the secondary source. This problem is known as acoustic feedback. One way of avoiding this is by using a feedback active noise control (FANC) system which dispenses with the reference sensor. The FANC technique originally proposed by Olson and May employs a high gain negative feedback amplifier. This system suffered from the drawback that the error microphone had to be placed very close to the loudspeaker. The operation of the system was restricted to low frequency range and suffered from instability due to the possibility of positive feedback. Feedback systems employing adaptive filtering techniques for active noise control were developed. This paper presents the FANC system modeled as an adaptive prediction scheme.

  4. 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

  5. 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

  6. Kokkos Array

    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.

  7. 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.

  8. 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.

  9. 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.

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

    PubMed

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

    2015-11-01

    The objective of this study is to evaluate the reliability of acoustic voice parameters obtained using smart phone (SP) microphones and investigate the utility of use of SP voice recordings for voice screening. Voice samples of sustained vowel/a/obtained from 118 subjects (34 normal and 84 pathological voices) were recorded simultaneously through two microphones: oral AKG Perception 220 microphone and SP Samsung Galaxy Note3 microphone. Acoustic voice signal data were measured for fundamental frequency, jitter and shimmer, normalized noise energy (NNE), signal to noise ratio and harmonic to noise ratio using Dr. Speech software. Discriminant analysis-based Correct Classification Rate (CCR) and Random Forest Classifier (RFC) based Equal Error Rate (EER) were used to evaluate the feasibility of acoustic voice parameters classifying normal and pathological voice classes. Lithuanian version of Glottal Function Index (LT_GFI) questionnaire was utilized for self-assessment of the severity of voice disorder. The correlations of acoustic voice parameters obtained with two types of microphones were statistically significant and strong (r = 0.73-1.0) for the entire measurements. When classifying into normal/pathological voice classes, the Oral-NNE revealed the CCR of 73.7% and the pair of SP-NNE and SP-shimmer parameters revealed CCR of 79.5%. However, fusion of the results obtained from SP voice recordings and GFI data provided the CCR of 84.60% and RFC revealed the EER of 7.9%, respectively. In conclusion, measurements of acoustic voice parameters using SP microphone were shown to be reliable in clinical settings demonstrating high CCR and low EER when distinguishing normal and pathological voice classes, and validated the suitability of the SP microphone signal for the task of automatic voice analysis and screening. PMID:26162450

  11. 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.

  12. 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.

  13. Evaluation of an adaptive directional system in a DSP hearing aid.

    PubMed

    Bentler, Ruth A; Tubbs, Jill L; Egge, Jessica L M; Flamme, Gregory A; Dittberner, Andrew B

    2004-06-01

    The effectiveness of an adaptive directional microphone design, as implemented in the Phonak Claro behind-the-ear hearing aid, is evaluated. Participants were fit bilaterally and tested in 2 environments, an anechoic chamber and a moderately reverberant classroom, with the microphones in the fixed (cardioid) setting and the adaptive setting. Five speakers were placed between 110 degrees and 250 degrees azimuth around the listener. Speech-weighted noise was presented from those speakers at an overall level (OAL) of 65 dB (A). Noise was increased by 8 dB from 1 speaker at a time, using 2-s modulation and random assignment, while the output from the other speakers was reduced to maintain the constant OAL. Results of 2 speech perception tasks used as outcome measures indicated that the adaptive system was not able to follow the dominant noise source in the presence of lower level noise sources. Self-report measures obtained after blinded home trials were consistent with laboratory findings that the participants did not perceive this adaptive microphone design to be more effective than the default fixed-microphone option. PMID:15248806

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

    PubMed Central

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

    2014-01-01

    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

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

    NASA Technical Reports Server (NTRS)

    Fields, Richard S., Jr.

    1995-01-01

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

  16. 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

  17. 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

  18. 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

  19. 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

  20. 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.

  1. Unidirectional Mechanical Amplification as a Design Principle for an Active Microphone

    NASA Astrophysics Data System (ADS)

    Reichenbach, Tobias; Hudspeth, A. J.

    2011-04-01

    Amplification underlies the operation of many biological and engineering systems. Simple electrical, optical, and mechanical amplifiers are reciprocal: the backward coupling of the output to the input equals the forward coupling of the input to the output. Unidirectional amplifiers that occur often in electrical and optical systems are special nonreciprocal devices in which the output does not couple back to the input even though the forward coupling persists. Here we propose a scheme for unidirectional mechanical amplification that we utilize to construct an active microphone. We show that amplification improves the microphone’s threshold for detecting weak signals and that unidirectionality prevents distortion.

  2. 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.

  3. MUSE - a systolic array for adaptive nulling with 64 degrees of freedom, using Givens transformations and wafer-scale integration. Technical report

    SciTech Connect

    Rader, C.M.; Allen, D.L.; gLASCO , D.B.; Woodward, C.E.

    1990-05-18

    This report describes an architecture for a highly parallel system of computational processors specialized for real-time adaptive antenna nulling computations with many degrees of freedom, which we call MUSE (Matrix Update Systolic Experiment), and a specific realization of MUSE for 64 degrees of freedom. Each processor uses the CORDIC algorithm and has been designed as a single integrated circuit. Ninety-six such processors working together can update the 64-element nulling weights based on 300 new observations in only 6.7 milliseconds. This is equivalent to 2.88 Giga-ops for a conventional processor. The computations are accurate enough to support 50 decibel of signal-to-noise improvement in a sidelobe canceller. The connectivity between processors is quite simple and permits MUSE to be realized on a single large wafer, using restructurable VLSI (Very Large Scale Integration). The complete design of such a wafer is described.

  4. A Broadly Adaptive Array of Dose-Constraint Templates for Planning of Intensity-Modulated Radiation Therapy for Advanced T-Stage Nasopharyngeal Carcinoma

    SciTech Connect

    Chau, R.M.-C. Leung, S.-F.; Kam, M.K.-M.; Cheung, K.-Y.; Kwan, W.-H.; Yu, K.-H.; Chiu, K.-W.; Cheung, M.L.-M.; Chan, A.T.-C.

    2009-05-01

    Purpose: To develop and validate adaptive dose-constraint templates in intensity-modulated radiotherapy (IMRT) planning for advanced T-stage nasopharyngeal carcinoma (NPC). Method and Materials: Dose-volume histograms of clinically approved plans for 20 patients with advanced T-stage NPC were analyzed, and the pattern of distribution in relation to the degree of overlap between targets and organs at risk (OARs) was explored. An adaptive dose constraint template (ADCT) was developed based on the degree of overlap. Another set of 10 patients with advanced T-stage NPC was selected for validation. Results of the manual arm optimization protocol and the ADCT optimization protocol were compared with respect to dose optimization time, conformity indices, multiple-dose end points, tumor control probability, and normal tissue complication probability. Results: For the ADCT protocol, average time required to achieve an acceptable plan was 9 minutes, with one optimization compared with 94 minutes with more than two optimizations of the manual arm protocol. Target coverage was similar between the manual arm and ADCT plans. A more desirable dose distribution in the region of overlap between planning target volume and OARs was achieved in the ADCT plan. Dose end points of OARs were similar between the manual arm and ADCT plans. Conclusions: With the developed ADCT, IMRT treatment planning becomes more efficient and less dependent on the planner's experience on dose optimization. The developed ADCT is applicable to a wide range of advanced T-stage NPC treatment and has the potential to be applied in a broader context to IMRT planning for other cancer sites.

  5. A 540-μW digital pre-amplifier with 88-dB dynamic range for electret microphones

    NASA Astrophysics Data System (ADS)

    Yan, Liu; Siliang, Hua; Donghui, Wang; Chaohuan, Hou

    2009-05-01

    We design a digital pre-amplifier which can be directly connected to an electret microphone. The amplifier can convert analog signals into digital signals, has a wide voltage swing and low power consumption, as is required in portable applications. Measurement results show that the dynamic range of the digital pre-amplifier reaches 88 dB, the equivalent input referred noise is 5 μVrms, the typical power consumption is 540 μW, and in standby mode the current does not exceed 10 μA. Compared with an analog microphone, an electret microphone with digital pre-amplifier offers a better SNR, higher integration, lower power consumption, and higher immunity to system noise.

  6. 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

  7. 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

  8. 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.

  9. 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.

  10. Dosimetry measurements using a probe tube microphone in the ear canal.

    PubMed

    Shotland, L I

    1996-02-01

    Federal and international standards recommended use of microphone placement either on or in the vicinity of the shoulder for dosimetry to minimize deviations from the undisturbed sound field. Probe microphone measurements from the ear canal were compared to shoulder and chest measures in order to investigate the validity of current dosimetry methodologies. Six subjects were monitored in an industrial setting. As expected, ear-canal levels exceeded other measures for all subjects. Shoulder and chest measures showed very low intersubject variability whereas ear-canal measures resulted in large intersubject variability. The ear-canal methodology has the potential to identify individuals whose external ear gain exceed the mean, putting them at increased risk of noise-induced permanent threshold shift (NIPTS). It is proposed that overall external ear pressure gain be used as an index to adjust exposure levels when predicting NIPTS using ISO 1999. A normative database of external ear pressure gain was constructed from 30 ears for this purpose. PMID:8609306

  11. 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.

  12. A BIPM/CIPM key comparison on microphone calibration--defining the state of the art

    NASA Astrophysics Data System (ADS)

    Barham, Richard G.

    2002-11-01

    The formation of the Consultative Committee on Acoustics, Ultrasound and Vibration (CCAUV) now gives acoustical quantities the same status as more established metrics like mass or voltage. The principle work of CCAUV is to establish degrees of equivalence between member states of the Metre Convention. This is achieved through key comparisons and subsequent regional comparison. For sound-in-air, the subject of this key comparison is most appropriately the calibration of laboratory standard microphones. The first such project considered the pressure calibration of IEC type LS1P microphones in the frequency range from 63 Hz to 8 kHz. Twelve national laboratories took part in the key comparison, piloted by the UK's National Physical Laboratory. The project has now been completed. The results for the measured pressure sensitivity level have a standard deviation of around 0.02 dB at frequencies below 1 kHz, rising to a maximum of 0.04 dB at 8 kHz. The mean of these results normalized to 0 dB at each frequency is considered as the key comparison reference value (KCRV) and the standard deviation provides an estimate of its standard uncertainty. The KCRV then defines the datum which enables the performance of all laboratories to be related.

  13. Two-Microphone Noise Reduction Using Spatial Information-Based Spectral Amplitude Estimation

    NASA Astrophysics Data System (ADS)

    Li, Kai; Guo, Yanmeng; Fu, Qiang; Li, Junfeng; Yan, Yonghong

    Traditional two-microphone noise reduction algorithms to deal with highly nonstationary directional noises generally use the direction of arrival or phase difference information. The performance of these algorithms deteriorate when diffuse noises coexist with nonstationary directional noises in realistic adverse environments. In this paper, we present a two-channel noise reduction algorithm using a spatial information-based speech estimator and a spatial-information-controlled soft-decision noise estimator to improve the noise reduction performance in realistic non-stationary noisy environments. A target presence probability estimator based on Bayes rules using both phase difference and magnitude squared coherence is proposed for soft-decision of noise estimation, so that they can share complementary advantages when both directional noises and diffuse noises are present. Performances of the proposed two-microphone noise reduction algorithm are evaluated by noise reduction, log-spectral distance (LSD) and word recognition rate (WRR) of a distant-talking ASR system in a real room's noisy environment. Experimental results show that the proposed algorithm achieves better noises suppression without further distorting the desired signal components over the comparative dual-channel noise reduction algorithms.

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

    PubMed Central

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

    2008-01-01

    In many micromachined sensors the thin (2–10 μm thick) air film between a compliant diaphragm and backplate electrode plays a dominant role in shaping both the dynamic and thermal noise characteristics of the device. Silicon microphone structures used in grating-based optical-interference microphones have recently been introduced that employ backplates with minimal area to achieve low damping and low thermal noise levels. Finite-element based modeling procedures based on 2-D discretization of the governing Reynolds equation are ideally suited for studying thin-film dynamics in such structures which utilize relatively complex backplate geometries. In this paper, the dynamic properties of both the diaphragm and thin air film are studied using a modal projection procedure in a commonly used finite element software and the results are used to simulate the dynamic frequency response of the coupled structure to internally generated electrostatic actuation pressure. The model is also extended to simulate thermal mechanical noise spectra of these advanced sensing structures. In all cases simulations are compared with measured data and show excellent agreement—demonstrating 0.8 pN/√Hz and 1.8 μPa/√Hz thermal force and thermal pressure noise levels, respectively, for the 1.5 mm diameter structures under study which have a fundamental diaphragm resonance-limited bandwidth near 20 kHz. PMID:19081811

  15. 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

  16. 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.

  17. Three-dimensional sound localization from a compact non-coplanar array of microphones using tree-based learning.

    PubMed

    Weng, J; Guentchev, K Y

    2001-07-01

    One of the various human sensory capabilities is to identify the direction of perceived sounds. The goal of this work is to study sound source localization in three dimensions using some of the most important cues the human uses. In an attempt to satisfy the requirements of portability and miniaturization in robotics, this approach employs a compact sensor structure that can be placed on a mobile platform. The objective is to estimate the relative sound source position in three-dimensional space without imposing excessive restrictions on its spatio-temporal characteristics and the environment structure. Two types of features are considered, interaural time and level differences. Their relative effectiveness for localization is studied, as well as a practical way of using these complementary parameters. A two-stage procedure was used. In the training stage, sound samples are produced from points with known coordinates and then are stored. In the recognition stage, unknown sounds are processed by the trained system to estimate the 3D location of the sound source. Results from the experiments showed under +/-3 degrees in average angular error and less than +/-20% in average radial distance error. PMID:11508957

  18. Comparison of voice relative fundamental frequency estimates derived from an accelerometer signal and low-pass filtered and unprocessed microphone signals

    PubMed Central

    Lien, Yu-An S.; Stepp, Cara E.

    2014-01-01

    The relative fundamental frequency (RFF) surrounding the production of a voiceless consonant has previously been estimated using unprocessed and low-pass filtered microphone signals, but it can also be estimated using a neck-placed accelerometer signal that is less affected by vocal tract formants. Determining the effects of signal type on RFF will allow for comparisons across studies and aid in establishing a standard protocol with minimal within-speaker variability. Here RFF was estimated in 12 speakers with healthy voices using unprocessed microphone, low-pass filtered microphone, and unprocessed accelerometer signals. Unprocessed microphone and accelerometer signals were recorded simultaneously using a microphone and neck-placed accelerometer. The unprocessed microphone signal was filtered at 350 Hz to construct the low-pass filtered microphone signal. Analyses of variance showed that signal type and the interaction of vocal cycle × signal type had significant effects on both RFF means and standard deviations, but with small effect sizes. The overall RFF trend was preserved regardless of signal type and the intra-speaker variability of RFF was similar among the signal types. Thus, RFF can be estimated using either a microphone or an accelerometer signal in individuals with healthy voices. Future work extending these findings to individuals with disordered voices is warranted. PMID:24815277

  19. Helicopter and aircraft detection and classification using adaptive beamforming and tracking techniques

    NASA Astrophysics Data System (ADS)

    van Koersel, Antonius C.; Beerens, S. P.

    2002-08-01

    Measurements of different types of aircraft are performed and used to obtain information on target characteristics and develop an algorithm to perform classification between jet aircraft, propeller aircraft and helicopters. To obtain a larger detection range, reduce background noise and to reduce classification errors in a multi-target environment, a real time adaptive beamformer algorithm is developed for a three microphone array. The output of the beamformer is submitted to a tracking algorithm. Acoustic signals from identified tracks are submitted to the classification algorithms. The algorithm is tested on data recorded during various field trials. The objective of the research, which is part of a research program for the Dutch Army, is to detect the passage of an aircraft with one or more mechanical wave sensors, either acoustic or seismic. After detection of a target, classification of the type of aircraft is requested (for example: helicopter-jet-propeller-rpv). If possible type identification is also requested. Earlier work showed promising results for detection and classification of helicopter targets. The projects resulted in an algorithm that can detect and classify helicopters, but it was developed to reject other targets. The chosen approach is to combine new aircraft detection and beamforming algorithms with the existing algorithms.

  20. 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-. PMID:21246202

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

    PubMed Central

    Falk, Ben; Williams, Tameeka; Aytekin, Murat

    2011-01-01

    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−. PMID:21246202

  2. 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.

  3. 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.

  4. Interagency arraying

    NASA Astrophysics Data System (ADS)

    Cox, Henry G.

    Activities performed to match ground aperture requirements for the Neptune encounter in August 1989 with the expected capabilities of the JPL Deep Space Network (DSN) are discussed. Ground aperture requirements, DSN capabilities, and the capabilities of other agencies are reviewed. The design and configurations of the receiver subsystem, combiner subsystem, monitor and control subsystem, recording subsystem, and supporting subsystems are described. The implementation of the Very Large Array-Goldstone Telemetry Array is discussed, and the differences involved with the Parkes-Canberra Telemetry Array implementation are highlighted. The operational concept is addressed.

  5. 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.

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

    PubMed

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

    2006-01-01

    A free-field (FF) substitution method for calibrating the pressure sensitivity of microphones at frequencies up to 80 kHz is demonstrated with both grazing and normal-incidence geometries. The substitution-based method, as opposed to a simultaneous method, avoids problems associated with the nonuniformity of the sound field and, as applied here, uses a 1/4-in. air-condenser pressure microphone as a known reference. Best results were obtained with a centrifugal fan, which is used as a random, broadband sound source. A broadband source minimizes reflection-related interferences that can plague FF measurements. Calibrations were performed on 1/4-in. FF air-condenser, electret, and microelectromechanical systems (MEMS) microphones in an anechoic chamber. The uncertainty of this FF method is estimated by comparing the pressure sensitivity of an air-condenser FF microphone, as derived from the FF measurement, with that of an electrostatic actuator calibration. The root-mean-square difference is found to be +/- 0.3 dB over the range 1-80 kHz, and the combined standard uncertainty of the FF method, including other significant contributions, is +/- 0.41 dB. PMID:16454287

  7. 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

  8. 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…

  9. 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

  10. 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

  11. Enthalpy arrays

    NASA Astrophysics Data System (ADS)

    Torres, Francisco E.; Kuhn, Peter; de Bruyker, Dirk; Bell, Alan G.; Wolkin, Michal V.; Peeters, Eric; Williamson, James R.; Anderson, Gregory B.; Schmitz, Gregory P.; Recht, Michael I.; Schweizer, Sandra; Scott, Lincoln G.; Ho, Jackson H.; Elrod, Scott A.; Schultz, Peter G.; Lerner, Richard A.; Bruce, Richard H.

    2004-06-01

    We report the fabrication of enthalpy arrays and their use to detect molecular interactions, including protein-ligand binding, enzymatic turnover, and mitochondrial respiration. Enthalpy arrays provide a universal assay methodology with no need for specific assay development such as fluorescent labeling or immobilization of reagents, which can adversely affect the interaction. Microscale technology enables the fabrication of 96-detector enthalpy arrays on large substrates. The reduction in scale results in large decreases in both the sample quantity and the measurement time compared with conventional microcalorimetry. We demonstrate the utility of the enthalpy arrays by showing measurements for two protein-ligand binding interactions (RNase A + cytidine 2'-monophosphate and streptavidin + biotin), phosphorylation of glucose by hexokinase, and respiration of mitochondria in the presence of 2,4-dinitrophenol uncoupler.

  12. Nonlinear photoacoustic response of opaque media in gas microphone signal detection

    NASA Astrophysics Data System (ADS)

    Madvaliev, U.; Salikhov, T. Kh.; Sharifov, D. M.; Khan, N. A.

    2006-03-01

    We have theoretically studied the effect of thermal nonlinearity, due to the temperature dependence of the thermophysical and optical parameters for thermally thick opaque media, on the characteristics of the fundamental photoacoustic signal when the signal is detected by a gas microphone. We have shown that the dependence of the amplitude of the nonlinear component of the signal on the intensity of the incident radiation I0 is expressed by means of the dependence of the temperature rise for the irradiated sample surface Θ0 on I0, and the thermal nonlinearity does not affect the phase of the photoacoustic signal. We propose a theory for generation of the second harmonic of the photoacoustic signal. We have established that the phase shift of the photoacoustic signal is equal to 3π/4, while its amplitude depends on the frequency (˜ω-3/2) and the intensity (˜ I{0/2}).

  13. 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.

  14. 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.

  15. 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.

  16. 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.

  17. 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

  18. Detecting impacts of sand grains with a microphone system in field conditions

    NASA Astrophysics Data System (ADS)

    Ellis, Jean T.; Morrison, Rebecca F.; Priest, Barry H.

    2009-04-01

    This paper describes the "miniphone," an instrument to measure aeolian saltation. This instrument is a modified electret microphone that detects the impacts of individual grains. The unidirectional miniphone is inexpensive (approximately US$10), small, and poses minimal disruption to the wind field. It can be sampled at rates up to 44,100 Hz using commonly available sound card technology or it can be interfaced with a data acquisition system. Data from deployments on beaches on Marco Island, FL, USA and near Shoalhaven Heads, NSW, Australia using sample rates of 44,100 Hz and 6000 Hz, are presented. An algorithm for identifying discrete impacts of grains is described. The number of saltation impacts was not reduced when sub-sampling a record from 44,100 Hz to 6000 Hz. The most immediate use for the miniphone is for short-term deployments to detect unsteadiness in the saltation field.

  19. 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

  20. Array tomography: production of arrays.

    PubMed

    Micheva, Kristina D; O'Rourke, Nancy; Busse, Brad; Smith, Stephen J

    2010-11-01

    Array tomography is a volumetric microscopy method based on physical serial sectioning. Ultrathin sections of a plastic-embedded tissue are cut using an ultramicrotome, bonded in an ordered array to a glass coverslip, stained as desired, and imaged. The resulting two-dimensional image tiles can then be reconstructed computationally into three-dimensional volume images for visualization and quantitative analysis. The minimal thickness of individual sections permits high-quality rapid staining and imaging, whereas the array format allows reliable and convenient section handling, staining, and automated imaging. Also, the physical stability of the arrays permits images to be acquired and registered from repeated cycles of staining, imaging, and stain elution, as well as from imaging using multiple modalities (e.g., fluorescence and electron microscopy). Array tomography makes it possible to visualize and quantify previously inaccessible features of tissue structure and molecular architecture. However, careful preparation of the tissue is essential for successful array tomography; these steps can be time consuming and require some practice to perfect. This protocol describes the sectioning of embedded tissues and the mounting of the serial arrays. The procedures require some familiarity with the techniques used for ultramicrotome sectioning for electron microscopy. PMID:21041397

  1. Array tomography: imaging stained arrays.

    PubMed

    Micheva, Kristina D; O'Rourke, Nancy; Busse, Brad; Smith, Stephen J

    2010-11-01

    Array tomography is a volumetric microscopy method based on physical serial sectioning. Ultrathin sections of a plastic-embedded tissue are cut using an ultramicrotome, bonded in an ordered array to a glass coverslip, stained as desired, and imaged. The resulting two-dimensional image tiles can then be reconstructed computationally into three-dimensional volume images for visualization and quantitative analysis. The minimal thickness of individual sections permits high-quality rapid staining and imaging, whereas the array format allows reliable and convenient section handling, staining, and automated imaging. Also, the physical stability of the arrays permits images to be acquired and registered from repeated cycles of staining, imaging, and stain elution, as well as from imaging using multiple modalities (e.g., fluorescence and electron microscopy). Array tomography makes it possible to visualize and quantify previously inaccessible features of tissue structure and molecular architecture. However, careful preparation of the tissue is essential for successful array tomography; these steps can be time-consuming and require some practice to perfect. In this protocol, tissue arrays are imaged using conventional wide-field fluorescence microscopy. Images can be captured manually or, with the appropriate software and hardware, the process can be automated. PMID:21041399

  2. 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. PMID:26553758

  3. A directional array approach for the measurement of rotor noise source distributions with controlled spatial resolution

    NASA Technical Reports Server (NTRS)

    Brooks, T. F.; Marcolini, M. A.; Pope, D. S.

    1987-01-01

    A special array system has been designed to examine noise source distributions over a helicopter rotor model. The particular measurement environment is for a rotor operating in the open jet of an anechoic wind tunnel. An out-of-flow directional microphone element array is used with a directivity pattern whose major directional lobe projects on the rotor disk. If significant contributions from extraneous tunnel noise sources in the direction of the side lobes are excluded, the dominant output from the array would be that noise emitted from the projected area on the rotor disk. The design incorporates an array element signal blending features which serves to control the spatial resolution of the size of the directional lobes. (Without blending, the resolution and side lobe size are very strong functions of frequency, which severely limits the array's usefulness).

  4. 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.

  5. 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.

  6. 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.

  7. 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.

  8. 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.

  9. Global Arrays

    SciTech Connect

    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 shared 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).

  10. 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.

  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. 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.

  13. 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.

  14. [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.

  15. 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.

  16. 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

  17. 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.

  18. 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.

  19. 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.

  20. 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. PMID:23419032

  1. Cochlear microphonics in sensorineural hearing loss: lesson from newborn hearing screening.

    PubMed

    Ahmmed, Ansar; Brockbank, Christopher; Adshead, June

    2008-08-01

    The diagnostic dilemma surrounding the presence of cochlear microphonics (CM) coupled with significantly elevated auditory brainstem response (ABR) thresholds in babies failing the newborn hearing screening is highlighted. A case report is presented where initial electo-diagnostic assessment could not help in differentiating between Auditory Neuropathy/Auditory Dys-synchrony (AN/AD) and sensorineural hearing loss (SNHL). In line with the protocol and guidelines provided by the national Newborn Hearing Screening Programme in the UK (NHSP) AN/AD was suspected in a baby due to the presence of CM at 85 dBnHL along with click evoked ABR thresholds of 95 dBnHL in one ear and 100 dBnHL in the other ear. Significantly elevated thresholds for 0.5 and 1kHz tone pip ABR fulfilled the audiological diagnostic criteria for AN/AD. However, the possibility of a SNHL could not be ruled out as the 85 dBnHL stimuli presented through inserts for the CM would have been significantly enhanced in the ear canals of the young baby to exceed the threshold level of the ABR that was carried out using headphones. SNHL was eventually diagnosed through clinical and family history, physical examination and imaging that showed enlarged vestibular aqueducts. Presence of CM in the presence of very high click ABR thresholds only suggests a pattern of test results and in such cases measuring thresholds for 0.5 and 1 kHz tone pip ABR may not be adequate to differentiate between SNHL and other conditions associated with AN/AD. There is a need for reviewing the existing AN/AD protocol from NHSP in the UK and new research to establish parameters for CM to assist in the differential diagnosis. A holistic audiological and medical approach is essential to manage babies who fail the newborn hearing screening. PMID:18571245

  2. 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

  3. 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

  4. A new kind of sensor array for measuring spatial coherence of surface pressure on a car's side window

    NASA Astrophysics Data System (ADS)

    Gabriel, C.; Müller, S.; Ullrich, F.; Lerch, R.

    2014-02-01

    The mechanical behavior of a car's side window and the resulting acoustic radiation into the cabin is mainly affected by the spatial coherence of the surface pressure exciting the glass plate. The surface pressure is a superposition of hydrodynamic and acoustic pressure whose levels differ by 2 or 3 orders of magnitude. To gain information about the coherence characteristics of the surface pressure and to separate its hydrodynamic and acoustic components, a measurement of high spatial resolution is needed. For that reason a novel pressure transducer array with a minimum distance between two adjacent measurement points of only 2 mm was developed. The pressure transducers of the array are arranged sparsely on a grid while all possible distances between the spots on the grid are covered. Due to this minimization of distance redundancy, the amount of microphones could be reduced from 1849 to 92 representing a virtual array of 43×43 measurement positions. A Nyquist wavenumber of 250 1/m and a resolution of 11.9 1/m using a sensor area of only 52×52 mm2 were achieved. Because of its small dimensions, this array allows for measurements at various test areas on the side window, which is a major improvement compared to former investigations. For the measurements conventional MEMS microphones are applied. It is shown that the used microphones are suitable for the requisite, even if operating in saturation. Hence, the existence of acoustic loads on the side window and the position-dependent spatial coherence of the surface pressure can be studied. Measurements using the sensor array are carried out in the anechoic wind tunnel to validate the methodology. Results of the separation between hydrodynamic and acoustic pressure as well as the identification of coherence properties are presented.

  5. 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.

  6. 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.

  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 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

  9. 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

  10. 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.

  11. 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.

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

    PubMed

    Ando, Shigeru; Nara, Takaaki; Levy, Tsukassa

    2013-10-01

    Wave source localization from a sensor array has long been the most active research topics in both theory and application. In this paper, an explicit and time-domain inversion method for the direction and distance of a monopole source from a circular array is proposed. The approach is based on a mathematical technique, the weighted integral method, for signal/source parameter estimation. It begins with an exact form of the source-constraint partial differential equation that describes the unilateral propagation of wide-band waves from a single source, and leads to exact algebraic equations that include circular Fourier coefficients (phase mode measurements) as their coefficients. From them, nearly closed-form, single-shot and multishot algorithms are obtained that is suitable for use with band-pass/differential filter banks. Numerical evaluation and several experimental results obtained using a 16-element circular microphone array are presented to verify the validity of the proposed method. PMID:24116418

  13. 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.

  14. 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.

  15. 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.

  16. Causality correlation analysis on a cold jet by means of simultaneous particle image velocimetry and microphone measurements

    NASA Astrophysics Data System (ADS)

    Henning, Arne; Koop, Lars; Schröder, Andreas

    2013-06-01

    The aeroacoustic sound generation processes of a cold jet were investigated by means of simultaneous Particle-Image-Velocimetry (PIV) in the near-field and microphone measurements in the far field. The measurements were conducted in a synchronized manner so as to enable the calculation of the cross correlation coefficient between the acoustic pressure and flow quantities derived from the measured velocity fluctuations. In this manner the regularities in the near-field fluctuations which are related to the radiated sound field could be identified. The tests were run for different Mach numbers M=0.5, 0.7 and 0.9 in air. Five microphones arranged at angles of 24°<θ<88° to the jet centre line were used. The correlation with the axial velocity was highest. The spatial as well as the temporal distribution of the coefficient was observed to be dominated by coherent structures and showed a strong dependency on M and θ. For the M=0.9 jet the correlation coefficients fell below the confidence interval for θ=67°.

  17. Wind noise in hearing aids with directional and omnidirectional microphones: Polar characteristics of custom-made hearing aids.

    PubMed

    Chung, King; McKibben, Nicholas; Mongeau, Luc

    2010-04-01

    The purpose of this study was to examine the characteristics of wind noise at the output of in-the-ear, in-the-canal, and completely-in-the-canal hearing aids. The hearing aids were programed to have linear amplification with matching flat frequency responses for directional (DIR) and omnidirectional (OMNI) microphones. The microphone output was then recorded in a quiet wind tunnel when the Knowles electronic manikin for acoustic research (KEMAR) head was turned from 0 degrees to 360 degrees . The overall, 125, 500, and 2000 Hz one-third octave band flow noise levels were calculated and plotted in polar patterns. Correlation coefficients, average differences, and level differences between DIR and OMNI were also calculated. Flow noise levels were the highest when KEMAR was facing the direction of the flow and angles between 190 degrees and 250 degrees . The noise levels were the lowest when the hearing aids were facing the direction of the flow. The polar patterns of DIR and OMNI had similar shapes and DIR generally had higher levels than OMNI. DIR, however, could have lower levels than OMNI in some angles because of its capability to reduce noise in the far field. Comparisons of polar characteristics with behind-the-ear hearing aids, and clinical and engineering design applications of current results are discussed. PMID:20370035

  18. 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. PMID:24907805

  19. 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.

  20. 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.