Science.gov

Sample records for adaptive microphone array

  1. A miniaturized adaptive microphone array under directional constraint utilizing aggregated microphones.

    PubMed

    Matsumoto, Mitsuharu; Hashimoto, Shuji

    2006-01-01

    This paper introduces a miniaturized microphone array using the Directionally Constrained Minimization of Power (DCMP) method, which utilizes the transfer functions of microphones located at the same place, namely aggregated microphones. The phased microphone array realizes a noise reduction and direction of arrival (DOA) estimation system according to differences in the arrival time, phase shift, and/or the level of the sound wave for each microphone. Hence it is difficult to miniaturize the microphone array. The objective of our research is to miniaturize the system size using aggregated microphones. In this paper, we first show that the phased microphone array system and the proposed aggregated microphone system can be described within the same framework. We then apply a microphone array under directional constraint to the aggregated microphones and compare the proposed method with the microphone array. We show the directional pattern of the aggregated microphones. We also show the experimental results regarding DOA estimation.

  2. Adaptive phase calibration of a microphone array for acoustic holography.

    PubMed

    Teal, Paul D; Poletti, Mark A

    2010-04-01

    Previous work has indicated that a limitation on the performance of a circular microphone array for holographic sound field recording at low frequencies is phase mismatch between the microphones in the array. At low frequencies these variations become more significant than at mid-range and high frequencies because the high order phase mode responses at low frequencies are lower in amplitude. This paper demonstrates the feasibility of a "self-calibration" method. The basis of the calibration is to estimate the location of one or more wide-band sources using mid-range frequencies and to use this source location information to perform correction to the array at low frequencies. In its simplest form the calibration must be performed in an anechoic environment, since multipath effects at widely differing frequencies are uncorrelated. The approach is first demonstrated in such an environment using recordings from an array of high quality microphones. The technique is then extended to an adaptive calibration that can be used in an environment that is somewhat reverberant. The validity of the adaptive approach is demonstrated using recordings from an array of inexpensive microphones.

  3. 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.

  4. 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.

  5. Microphone array based novel infant deafness detector.

    PubMed

    Agnihotri, Chinmayee; Thiyagarajan, S; Kalyansundar, Archana

    2010-01-01

    This work focuses on an infant deafness detector unit, using the concept of microphone array. This instrument is based on the principle of evoked acoustic emissions (OAEs). The key feature of the microphone array is its ability to increase signal-to-noise ratio (SNR) and reproducibility of the OAE responses. These further significantly contribute to improve the sensitivity and specificity of the overall system. Low level sound pressure values are recorded by the sensitive microphones in microphone array unit and processed using TI's DSP6416. The sound stimulus transmitted to human ear is generated and controlled by the 6416 DSP (Digital signal processor). Hardware circuit details and the algorithm used in signal processing are discussed in this paper. Standard averaging technique is used in the implemented algorithm. The final result speaks about the hearing capacity of a patient. The proof that the usage of microphone arrays leads to better SNR values than using a single microphone in an OAE probe, is successfully carried out in this work.

  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. Acoustic positioning using multiple microphone arrays.

    PubMed

    Liu, Hui; Milios, Evangelos

    2005-05-01

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

  8. Design theory of microphone arrays for teleconferencing

    NASA Astrophysics Data System (ADS)

    Macomber, Dwight Frank

    2001-07-01

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

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

    NASA Astrophysics Data System (ADS)

    Wu, Hongyan; Wang, Jian

    2015-05-01

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

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

    NASA Astrophysics Data System (ADS)

    Sanz-Robinson, Josue

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

  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. Design of robust differential microphone arrays with orthogonal polynomials.

    PubMed

    Pan, Chao; Benesty, Jacob; Chen, Jingdong

    2015-08-01

    Differential microphone arrays have the potential to be widely deployed in hands-free communication systems thanks to their frequency-invariant beampatterns, high directivity factors, and small apertures. Traditionally, they are designed and implemented in a multistage way with uniform linear geometries. This paper presents an approach to the design of differential microphone arrays with orthogonal polynomials, more specifically with Jacobi polynomials. It first shows how to express the beampatterns as a function of orthogonal polynomials. Then several differential beamformers are derived and their performance depends on the parameters of the Jacobi polynomials. Simulations show the great flexibility of the proposed method in terms of designing any order differential microphone arrays with different beampatterns and controlling white noise gain.

  13. 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.

  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.

  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. 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.

  18. 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?

  19. Acoustic Detection and Tracking of a Class I UAS with a Small Tetrahedral Microphone Array

    DTIC Science & Technology

    2014-09-01

    Acoustic Detection and Tracking of a Class I UAS with a Small Tetrahedral Microphone Array by Minas Benyamin and Geoffrey H Goldman ARL...20783-1138 ARL-TR-7086 September 2014 Acoustic Detection and Tracking of a Class I UAS with a Small Tetrahedral Microphone Array Minas...with a Small Tetrahedral Microphone Array 5a. CONTRACT NUMBER 5b. GRANT NUMBER 5c. PROGRAM ELEMENT NUMBER 6. AUTHOR(S) Minas Benyamin and

  20. 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.

  1. On the design and implementation of linear differential microphone arrays.

    PubMed

    Chen, Jingdong; Benesty, Jacob; Pan, Chao

    2014-12-01

    Differential microphone array (DMA), a particular kind of sensor array that is responsive to the differential sound pressure field, has a broad range of applications in sound recording, noise reduction, signal separation, dereverberation, etc. Traditionally, an Nth-order DMA is formed by combining, in a linear manner, the outputs of a number of DMAs up to (including) the order of N - 1. This method, though simple and easy to implement, suffers from a number of drawbacks and practical limitations. This paper presents an approach to the design of linear DMAs. The proposed technique first transforms the microphone array signals into the short-time Fourier transform (STFT) domain and then converts the DMA beamforming design to simple linear systems to solve. It is shown that this approach is much more flexible as compared to the traditional methods in the design of different directivity patterns. Methods are also presented to deal with the white noise amplification problem that is considered to be the biggest hurdle for DMAs, particularly higher-order implementations.

  2. 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.

  3. 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.

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

    NASA Astrophysics Data System (ADS)

    Rafaely, Boaz; Park, Munhum

    2004-05-01

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

  5. Microphone matching for hybrid-order directional arrays in hearing aid applications

    NASA Astrophysics Data System (ADS)

    Warren, Daniel M.; Thompson, Steve C.

    2003-04-01

    The ability of a hearing aid user to distinguish a single speech source amidst general background noise (for example, dinner table or cocktail party conversation) may be improved by a directional array of microphones in the hearing instrument. The theoretical maximum directivity index (DI) of a first-order pairing of microphones is 6 dB, and a second-order array of three microphones is 9.5 dB, assuming all three microphones have identical frequency responses. The close spacing of microphone ports in a hearing aid body means that directivity degrades rapidly with differences in microphone sensitivities. A hybrid of first- and second-order arrays can mitigate this effect, although close microphone matching is still necessary for high directivity. This paper explores the effect of microphone mismatch on the directivity of such arrays, and describes practical criteria for selecting matched microphones out of production batches to maximize a speech intelligibility weighted directivity index. [Work supported by Knowles Electronics, LLC.

  6. 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

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

    PubMed

    Groschup, Robin; Grosse, Christian U

    2015-06-25

    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.

  8. 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.

  9. 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.

  10. 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.

  11. 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.

  12. Blind source separation and localization using microphone arrays

    NASA Astrophysics Data System (ADS)

    Sun, Longji

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

  13. Acoustic source localization using a polyhedral microphone array and an improved generalized cross-correlation technique

    NASA Astrophysics Data System (ADS)

    Padois, Thomas; Sgard, Franck; Doutres, Olivier; Berry, Alain

    2017-01-01

    Millions of workers are exposed to excessive noise levels each day. Acoustic solutions have to be developed to protect workers from hearing loss. The first step of an acoustic diagnosis is the source localization which can be performed with a microphone array. Spherical microphone arrays can be used to detect the acoustic source positions in a workplace. In this study, a spherical microphone array, with polyhedral discretization, is proposed and compared with a spherical array with a slightly different geometry. The generalized cross-correlation technique is used to detect the source positions. Moreover, two criteria are introduced to improve the noise source map. The first is based on the geometric properties of the microphone array and the scan zone whereas the second is based on the energy of the spatial likelihood function. Numerical data are used to provide a systematic comparison of both geometries and criteria. Finally, experiments in a reverberant room reveal that the polyhedral microphone array associated with both criteria provides the best noise source map.

  14. 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.

  15. 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}.

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

    PubMed Central

    Jones, Douglas L.; Ratnam, Rama

    2009-01-01

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

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

    PubMed

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

    2015-09-01

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

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

    NASA Astrophysics Data System (ADS)

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

    2016-08-01

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

  19. Deconvolution 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.

  20. Deconvolution Methods and Systems for the Mapping of Acoustic Sources from Phased Microphone Arrays

    NASA Technical Reports Server (NTRS)

    Brooks, Thomas F. (Inventor); Humphreys, Jr., William M. (Inventor)

    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.

  1. 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.

  2. Effects of directional microphone and adaptive multichannel noise reduction algorithm on cochlear implant performance.

    PubMed

    Chung, King; Zeng, Fan-Gang; Acker, Kyle N

    2006-10-01

    Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed.

  3. Deduction of the acoustic impedance of the ground via a simulated three-dimensional microphone array.

    PubMed

    Alberts, W C Kirkpatrick; Sanchez, Kevin J

    2013-11-01

    While commonly used ground impedance deduction methods often utilize pairs of vertically separated microphones, deployed arrays rarely have this configuration, which increases the difficulty in automatically deducing local ground impedance from these arrays. The ability to deduce ground impedance using random sounds incident on a three-dimensional array would increase, for example, the accuracy of estimated elevation angles. The methods described by the American National Standards Institute Method for Determining the Acoustic Impedance of Ground Surfaces are extended to simulate deducing ground impedance by a three-dimensional array. Ground parameters indicative of grassland are successfully determined using a simulated three-dimensional array.

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

    NASA Technical Reports Server (NTRS)

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

    2015-01-01

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

  5. Objective performance analysis of spherical microphone arrays for speech enhancement in rooms.

    PubMed

    Peled, Yotam; Rafaely, Boaz

    2012-09-01

    Reverberation and noise have a significant effect on the intelligibility of speech in rooms. The detection of clear speech in highly reverberant and noisy enclosures is an extremely difficult task. Recently, spherical microphone arrays have been studied for processing of sound fields in three-dimensions, with applications ranging from acoustic analysis to speech enhancement. This paper presents the derivation of a model that facilitates the prediction of spherical array configurations that guarantee an acceptable level of speech intelligibility in reverberant and noisy environments. A spherical microphone array is employed to generate a spatial filter that maximizes speech intelligibility according to an objective measure that combines the effects of both reverberation and noise. The spherical array beamformer is designed to enhance the speech signal while minimizing noise power and maintaining robustness over a wide frequency range. The paper includes simulation and experimental studies with a comparison to speech transmission index based analysis to provide initial validation of the model. Examples are presented in which the minimum number of microphones in a spherical array can be determined from environment conditions such as reverberation time, noise level, and distance of the array to the speech source.

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

    PubMed

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

    2013-05-01

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

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

    PubMed

    Wang, Yan; Chen, Kean

    2016-07-01

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

  8. 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.

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

    PubMed

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

    2006-06-01

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

  10. Development of a Microphone Phased Array Capability for the Langley 14- by 22-Foot Subsonic Tunnel

    NASA Technical Reports Server (NTRS)

    Humphreys, William M.; Brooks, Thomas F.; Bahr, Christopher J.; Spalt, Taylor B.; Bartram, Scott M.; Culliton, William G.; Becker, Lawrence E.

    2014-01-01

    A new aeroacoustic measurement capability has been developed for use in open-jet testing in the NASA Langley 14- by 22-Foot Subsonic Tunnel (14x22 tunnel). A suite of instruments has been developed to characterize noise source strengths, locations, and directivity for both semi-span and full-span test articles in the facility. The primary instrument of the suite is a fully traversable microphone phased array for identification of noise source locations and strengths on models. The array can be mounted in the ceiling or on either side of the facility test section to accommodate various test article configurations. Complementing the phased array is an ensemble of streamwise traversing microphones that can be placed around the test section at defined locations to conduct noise source directivity studies along both flyover and sideline axes. A customized data acquisition system has been developed for the instrumentation suite that allows for command and control of all aspects of the array and microphone hardware, and is coupled with a comprehensive data reduction system to generate information in near real time. This information includes such items as time histories and spectral data for individual microphones and groups of microphones, contour presentations of noise source locations and strengths, and hemispherical directivity data. The data acquisition system integrates with the 14x22 tunnel data system to allow real time capture of facility parameters during acquisition of microphone data. The design of the phased array system has been vetted via a theoretical performance analysis based on conventional monopole beamforming and DAMAS deconvolution. The performance analysis provides the ability to compute figures of merit for the array as well as characterize factors such as beamwidths, sidelobe levels, and source discrimination for the types of noise sources anticipated in the 14x22 tunnel. The full paper will summarize in detail the design of the instrumentation

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

    PubMed

    van Hoesel, R J; Clark, G M

    1995-04-01

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

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

    PubMed

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

    2010-03-01

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

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

    NASA Astrophysics Data System (ADS)

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

    2011-08-01

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

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

    NASA Technical Reports Server (NTRS)

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

    1999-01-01

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

  15. Wind turbine noise measurement using a compact microphone array with advanced deconvolution algorithms

    NASA Astrophysics Data System (ADS)

    Ramachandran, Rakesh C.; Raman, Ganesh; Dougherty, Robert P.

    2014-07-01

    This paper experimentally investigates the noise from a large wind turbine (GE 1.5 MW) with a compact microphone array (OptiNav 24) using advanced deconvolution based beamforming methods, such as DAMAS and CLEAN-SC beamforming algorithms, for data reduction. Our study focuses on the ability of a compact microphone array to successfully locate both mechanical and aerodynamic noise sources on the wind turbine. Several interesting results have emerged from this study: (i) A compact microphone array is sufficient to perform a detailed study on wind turbine noise if advanced deconvolution methods are applied. (ii) Noise sources on the blade and on the nacelle can clearly be separated. (iii) Noise of the blades is dominated by trailing edge noise which is frequency dependent and is distributed along the length of the blade with the dominant noise source closer to the tip of the blade. (iv) The LP and DAMAS algorithms represent the distributed trailing edge noise source better than CLEAN-SC and classical beamforming. (v) Additional tonal noise produced during yawing operation is believed to be radiating from the tower of the wind turbine that acts like a resonator. (vi) Ground reflection is not believed to have a significant effect on noise source location estimates in this study.

  16. 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

  17. 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

  18. 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

  19. 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.

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

    NASA Astrophysics Data System (ADS)

    Bahr, Christopher J.; Cattafesta, Louis N.

    2016-11-01

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

  1. 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.

  2. 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.

  3. Adaptation to New Microphones Using Artificial Neural Networks With Trainable Activation Functions.

    PubMed

    Siniscalchi, Sabato Marco; Salerno, Valerio Mario

    2016-04-14

    Model adaptation is a key technique that enables a modern automatic speech recognition (ASR) system to adjust its parameters, using a small amount of enrolment data, to the nuances in the speech spectrum due to microphone mismatch in the training and test data. In this brief, we investigate four different adaptation schemes for connectionist (also known as hybrid) ASR systems that learn microphone-specific hidden unit contributions, given some adaptation material. This solution is made possible adopting one of the following schemes: 1) the use of Hermite activation functions; 2) the introduction of bias and slope parameters in the sigmoid activation functions; 3) the injection of an amplitude parameter specific for each sigmoid unit; or 4) the combination of 2) and 3). Such a simple yet effective solution allows the adapted model to be stored in a small-sized storage space, a highly desirable property of adaptation algorithms for deep neural networks that are suitable for large-scale online deployment. Experimental results indicate that the investigated approaches reduce word error rates on the standard Spoke 6 task of the Wall Street Journal corpus compared with unadapted ASR systems. Moreover, the proposed adaptation schemes all perform better than simple multicondition training and comparable favorably against conventional linear regression-based approaches while using up to 15 orders of magnitude fewer parameters. The proposed adaptation strategies are also effective when a single adaptation sentence is available.

  4. Adaptive multibeam antenna array

    NASA Astrophysics Data System (ADS)

    Novikov, V. I.

    1984-01-01

    An adaptive multibeam antenna array is considered which will enhance the advantages of a plain one. By providing simultaneous reception of signals from different directions and their sequential processing. The optimization of the array control for maximum interference suppression in the radiation pattern is emphasized. The optimum control is sought with respect to the signal-to-interference power ratio as a genaralized criterion. Sampled useful signals and transmission coefficients are found to be complex-conjugate quantities, assuming compatible formation of beams, so that synphasal equiamplitude addition of signals from all array element is attainable by unique settings of the weight factors. Calculations are simplified by letting the useful signal power in the 1-th beam be approximately equal to the k-th weight factor, before optimizing the weight vector for maximum signal-to-interference ratio. A narrowband interference described by power P and vector V of signal distribution over the array is considered as an example, to demonstrate the algorithm of synthesis. The algorithm, using the Butler matrix, was executed experimentally on a computer for a linear equidistant antenna array of 32 elements with compatible formation of beams.

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

    PubMed

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

    2017-03-04

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

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

    PubMed Central

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

    2017-01-01

    Automated surveillance of remote locations in a wireless sensor network is dominated by the detection algorithm because actual intrusions in such locations are a rare event. Therefore, a detection method with low power consumption is crucial for persistent surveillance to ensure longevity of the sensor networks. A simple and effective two-stage algorithm composed of energy detector (ED) and delay detector (DD) with all its operations in time-domain using small-aperture microphone array (SAMA) is proposed. The algorithm analyzes the quite different velocities between wind noise and sound waves to improve the detection capability of ED in the surveillance area. Experiments in four different fields with three types of vehicles show that the algorithm is robust to wind noise and the probability of detection and false alarm are 96.67% and 2.857%, respectively. PMID:28273838

  7. 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.

  8. 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

  9. Airy pattern approximation of a phased microphone array response to a rotating point source.

    PubMed

    Debrouwere, Maarten; Angland, David

    2017-02-01

    Deconvolution of phased microphone array source maps is a commonly applied technique in order to improve the dynamic range and resolution of beamforming. Most deconvolution algorithms require a point spread function (PSF). In this work, it is shown that the conventional definition of the PSF, based on steering vectors, is changed when the source is rotating. The effect of rotation results in an increase in the resolution and aperture of the array. The concept of virtual array positions created by source rotation is used to derive an approximation of the PSF based on an Airy pattern. The Airy pattern approximation is suitable for use in deconvolution of rotating source maps as it is more accurate and computationally less expensive than the conventional PSF definition. The proposed Airy pattern approximation was tested with both CLEAN and DAMAS deconvolution algorithms. On the same hardware, it was significantly faster when compared to the conventional definition. The limitations of the Airy pattern approximation are shown in a synthesized broadband test case with a high dynamic range. However, in most practical beamforming applications, the proposed Airy pattern approximated PSF for deconvolution is a suitable option considering its accuracy and speed.

  10. Real-time dual-microphone noise classification for environment-adaptive pipelines of cochlear implants.

    PubMed

    Mirzahasanloo, Taher; Kehtarnavaz, Nasser

    2013-01-01

    This paper presents an improved noise classification in environment-adaptive speech processing pipelines of cochlear implants. This improvement is achieved by using a dual-microphone and by using a computationally efficient feature-level combination approach to achieve real-time operation. A new measure named Suppression Advantage is also defined in order to quantify the noise suppression improvement of an entire pipeline due to noise classification. The noise classification and suppression improvement results are presented for four commonly encountered noise environments.

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

    NASA Astrophysics Data System (ADS)

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

    2016-10-01

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

  12. 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

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

    PubMed

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

    2014-01-23

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

  14. 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.

  15. 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.

  16. 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.

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

    PubMed

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

    2003-10-01

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

  18. 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.

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

    NASA Astrophysics Data System (ADS)

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

    2010-12-01

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

  20. A hybrid deconvolution approach to separate static and moving single-tone acoustic sources by phased microphone array measurements

    NASA Astrophysics Data System (ADS)

    Mo, Pinxi; Jiang, Weikang

    2017-02-01

    Beamforming approaches are developed to locate and quantify either static or moving acoustic sources by phased microphone array measurements. They would meet difficulties in mapping combined sources consisting of both static and moving sources. In this work, a hybrid deconvolution approach is proposed to separate static and moving single-tone sources. The approach is derived based on the source independence assumption as in the Deconvolution Approach for the Mapping of Acoustic Sources (DAMAS). The static beamforming and the moving beamforming are integrated to construct a linear matrix equation. The source distributions for the static sources and moving sources are simultaneously obtained by solving the equation. Numerical simulations and experiments were implemented on the combined sources with one static source and one rotating source. From the results, the hybrid deconvolution approach shows its effectiveness in separating the two sources, even with large source strength differences.

  1. 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.

  2. Adaptive ground implemented phase array

    NASA Technical Reports Server (NTRS)

    Spearing, R. E.

    1973-01-01

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

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

    NASA Technical Reports Server (NTRS)

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

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

    NASA Technical Reports Server (NTRS)

    Brooks, Thomas F.; Humphreys, William M.

    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.

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

    NASA Technical Reports Server (NTRS)

    Miles, J. H.

    1975-01-01

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

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

    PubMed

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

    2012-03-01

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

  7. 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

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

    PubMed

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

    2015-06-05

    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.

  9. Combined microphone array and lock-in amplifier operations for outdoor photo-acoustic sensing

    NASA Astrophysics Data System (ADS)

    Islam, Mohammad; Lay, Joshua; Wang, Chen-Chia; Trivedi, Sudhir; Samuels, Alan; Khurgin, Jacob; Sen-Choa, Fow

    2005-05-01

    Mid-infrared (MIR) standoff photoacoustic (PA) sensing of explosive chemicals and nerve gas stimulants at calibrated concentration have been demonstrated in door. When they are operated out door, array beam forming technique has to be employed to reject ambient noise and enhance signal. Lock-in amplifier usually needs to be used to achieve weak signal detection in a noisy environment. If we can combine these two techniques we will be able to reject both spatial and temporal noise and achieve a great signal to noise ratio (SNR) performance. From the best of our knowledge no literature has described how to combine these two techniques. In this work we demonstrated combined array and lock in amplifier operation in outdoor environment. A simplified system includes a signal generator, a speaker source, a lock in amplifier, 4 spy-phones with 4 parabolic reflectors to collect the acoustic signal, a National-Instrument NI6259 data acquisition system with both A to D (ADC) and D to A converters (DAC), and a PC. To combine these two techniques, each of the array collected signals was digitized by the ADC. Their path delays were adjusted in the computer to synchronize the phase. By using a PC controlled ADC the processing time is very long (~1s). To synchronize them without using costly high-speed customer made hardware, we delayed the reference signal by send it through the same ADC- PCDAC path as the array signals. By doing so, a good lock-in operation with stable phase was obtained.

  10. The use of an infrasound microphone array to study wind noise spectra and correlation

    NASA Astrophysics Data System (ADS)

    Shields, F. Douglas; Talmadge, Carrick

    2003-04-01

    A three dimensional array of infrasound sensors of original design has been constructed and used to study wind generated pressure signals in the frequency range from 0.1 to 100 Hz. The ten sensors in each arm of the array are 2 feet apart. An ultrasonic anemometer ten feet off the ground was used to make simultaneous measurements of the three components of the wind velocity. Several sets of data have been taken in open fields with different ground cover. The data have been spectrally analyzed and, over a limited frequency range, the velocity and pressure variations found to obey the 5/3 and 7/3 power law that is expected for the inertial range. A study has also been made of the dependence of the correlation between the pressure signals and the sensor separation. The coherence of the pressure signals indicates that the convection velocity is nearly independent of frequency, and the correlation has an exponentially decaying sinusoidal dependence on the sensor separation. The array has also been used successfully to localize infrasound sources. [Work supported by the U.S. Army Armament Research Development and Engineering Center.

  11. A multichannel speech enhancement method for functional MRI systems using a distributed microphone array.

    PubMed

    Milani, Ali A; Kannan, Govind; Panahi, Issa M S; Briggs, Richard

    2009-01-01

    Multichannel speech enhancement has been shown to be an effective method to decrease speech distortion introduced during speech enhancement, especially in environments like MRI (magnetic resonance imaging) which have a distributed noise source. However, these methods suffer from high computational complexity which makes them almost impractical. The use of subband filtering has been suggested to reduce this complexity but the performance of the existing subband methods deteriorate as the number of subbands increases. In this paper we introduce a new multichannel speech enhancement algorithm based on subband adaptive filtering that works for higher number of subbands at a lower complexity. The real-world experiments demonstrate the performance of the new scheme in an MRI room.

  12. 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…

  13. Optical microphone

    DOEpatents

    Veligdan, James T.

    2000-01-11

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

  14. Photorefractive processing for large adaptive phased arrays

    NASA Astrophysics Data System (ADS)

    Weverka, Robert T.; Wagner, Kelvin; Sarto, Anthony

    1996-03-01

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

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

    PubMed Central

    Beck, Christoph; Garreau, Guillaume; Georgiou, Julius

    2016-01-01

    Sand-scorpions and many other arachnids perceive their environment by using their feet to sense ground waves. They are able to determine amplitudes the size of an atom and locate the acoustic stimuli with an accuracy of within 13° based on their neuronal anatomy. We present here a prototype sound source localization system, inspired from this impressive performance. The system presented utilizes custom-built hardware with eight MEMS microphones, one for each foot, to acquire the acoustic scene, and a spiking neural model to localize the sound source. The current implementation shows smaller localization error than those observed in nature. PMID:27833526

  16. On grating nulls in adaptive arrays

    NASA Astrophysics Data System (ADS)

    Ishide, A.; Compton, R. T., Jr.

    1980-07-01

    The effect of element patterns on grating nulls in adaptive arrays is considered. Two simple array models, a two-element and a three-element array with dipole element patterns, are used to study this question. The element patterns are assumed unequal (i.e., the beam maxima point in different directions). It is shown that element patterns greatly affect the occurrence of grating nulls in the array. Unequal element patterns cause extra grating nulls ('sign reversal grating nulls') to occur, in addition to conventional grating nulls. These sign reversal grating nulls can occur even with element spacing less than a half-wavelength. For a two-element array with dipole element patterns, it turns out that grating nulls cannot be avoided if the spacing is greater than a half-wavelength. However, with more than two elements, the situation is not so bleak. An example is given of a three-element array with dipole patterns and one-wavelength spacing in which all grating nulls are eliminated.

  17. On grating nulls in adaptive arrays

    NASA Astrophysics Data System (ADS)

    Ishide, A.; Compton, R. T., Jr.

    1980-03-01

    This report considers the effect of element patterns on grating nulls in adaptive arrays. Two simple array models, a two-element and a three-element array with dipole element patterns, are used to study this question. The element patterns are assumed unequal (i.e., the beam maxima point in different directions). It is shown that element patterns greatly affect the occurrence of grating nulls in the array. Unequal element patterns cause extra grating nulls (sign reversal grating nulls) to occur, in addition to conventional grating nulls. These sign reversal grating nulls can occur even with element spacing less than a half-wavelength. For a two-element array with dipole element patterns, it turns out that grating nulls cannot be avoided if the spacing is greater than a half wavelength. However, with more than two elements, the situation is not so bleak. An example is given of a three-element array with dipole patterns and one wavelength spacing in which all grating nulls are eliminated.

  18. 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

  19. 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.

  20. 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.

  1. Adaptive Algorithms for HF Antenna Arrays.

    DTIC Science & Technology

    1987-07-01

    SUBJECT TERMS (Contnue on reverse dfnoceaq and identiy by bkICk numnber) FIELD GROUP SUB-GROUP HP Adaptive Arrays HrF Comunications Systems 4 HP...Although their heavy computational load renders them impractical *1 for many applications, the advancements in cheap, fast digital hardware have...or digital form. For many applications, the LMS algorithm represents a good trade off between speed of convergence* and implementational The speed of

  2. Adapting to the new radiology landscape: challenges and solutions discussed at the 2014 AMCLC open-microphone sessions.

    PubMed

    Hawkins, C Matthew; Flug, Jonathan A; Metter, Darlene; Strax, Richard; Lozano, Kay Denise Spong; Herrington, William; Applegate, Kimberly E

    2015-02-01

    Every year, multiple open-microphone sessions are hosted at the ACR AMCLC. These sessions allow members of the College to offer opinions, experiences, and questions regarding challenges facing radiologists and the future of the profession. At the 2014 AMCLC, 3 such sessions focused, respectively, on radiology's workforce, the obstacles slowing the shift from volume to value, and alternative reimbursement models and the shifting physician employment landscape. These open-microphone sessions framed contemporary obstacles and emerging challenges that professional radiology societies, such as the ACR, should target with new initiatives and use of resources; in addition, the sessions revealed opportunities for members, councilors, and state chapters to respond with meaningful resolutions and policy proposals.

  3. 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

  4. 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.

  5. 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.

  6. 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.

  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. 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)

  9. Adaptive optics for the CHARA array

    NASA Astrophysics Data System (ADS)

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

    2012-07-01

    The CHARA Array is a six telescope optical/IR interferometer run by the Center for High Angular Resolution Astronomy of Georgia State University and is located at Mount Wilson Observatory just to the north of Los Angeles California. The CHARA Array has the largest operational baselines in the world and has been in regular use for scientific observations since 2004. In 2011 we received funding from the NSF to begin work on Adaptive Optics for our six telescopes. Phase I of this project, fully funded by the NSF grant, consists of designing and building wavefront sensors for each telescope that will also serve as tip/tilt detectors. Having tip/tilt at the telescopes, instead of in the laboratory, will add several magnitudes of sensitivity to this system. Phase I also includes a slow wavefront sensor in the laboratory to measure non-common path errors and small deformable mirrors in the laboratory to remove static and slowly changing aberrations. Phase II of the project will allow us to place high-speed deformable mirrors at the telescopes thereby enabling full closed loop operation. We are currently seeking funding for Phase II. This paper will describe the scientific rational and design of the system and give the current status of the project.

  10. 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.

  11. 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

  12. 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

  13. 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.

  14. 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.

  15. Multiple-User Adaptive-Array Communication System

    NASA Technical Reports Server (NTRS)

    Zohar, S.

    1985-01-01

    Weights for K-beam system computed K/6 times faster. In single-frequency adaptive-array communication system in whick K mobile users communicate with central station equipped with n-antenna array. Each K signal recoverable by taking specific weighted sum of n complex antenna voltages.

  16. 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.

  17. 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

  18. Packaging of MEMS microphones

    NASA Astrophysics Data System (ADS)

    Feiertag, Gregor; Winter, Matthias; Leidl, Anton

    2009-05-01

    To miniaturize MEMS microphones we have developed a microphone package using flip chip technology instead of chip and wire bonding. In this new packaging technology MEMS and ASIC are flip chip bonded on a ceramic substrate. The package is sealed by a laminated polymer foil and by a metal layer. The sound port is on the bottom side in the ceramic substrate. In this paper the packaging technology is explained in detail and results of electro-acoustic characterization and reliability testing are presented. We will also explain the way which has led us from the packaging of Surface Acoustic Wave (SAW) components to the packaging of MEMS microphones.

  19. 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.

  20. 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.

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

    PubMed

    Legg, Mathew; Bradley, Stuart

    2013-10-01

    We present a calibration technique for an acoustic imaging microphone array, combined with a digital camera. Computer vision and acoustic time of arrival data are used to obtain microphone coordinates in the camera reference frame. Our new method allows acoustic maps to be plotted onto the camera images without the need for additional camera alignment or calibration. Microphones and cameras may be placed in an ad-hoc arrangement and, after calibration, the coordinates of the microphones are known in the reference frame of a camera in the array. No prior knowledge of microphone positions, inter-microphone spacings, or air temperature is required. This technique is applied to a spherical microphone array and a mean difference of 3 mm was obtained between the coordinates obtained with this calibration technique and those measured using a precision mechanical method.

  2. 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.

  3. A new microphone system for near whispering.

    PubMed

    Choi, Sungjoon; Moon, Wonkyu; Lee, Jeong Hyun

    2003-08-01

    A new microphone system was developed to monitor the human voice near the microphone in a noisy environment. The system is equipped with two special functions in addition to the usual microphone functions: reduction of air-blow effects by the mouth and focused reception to a sound source. A wind filter was developed to reduce the air-blow effects from the mouth during speaking. This filter is a plate perforated by an array of small holes; the method used to design the filter is also presented. To achieve focused reception, four microphones were used in conjunction with a new signal-processing method. The proposed signal-processing method effectively increases the directivity in the desired direction. Additionally, it provides the system with focusing on the source since the source is located adjacent to the system. A prototype of the proposed system was fabricated and subjected to performance tests. The results showed that air-blow effects can be reduced by up to 20 dB and the directional gain is more than 4 dB. The proposed microphone system shows such good performance that it can be used in mobile phones for whispering communication.

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

    NASA Astrophysics Data System (ADS)

    Kriehn, G. R.; Wagner, K.

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

  5. 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.

  6. Evaluation of Two Dimensional HF Adaptive Antenna Arrays

    DTIC Science & Technology

    1974-10-01

    CONVENTIONAL WEIGHTS 2. J. ADAPTIVE WEIGHTS 1. 4 RECEIVER ERRORS 2.4. 1 Error Analysis 2.4.2 Simulation SIMULATION RESULTS 3. i PROPERTIES OF...conditions, to document the research which has been done on the properties ot lov-redundancy sparse arrays when used with adaptive beam- formers...optimum processor*! a d.-grading effect on the resolution propertier . near a discrete source is noted. As McOonough notes, the effec of

  7. A Novel Monopulse Technique for Adaptive Phased Array Radar.

    PubMed

    Zhang, Xinyu; Li, Yang; Yang, Xiaopeng; Zheng, Le; Long, Teng; Baker, Christopher J

    2017-01-08

    The monopulse angle measuring technique is widely adopted in radar systems due to its simplicity and speed in accurately acquiring a target's angle. However, in a spatial adaptive array, beam distortion, due to adaptive beamforming, can result in serious deterioration of monopulse performance. In this paper, a novel constrained monopulse angle measuring algorithm is proposed for spatial adaptive arrays. This algorithm maintains the ability to suppress the unwanted signals without suffering from beam distortion. Compared with conventional adaptive monopulse methods, the proposed algorithm adopts a new form of constraint in forming the difference beam with the merit that it is more robust in most practical situations. At the same time, it also exhibits the simplicity of one-dimension monopulse, helping to make this algorithm even more appealing to use in adaptive planar arrays. The theoretical mean and variance of the proposed monopulse estimator is derived for theoretical analysis. Mathematical simulations are formulated to demonstrate the effectiveness and advantages of the proposed algorithm. Both theoretical analysis and simulation results show that the proposed algorithm can outperform the conventional adaptive monopulse methods in the presence of severe interference near the mainlobe.

  8. A Novel Monopulse Technique for Adaptive Phased Array Radar

    PubMed Central

    Zhang, Xinyu; Li, Yang; Yang, Xiaopeng; Zheng, Le; Long, Teng; Baker, Christopher J.

    2017-01-01

    The monopulse angle measuring technique is widely adopted in radar systems due to its simplicity and speed in accurately acquiring a target’s angle. However, in a spatial adaptive array, beam distortion, due to adaptive beamforming, can result in serious deterioration of monopulse performance. In this paper, a novel constrained monopulse angle measuring algorithm is proposed for spatial adaptive arrays. This algorithm maintains the ability to suppress the unwanted signals without suffering from beam distortion. Compared with conventional adaptive monopulse methods, the proposed algorithm adopts a new form of constraint in forming the difference beam with the merit that it is more robust in most practical situations. At the same time, it also exhibits the simplicity of one-dimension monopulse, helping to make this algorithm even more appealing to use in adaptive planar arrays. The theoretical mean and variance of the proposed monopulse estimator is derived for theoretical analysis. Mathematical simulations are formulated to demonstrate the effectiveness and advantages of the proposed algorithm. Both theoretical analysis and simulation results show that the proposed algorithm can outperform the conventional adaptive monopulse methods in the presence of severe interference near the mainlobe. PMID:28075348

  9. Biomimetic micromechanical adaptive flow-sensor arrays

    NASA Astrophysics Data System (ADS)

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

    2007-05-01

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

  10. 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.

  11. Dynamic Adaptive Neural Network Arrays: A Neuromorphic Architecture

    SciTech Connect

    Disney, Adam; Reynolds, John

    2015-01-01

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

  12. 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.

  13. Adaptive feed array compensation system for reflector antenna surface distortion

    NASA Technical Reports Server (NTRS)

    Acosta, Roberto J.; Zaman, A.

    1989-01-01

    The feasibility of a closed loop adaptive feed array system for compensating reflector surface deformations has been investigated. The performance characteristics (gain, sidelobe level, pointing, etc.) of large communication antenna systems degrade as the reflector surface distorts mainly due to thermal effects from a varying solar flux. The compensating systems described in this report can be used to maintain the design performance characteristics independent of thermal effects on the reflector surface. The proposed compensating system employs the concept of conjugate field matching to adjust the feed array complex excitation coefficients.

  14. 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.

  15. A multiple access communication network based on adaptive arrays

    NASA Technical Reports Server (NTRS)

    Zohar, S.

    1981-01-01

    A single frequency communication system is considered consisting of K possibly moving users distributed in space simultaneously communicating with a central station equipped with a computationally adapted array of n = or K antennas. Such a configuration could result if K spacecraft were to be simultaneously tracked by a single DSN complex consisting of an n antennas array. 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 known position information of the K users. Currently known techniques require (for n = K) about (4/3)K to the 4th power computer operations (multiply and add) to perform such computations. A technique that accomplishes this same goal in 8 K to the 3rd power operations, yielding a reduction by a factor K/6, was developed.

  16. 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.

  17. 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.

  18. 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.

  19. 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

  20. 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.

  1. 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.

  2. 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.

  3. Vibration Isolation of a Microphone.

    DTIC Science & Technology

    1985-09-01

    a. Microphone Replica A microphone replica identical in shape to the test microphone was machined from aluminum. A cutout was made on one side to...tion Research Program. 19. KEY WORDS (Continue on ,ererbe side if necesseary and identify by block number) vibration isolator, microphone, Space...34 ,.72_,IM=IN T S: -...S’L.’fMISS: READ ’SPL FOR ME..URED VOLTAGE RE.SF.E. READ XMISS’ FOR MEASURED TRANSMISEIBILITY . ’, ....... O ,IHED WERE WITHIN

  4. 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

  5. Toward a neuromorphic microphone.

    PubMed

    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.

  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. Fiber Optic Microphone

    NASA Technical Reports Server (NTRS)

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

    1999-01-01

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

  8. Superconducting microphone for photoacoustic spectroscopy

    SciTech Connect

    Costa Ribeiro, P.; Labrunie, M.; Von der Weid, J.P.; Symko, O.G.

    1982-11-01

    A superconducting microphone has been developed for photoacoustic spectroscopy at low temperatures. The microphone consists of a thin mylar membrane coated with a film of lead whose motion is detected by SQUID magnetometer. For the simple set-up presented here, the limiting pressure sensitivity is 7.5 x 10/sup -14/ atmospheres/Hz/sup 1/2/.

  9. 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

  10. Hydrogel microphones for stealthy underwater listening

    NASA Astrophysics Data System (ADS)

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

    2016-08-01

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

  11. 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

    2016-08-24

    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.

  12. Adaptive antenna arrays for satellite communications: Design and testing

    NASA Technical Reports Server (NTRS)

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

    1985-01-01

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

  13. 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.

  14. 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

  15. 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).

  16. Adaptive lesion formation using dual mode ultrasound array system

    NASA Astrophysics Data System (ADS)

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

    2017-03-01

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

  17. Adaptive nulling at HF using a compact array of active parasitic antennas

    NASA Astrophysics Data System (ADS)

    Jones, T. E.; Zeger, A. E.

    1985-03-01

    This parasitic array is a promising method of building an adaptive array in a tiny aperture. This is particularly attractive for HF. Both theory and experimental results are presented. The theory is established with respect to two models of a compact array, an impedance model and a transmission line model. The relationship between the models are derived. The models are then used to explore the null forming and adaptive control, with an emphasis on active terminations. Zeger-Abrams designed an electronically variable active termination. Experimental results were obtained from an array of both passive and active terminations. Empirical results with active complex-valued terminations demonstrated the feasibility of simultaneously nulling multiple HF jammers with an adaptively controlled compact array having parasitic auxilliary elements and an unweighted main antenna. Finally, an algorithm to adaptively control the active terminations is derived.

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

    NASA Technical Reports Server (NTRS)

    Gupta, Inder J.

    1986-01-01

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

  19. Microphone Boom for Aircraft-Engine Monitoring

    NASA Technical Reports Server (NTRS)

    Cohn, R.; Economu, M.; Albrecht, W.

    1986-01-01

    Microphone for measuring aircraft engine noise mounted on lengthwise boom supported away from fuselage and engine. This configuration minimizes boundary-layer effects and pressure doubling that is present if microphone were mounted in aircraft fuselage.

  20. 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.

  1. Optical microphone with fiber Bragg grating and signal processing techniques

    NASA Astrophysics Data System (ADS)

    Tosi, Daniele; Olivero, Massimo; Perrone, Guido

    2008-06-01

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

  2. Wind Noise Measurements of Microphones Embedded in the Airfoil of an UAV

    DTIC Science & Technology

    2012-10-01

    Embedded in the Airfoil of an UAV October 2012 Wayne E. Prather and William G. Frazier National Center for Physical Acoustics...platforms has been increasing in recent years. Embedding microphones in the skin of an airfoil is one option for how acoustic sensors might be...employed on a small Unmanned Aerial Vehicle (UAV). The analysis of wind noise measurements made by an array of microphones embedded in the airfoil of a

  3. Micromachined microphones with diffraction-based optical displacement detection.

    SciTech Connect

    Lee, Wook; Hall, Neal A.; Jeelani, M. Kamran; Bicen, Baris; Okandan, Murat; Degertekin, F. Levent; Qureshi, Shakeel

    2005-07-01

    Micromachined microphones with diffraction-based optical displacement detection are introduced. The approach enables interferometric displacement detection sensitivity in a system that can be optoelectronically integrated with a multichip module into mm{sup 3} volumes without beamsplitters, focusing optics, or critical alignment problems. Prototype devices fabricated using Sandia National Laboratories silicon based SwIFT-Lite{trademark} process are presented and characterized in detail. Integrated electrostatic actuation capabilities of the microphone diaphragm are used to perform dynamic characterization in vacuum and air environments to study the acoustic impedances in an equivalent circuit model of the device. The characterization results are used to predict the thermal mechanical noise spectrum, which is in excellent agreement with measurements performed in an anechoic test chamber. An A weighted displacement noise of 2.4 x 10{sup -2} {angstrom} measured from individual prototype 2100 {micro}m x 2100 {micro}m diaphragms demonstrates the potential for achieving precision measurement quality microphone performance from elements 1 mm{sup 2} in size. The high sensitivity to size ratio coupled with the ability to fabricate elements with precisely matched properties on the same silicon chip may make the approach ideal for realizing high fidelity miniature microphone arrays (sub-cm{sup 2} aperture) employing recently developed signal processing algorithms for sound source separation and localization in the audio frequency range.

  4. 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.

  5. Experimental Results for a Photonic Time Reversal Processor for Adaptive Control of an Ultra Wideband Phased Array Antenna

    DTIC Science & Technology

    2008-03-01

    Radar , Boston: Artech House, 1994. 2. H. Zmuda, “ Optical Beamforming for Phased Array Antennas,” Chapter 19, R...Beamforming, Phased Array Antennas, Time Reversal, Ultra Wideband Radar 1 INTRODUCTION 1.1 Photonic Processing for Microwave Phased Array ...Architecture for Broadband Adaptive Nulling with Linear and Conformal Phased Array Antennas”, Fiber and Integrated Optics , vol. 19, no. 2, March 2000, pp.

  6. 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.

  7. 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.

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

    DTIC Science & Technology

    2008-09-01

    correlation coefficient (CC), or some comparable detection statistic, exceeds a given threshold. Since these methods exploit characteristic details of the...multiple channels since stacking can be performed on the correlation coefficient traces with a significant array-gain. A detected event that is co-located...with the master event will record the same time-difference at every site in an arbitrarily spaced network which means that the correlation coefficient traces

  9. Optimal irregular microphone distributions with enhanced beamforming performance in immersive environments.

    PubMed

    Yu, Jingjing; Donohue, Kevin D

    2013-09-01

    Complex relationships between array gain patterns and microphone distributions limit the application of optimization algorithms on irregular arrays. This paper proposes a Genetic Algorithm (GA) for microphone array optimization in immersive (near-field) environments. Geometric descriptors for irregular arrays are proposed for use as objective functions to reduce optimization time by circumventing the need for direct array gain computations. In addition, probabilistic descriptions of acoustic scenes are introduced for incorporating prior knowledge of the source distribution. To verify the effectiveness of the proposed optimization, signal-to-noise ratios are compared for GA-optimized arrays, regular arrays, and arrays optimized through direct exhaustive simulations. Results show enhancements for GA-optimized arrays over arbitrary randomly generated arrays and regular arrays, especially at low microphone densities where placement becomes critical. Design parameters for the GA are identified for improving optimization robustness for different applications. The rapid convergence and acceptable processing times observed during the experiments establish the feasibility of this approach for optimizing array geometries in immersive environments where rapid deployment is required with limited knowledge of the acoustic scene, such as in mobile platforms and audio surveillance applications.

  10. 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.

  11. Analysis of nonstandard noise dosimeter microphone positions.

    PubMed

    Byrne, David C; Reeves, Efrem R

    2008-03-01

    This study was conducted as part of a project involving the evaluation of a new type of noise exposure monitoring paradigm. Laboratory tests were conducted to assess how "nonstandard" dosimeter microphones and microphone positions measured noise levels under different acoustical conditions (i.e., diffuse field and direct field). The data presented in this article reflect measurement differences due to microphone position and mounting/supporting structure only and are not an evaluation of any particular complete dosimeter system. To varying degrees, the results obtained with the dosimeter microphones used in this study differed from the reference results obtained in the unperturbed (subject absent) sound field with a precision (suitable for use in an ANSI Type 1 sound level meter) (1)/(2)-inch (12.7 mm) measurement microphone. Effects of dosimeter microphone placement in a diffuse field were found to be minor for most of the test microphones/locations, while direct field microphone placement effects were found to be quite large depending on the microphone position and supporting structure, sound source location, and noise spectrum.

  12. 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.

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

    PubMed

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

    2014-01-01

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

  14. 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.

  15. Surface estimation methods with phased-arrays for adaptive ultrasonic imaging in complex components

    NASA Astrophysics Data System (ADS)

    Robert, S.; Calmon, P.; Calvo, M.; Le Jeune, L.; Iakovleva, E.

    2015-03-01

    Immersion ultrasonic testing of structures with complex geometries may be significantly improved by using phased-arrays and specific adaptive algorithms that allow to image flaws under a complex and unknown interface. In this context, this paper presents a comparative study of different Surface Estimation Methods (SEM) available in the CIVA software and used for adaptive imaging. These methods are based either on time-of-flight measurements or on image processing. We also introduce a generalized adaptive method where flaws may be fully imaged with half-skip modes. In this method, both the surface and the back-wall of a complex structure are estimated before imaging flaws.

  16. 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.

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

    NASA Astrophysics Data System (ADS)

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

    2016-10-01

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

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

    PubMed

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

    2016-10-10

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

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

    PubMed Central

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

    2016-01-01

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

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

    PubMed

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

    1991-07-01

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

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

    NASA Astrophysics Data System (ADS)

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

    2010-07-01

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

  2. Field calibration of two types of microphones in hyperbaric air

    NASA Astrophysics Data System (ADS)

    Smith, Paul F.; Carpenter, Susan; Green, John

    1990-01-01

    The response of two microphones, one a condenser microphone and the other a diaphragm-activated piezoelectric ceramic microphone, were measured in compressed air at pressures as great as 810 kilopascals (8 atmospheres). The response of each microphone was compared to that of a hydrophone operated in air as a microphone. The results show that the two types of microphone respond similarly to high ambient pressure. Both types are less sensitive to sound pressure in compressed air than in air at normal pressures, and the frequency responses of both microphones are altered. The results are useful in the analyses of ambient noise measurements done during experiments in compressed air.

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

    NASA Astrophysics Data System (ADS)

    Xu, Miao; Ren, Hongwen

    2016-09-01

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

  4. Adaptive arrays - A new approach to the steady-state analysis

    NASA Technical Reports Server (NTRS)

    Zohar, S.

    1982-01-01

    A closed-form expression for the steady-state output signal-to-noise ratio (SNR) of an n-element adaptive array excited by one desired narrow-band signal and K - 1 narrow-band jammers is obtained. This is facilitated by representing each excitation by a complex n-dimensional vector - the excitation vector. It is shown that the important system parameters are functions of scalar products of pairs of these excitation vectors. In particular, the normalized output SNR of the array is shown to be the ratio of determinants whose elements involve these scaler products. Such determinants are also shown to be involved in the expressions for the optimal array weights.

  5. 77 FR 64446 - Wireless Microphones Proceeding

    Federal Register 2010, 2011, 2012, 2013, 2014

    2012-10-22

    ... Commission noted, most other radio communications services have shifted from analog to digital technology to... From the Federal Register Online via the Government Publishing Office FEDERAL COMMUNICATIONS COMMISSION 47 CFR Parts 15, 74, and 90 Wireless Microphones Proceeding AGENCY: Federal...

  6. DNA motifs determining the efficiency of adaptation into the Escherichia coli CRISPR array.

    PubMed

    Yosef, Ido; Shitrit, Dror; Goren, Moran G; Burstein, David; Pupko, Tal; Qimron, Udi

    2013-08-27

    Clustered regularly interspaced short palindromic repeats (CRISPR) and their associated proteins constitute a recently identified prokaryotic defense system against invading nucleic acids. DNA segments, termed protospacers, are integrated into the CRISPR array in a process called adaptation. Here, we establish a PCR-based assay that enables evaluating the adaptation efficiency of specific spacers into the type I-E Escherichia coli CRISPR array. Using this assay, we provide direct evidence that the protospacer adjacent motif along with the first base of the protospacer (5'-AAG) partially affect the efficiency of spacer acquisition. Remarkably, we identified a unique dinucleotide, 5'-AA, positioned at the 3' end of the spacer, that enhances efficiency of the spacer's acquisition. Insertion of this dinucleotide increased acquisition efficiency of two different spacers. DNA sequencing of newly adapted CRISPR arrays revealed that the position of the newly identified motif with respect to the 5'-AAG is important for affecting acquisition efficiency. Analysis of approximately 1 million spacers showed that this motif is overrepresented in frequently acquired spacers compared with those acquired rarely. Our results represent an example of a short nonprotospacer adjacent motif sequence that affects acquisition efficiency and suggest that other as yet unknown motifs affect acquisition efficiency in other CRISPR systems as well.

  7. 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.

  8. Ultrasound Nondestructive Evaluation (NDE) Imaging with Transducer Arrays and Adaptive Processing

    PubMed Central

    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

  9. 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.

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

    NASA Technical Reports Server (NTRS)

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

    1977-01-01

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

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

    PubMed

    Divett, T; Vennell, R; Stevens, C

    2013-02-28

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

  12. 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.

  13. 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.

  14. 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).

  15. Wavefront sensing and adaptive control in phased array of fiber collimators

    NASA Astrophysics Data System (ADS)

    Lachinova, Svetlana L.; Vorontsov, Mikhail A.

    2011-03-01

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

  16. 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.

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

    PubMed

    Nelson, J A; Hawkins, D B

    1994-03-01

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

  18. Computationally Efficient Blind Code Synchronization for Asynchronous DS-CDMA Systems with Adaptive Antenna Arrays

    NASA Astrophysics Data System (ADS)

    Hu, Chia-Chang

    2005-12-01

    A novel space-time adaptive near-far robust code-synchronization array detector for asynchronous DS-CDMA systems is developed in this paper. There are the same basic requirements that are needed by the conventional matched filter of an asynchronous DS-CDMA system. For the real-time applicability, a computationally efficient architecture of the proposed detector is developed that is based on the concept of the multistage Wiener filter (MWF) of Goldstein and Reed. This multistage technique results in a self-synchronizing detection criterion that requires no inversion or eigendecomposition of a covariance matrix. As a consequence, this detector achieves a complexity that is only a linear function of the size of antenna array ([InlineEquation not available: see fulltext.]), the rank of the MWF ([InlineEquation not available: see fulltext.]), the system processing gain ([InlineEquation not available: see fulltext.]), and the number of samples in a chip interval ([InlineEquation not available: see fulltext.]), that is,[InlineEquation not available: see fulltext.]. The complexity of the equivalent detector based on the minimum mean-squared error (MMSE) or the subspace-based eigenstructure analysis is a function of[InlineEquation not available: see fulltext.]. Moreover, this multistage scheme provides a rapid adaptive convergence under limited observation-data support. Simulations are conducted to evaluate the performance and convergence behavior of the proposed detector with the size of the[InlineEquation not available: see fulltext.]-element antenna array, the amount of the[InlineEquation not available: see fulltext.]-sample support, and the rank of the[InlineEquation not available: see fulltext.]-stage MWF. The performance advantage of the proposed detector over other DS-CDMA detectors is investigated as well.

  19. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2012 CFR

    2012-10-01

    ... 49 Transportation 5 2012-10-01 2012-10-01 false Microphone distance correction factors. 1 325.73... MOTOR CARRIER NOISE EMISSION STANDARDS Correction Factors § 325.73 Microphone distance correction... factors contained in § 325.75. If the distance between the microphone location point and the...

  20. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2014 CFR

    2014-10-01

    ... 49 Transportation 5 2014-10-01 2014-10-01 false Microphone distance correction factors. 1 325.73... MOTOR CARRIER NOISE EMISSION STANDARDS Correction Factors § 325.73 Microphone distance correction... factors contained in § 325.75. If the distance between the microphone location point and the...

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

    Code of Federal Regulations, 2013 CFR

    2013-10-01

    ... 49 Transportation 5 2013-10-01 2013-10-01 false Microphone distance correction factors. 1 325.73... MOTOR CARRIER NOISE EMISSION STANDARDS Correction Factors § 325.73 Microphone distance correction... factors contained in § 325.75. If the distance between the microphone location point and the...

  2. 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…

  3. Speech Intelligibility with Acoustic and Contact Microphones

    DTIC Science & Technology

    2005-04-01

    each category containing 16 word pairs that differ only in the initial consonant. The six consonant categories are voicing, nasality, sustention ...voiced) are paired with their bilabial stop counterparts. meat (nasal) vs. beat (voiced, bilabial stop) Sustention (Sust) No movement compared...and sustention categories. The current results clearly demonstrate that while the throat microphone enhances the signal-to-noise ratio, the

  4. Microphone Detects Boiler-Tube Leaks

    NASA Technical Reports Server (NTRS)

    Parthasarathy, S. P.

    1985-01-01

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

  5. 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.

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

    NASA Technical Reports Server (NTRS)

    Smith, J. M.

    1973-01-01

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

  7. A new type of microphone using flexoelectric barium strontium titnate

    NASA Astrophysics Data System (ADS)

    Kwon, Seol ryung; Huang, Wenbin; Zhang, Shujun; Yuan, Fuh-Gwo; Jiang, Xiaoning

    2014-03-01

    A flexoelectric bridge-structured microphone using bulk barium strontium titanate (Ba0.65Sr0.35TiO3 or BST) ceramic was investigated in this study. The flexoelectric microphone was installed in an anechoic box and exposed to the sound pressure emitted from a loud speaker. Charge sensitivity of the flexoelectric microphone was measured and calibrated using a reference microphone. The 1.5 mm×768 μm×50 μm micro-machined bridge-structured flexoelectric microphone has a sensitivity of 0.92 pC/Pa, while its resonance frequency was calculated to be 98.67 kHz. The analytical and experimental results show that the flexoelectric microphone has both high sensitivity and broad bandwidth, indicating that flexoelectric microphones are potential candidates for many applications.

  8. Adaptive beamforming with imperfect arrays: Pattern effects and their partial correction

    NASA Astrophysics Data System (ADS)

    Pettersson, Lars E.

    1992-12-01

    The effects of various types of imperfections on the radiation pattern of adaptive antenna arrays, using the Applebaum and Frost algorithms, are investigated. The imperfections considered are nonuniform noise figures in the channels, gain and phase shift error, IQ imbalance, DC offset, finite number of samples when forming the covariance matrix, quantization of the signals, and the weights and saturation of the signals. In particular the effects on the resulting sidelobe level, null depth, and gain are studied, and simulated results and analytical expressions for the expected pattern parameters are given. The effects of some methods to partly correct for the imperfections are also investigated. A method to obtain low sidelobe pattern when using the Frost algorithm is also discussed.

  9. 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.

  10. On adaptive phased-array tracking in the presence of main-lobe jammer suppression

    NASA Astrophysics Data System (ADS)

    Blanding, Wayne R.; Koch, Wolfgang; Nickel, Ulrich

    2006-05-01

    Advances in characterizing the angle measurement covariance for phased array monopulse radar systems that use adaptive beamforming to null out a jammer source allow for the use of improved sensor models in tracking algorithms. Using a detection probability likelihood function consisting of a Gaussian sum that incorporates negative contact measurement information, three tracking systems are compared when used to track a maneuvering target passing into and through standoff jammer interference. Each tracker differs in how closely it replicates sensor performance in terms of accuracy of measurement covariance and the use of negative information. Only the tracker that uses both the negative contact information and corrected angle measurement covariance is able to consistently reacquire the target when it exits the jammer interference.

  11. 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.

  12. Efficient architecture for a multichannel array subbanding system with adaptive processing

    NASA Astrophysics Data System (ADS)

    Rabinkin, Daniel V.; Nguyen, Huy T.

    2000-11-01

    An architecture is presented for front-end processing in a wideband array system which samples real signals. Such a system may be encountered in cellular telephony, radar, or low SNR digital communications receivers. The subbanding of data enables system data rate reduction, and creates a narrowband condition for adaptive processing within the subbands. The front-end performs passband filtering, equalization, subband decomposition and adaptive beamforming. The subbanding operation is efficiently implemented using a prototype lowpass finite impulse response (FIR) filter, decomposed into polyphase form, combined with a Fast Fourier Transform (FFT) block and a bank of modulating postmultipliers. If the system acquires real inputs, a single FFT may be used to operate on two channels, but a channel separation network is then required for recovery of individual channel data. A sequence of steps is described based on data transformation techniques that enables a maximally efficient implementation of the processing stages and eliminates the need for channel separation. Operation count is reduced, and several layers of processing are eliminated.

  13. Graphene electrostatic microphone and ultrasonic radio

    PubMed Central

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

    2015-01-01

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

  14. 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.

  15. 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

  16. 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.

    2004-05-01

    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.

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

    NASA Astrophysics Data System (ADS)

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

    2012-12-01

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

  18. Feasibility of an implanted microphone for cochlear implant listening.

    PubMed

    Gérard, Jean-Marc; Demanez, Laurent; Salmon, Caroline; Vanpoucke, Filiep; Walraevens, Joris; Plasmans, Anke; De Siati, Daniele; Lefèbvre, Philippe

    2017-03-01

    This study aimed at evaluating the feasibility of an implanted microphone for cochlear implants (CI) by comparison of hearing outcomes, sound quality and patient satisfaction of a subcutaneous microphone to a standard external microphone of a behind-the-ear sound processor. In this prospective feasibility study with a within-subject repeated measures design comparing the microphone modalities, ten experienced adult unilateral CI users received an implantable contralateral subcutaneous microphone attached to a percutaneous plug. The signal was pre-processed and fed into their CI sound processor. Subjects compared listening modes at home for a period of up to 4 months. At the end of the study the microphone was explanted. Aided audiometric thresholds, speech understanding in quiet, and sound quality questionnaires were assessed. On average thresholds (250, 500, 750, 1k, 2k, 3k, 4k and 6 kHz) with the subcutaneous microphone were 44.9 dB, compared to 36.4 dB for the external mode. Speech understanding on sentences in quiet was high, within approximately 90% of performance levels compared to hearing with an external microphone. Body sounds were audible but not annoying to almost all subjects. This feasibility study with a research device shows significantly better results than previous studies with implanted microphones. This is attributed to technology enhancements and careful fitting. Listening effort was somewhat increased with an implanted microphone. Under good sound conditions, speech performance is nearly similar to that of external microphones demonstrating that an implanted microphone is feasible in a range of normal listening conditions.

  19. Acoustic Source Elevation Angle Estimates Using Two Microphones

    DTIC Science & Technology

    2014-06-01

    Acoustic Source Elevation Angle Estimates Using Two Microphones by Kirsten A. Walker and W.C. Kirkpatrick Alberts, II ARL-TR-6976 June...TR-6976 June 2014 Acoustic Source Elevation Angle Estimates Using Two Microphones Kirsten A. Walker and W.C. Kirkpatrick Alberts, II...3. DATES COVERED (From - To) 4. TITLE AND SUBTITLE Acoustic Source Elevation Angle Estimates Using Two Microphones 5a. CONTRACT NUMBER 5b

  20. 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.

  1. Towards a sub 15-dBA optical micromachined microphone.

    PubMed

    Kim, Donghwan; Hall, Neal A

    2014-05-01

    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.

  2. 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

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

    PubMed

    Xu, Hailong; Cui, Xiaowei; Lu, Mingquan

    2016-03-11

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

  4. 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

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

    NASA Astrophysics Data System (ADS)

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

    2016-09-01

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

  6. Neuron unit arrays and Nature/Nurture adaptation for photonic multichip modules

    NASA Astrophysics Data System (ADS)

    Lue, Jaw-Chyng Lormen

    To implement a previously proposed 3-D hybrid electronic/photonic multichip module (PMCM) (mimicking a primate retina structure) capable of low-latency, high-throughput, parallel-processing computations, several critical hardware components are designed, fabricated, and tested. All components are made of MOSIS 1.5 mum n-well BiCMOS (bipolar complimentary metal oxide silicon) fabrication process. A 12-by-12 dual-input, dual-output silicon neuron unit array chip has been fabricated, and characterized. A desired sigmoid-shape optical output from a vertical surface emitting laser (VCSEL) driven by this chip (with a linear-optical-input) was obtained. A logarithmic amplifier circuitry has been fabricated, and characterized. The dynamic range of its sensed brightness is multiple decades wide. This bipolar-based circuit's high sensitivity at low input signal range can improve the overall optical responsivity of the PMCM if it is integrated. A floating gate design is verified to be a good candidate for the long-term analog weight storage. The floating gate controlled channel resistance can represent the lateral weighted interconnection in the PMCM. A preliminary active pixel sensor design is also characterized, and evaluated for weight storage. Physical constraints, trade-offs, and relationships among the components for optimizing the performance of the PMCM are discussed. Software-wise, an artificial neural learning algorithm (Nature/Nurture algorithm) is developed for modeling the PMCM. This algorithm describes the weight updating rules for both the vertical fixed (nature-like) and the lateral adaptive (nurture-like) weighted interconnections in the PMCM. The learning algorithm for the lateral weight adaptations is new, and derived based on the multi-layer error back-propagation (BP) supervised learning algorithm using gradient descent method. Results from a simple optical character recognition (OCR) simulation show: (1) A PMCM with only one hidden neuron layer is

  7. 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.

  8. Cochlear microphonics generated by microwave pulses.

    PubMed

    Chou, C; Galambos, R; Guy, A W; Lovely, R H

    1975-12-01

    Oscillations at 50 kHz have been recorded from the round window of guinea pigs during irradiation by 918-MHz pulsed microwaves. The oscillations promptly follow the stimulas, outlast it by about 200 musec and measure to 50 muV in amplitude. They precede the auditory nerve's response and disappear with death. They are interpreted to be a cochlear microphonic and hence to demonstrate that the microwave auditory effect, in the guinea pig at least, is accompanied by a mechanical disturbance of the hari cells of the cochlea.

  9. 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.

  10. 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...

  11. 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...

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

    NASA Astrophysics Data System (ADS)

    Liu, Ling

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

  13. 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.

  14. 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).

  15. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2010 CFR

    2010-10-01

    ... 49 Transportation 5 2010-10-01 2010-10-01 false Microphone distance correction factors. 1 325.73... MOTOR CARRIER NOISE EMISSION STANDARDS Correction Factors § 325.73 Microphone distance correction... account both the distance correction factors contained in § 325.73 and the ground surface...

  16. 49 CFR 325.73 - Microphone distance correction factors. 1

    Code of Federal Regulations, 2011 CFR

    2011-10-01

    ... 49 Transportation 5 2011-10-01 2011-10-01 false Microphone distance correction factors. 1 325.73... MOTOR CARRIER NOISE EMISSION STANDARDS Correction Factors § 325.73 Microphone distance correction... account both the distance correction factors contained in § 325.73 and the ground surface...

  17. Microphone multiplex system provides multiple outlets from single source

    NASA Technical Reports Server (NTRS)

    Lauver, R. E.

    1966-01-01

    Microphone multiplex system accepts an audio signal from a single source and provides any number of low impedance outputs at microphone level with complete isolation between output channels. Any input or output may be converted to high impedance by eliminating the associated transformer.

  18. Directional Microphone Hearing Aids in School Environments: Working toward Optimization

    ERIC Educational Resources Information Center

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

    2017-01-01

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

  19. 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.

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

    NASA Technical Reports Server (NTRS)

    Godara, Lal C.

    1990-01-01

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

  1. Performance analysis of adaptive fiber laser array propagating in atmosphere with correction of high order aberrations in sub-aperture

    NASA Astrophysics Data System (ADS)

    Li, Feng; Geng, Chao; Li, Xinyang; Qiu, Qi

    2016-10-01

    Recently developed adaptive fiber laser array technique provides a promising way incorporating aberrations correction with laser beams transmission. Existing researches are focused on sub-aperture low order aberrations (pistons and tips/tilts) compensation and got excellent correction results for weak and moderate turbulence in short range. While such results are not adequate for future laser applications which face longer range and stronger turbulence. So sub-aperture high aberrations compensation is necessary. Relationship between corrigible orders of sub-aperture aberrations and far-field metrics as power-in-the-bucket (PIB) and Strehl ratio is investigated with numeric simulation in this paper. Numerical investigation results shows that increment in array number won't result in effective improvement of the far-field metric if sub-aperture size is fixed. Low order aberrations compensation in sub-apertures gets its best performances only when turbulence strength is weak. Pistons compensation becomes invalid and higher order aberrations compensation is necessary when turbulence gets strong enough. Cost functions of the adaptive fiber laser array with high order aberrations correction in sub-apertures are defined and the optimum corrigible orders are discussed. Results shows that high order (less than first ten Zernike orders) compensation is acceptable where balance between increment of the far-field metric and the cost and complexity of the system could be reached.

  2. 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.

  3. 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.

  4. 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.

  5. Design of Robust Adaptive Array Processors for Non-Stationary Ocean Environments

    DTIC Science & Technology

    2009-02-20

    arrays deployed in the North Pacific . In 2007-2008 horizontal line array data from SWellEx-96, another ONR-sponsored experiment, was also analyzed...element shallow VLA deployed for SPICE04 was completed. Results of this analysis were presented at North Pacific Acoustic Laboratory (NPAL) Workshop...noise in the North Pacific , focusing primarily on whether an eigenanalysis of noise covariance matrices revealed anything about the underlying

  6. 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.

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

    NASA Technical Reports Server (NTRS)

    Kumar, R.

    1992-01-01

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

  8. Adaptation to New Microphones Using Tied-Mixture Normalization

    DTIC Science & Technology

    1994-01-01

    and WSJ1 corpora with 12 and 50 hours of recorded speech respectively. We trained two sets of phonetic models using the WSJ0 corpus and the com...speech. However the error rate reduces by a factor of 2 for some con- ditions by adding 50 hours of training high quality recorded speech...under Contract Nos. N00014-91-C-0115, and N00014-92-C-0035. References 1. A. Acero , and R.M. Stem, "Environmental Robustness in Au- tomatic Speech

  9. 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.

  10. Adapting the waterproof BMS-96 diode array for isodose determination of dynamic beams.

    PubMed

    Muren, L P; Hafslund, R; Valen, H; Schuster, G; Dahl, O

    2000-01-01

    We report the application of the Schuster BMS-96 waterproof linear diode array for isodose determination of dynamic beams. The array recorded beam profiles correctly, while depth dose distributions of dynamic beams with large variations in dose rate were registered erroneously. The deviations could be eliminated by appropriate software modifications. Until the software is revised, true isodoses can be obtained by rescaling each individual profile to the depth dose curve as measured with a single ionization chamber. After the corrections presented in the paper, isodoses interpolated from these corrected data sets agreed with ionization chamber measurements within 1-2%.

  11. 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.

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

    PubMed

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

    2014-05-30

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

  13. 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.

  14. 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

  15. 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

  16. 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.

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

    NASA Astrophysics Data System (ADS)

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

    2013-03-01

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

  18. Adaptive beamforming for array imaging of plate structures using lamb waves.

    PubMed

    Engholm, Marcus; Stepinski, Tadeusz

    2010-12-01

    Lamb waves are considered a promising tool for the monitoring of plate structures. Large areas of plate structures can be monitored using active arrays employing beamforming techniques. Dispersion and multiple propagating modes are issues that need to be addressed when working with Lamb waves. Previous work has mainly focused on standard delay-and-sum (DAS) beamforming while reducing the effects of multiple modes through frequency selectivity and transducer design. This paper presents a minimum variance distortionless response (MVDR) approach for Lamb waves using a uniform rectangular array (URA) and a single transmitter. Theoretically calculated dispersion curves are used to compensate for dispersion. The combination of the MVDR approach and the two-dimensional array improves the suppression of interfering Lamb modes. The proposed approach is evaluated on simulated and experimental data and compared with the standard DAS beamformer. It is shown that the MVDR algorithm performs better in terms of higher resolution and better side lobe and mode suppression capabilities. Known issues of the MVDR approach, such as signal cancellation in highly correlated environments and poor robustness, are addressed using methods that have proven effective for the purpose in other fields of active imaging.

  19. Remote probe microphone measurement to verify hearing aid performance.

    PubMed

    Ferrari, Deborah Viviane; Bernardez-Braga, Gabriela Rosito Alvarez

    2009-01-01

    We assessed the feasibility of obtaining probe microphone measurements of hearing aids at a distance. Face-to-face and remote probe microphone measurements were carried out in 60 hearing aid users (mean age 67 yrs) with uni- or bilateral hearing losses (105 ears tested). The participant and a facilitator were located in a room equipped with a probe microphone system interfaced to a PC. Desktop videoconferencing and application sharing was used to allow an audiologist in another room to instruct the facilitator and control the equipment via the LAN. There were significant correlations between face-to-face and remote real ear unaided response (REUR), aided response (REAR) and insertion gain (REIG) at seven discrete frequencies from 250 to 6000 Hz. Differences between face-to-face and remote responses were within the reported variability for probe microphone measurements themselves. The results show that remote probe microphone measurements are feasible and might improve the quality of public hearing aid services and professional training in Brazil.

  20. 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.

  1. Least Squares Adaptive and Bayes Optimal Array Processors for the Active Sonar Problem

    DTIC Science & Technology

    1989-10-01

    False Alarm: Q10 = f po(z)dz (3.15) RI Miss: Q 1 = f pl(z)dz (3.16) RO Detection: Q11 = f P(z)dz (3.17) RI Null Decision: QDo f po(z)dz (3.18) RO where...least squares lattice filter structures for adaptive processing of complex acoustic data, The Pennsylvania State University, 1984. (Masters Thesis ) 54

  2. 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

  3. 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

  4. 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.

  5. 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.

  6. 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.

  7. 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.

  8. 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.

  9. Miniaturized Nanocomposite Piezoelectric Microphones for UAS Applications

    DTIC Science & Technology

    2012-10-22

    Law, J. Goldberger, F. Kim, J. C. Johnson, Y. Zhang, R. J. Saykally, and P. . Yang, “Low-temperature wafer-scale production of ZnO nanowire arrays...microfabrication process flow that incorporates a technique for integrating ZnO nanorods into a polymer diaphragm was then developed. Sensors were... ZnO nanorods into a polymer diaphragm was then developed. Sensors were fabricated, released, and packaged on printed circuit boards in preparation for

  10. 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...

  11. 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...

  12. High-Temperature Microphone With Fiber-Optic Output

    NASA Technical Reports Server (NTRS)

    Hellbaum, Richard F.; Gunther, Michael F.; Clause, Richard O.; Murphy, Kent A.

    1994-01-01

    Acoustic-pressure transducer (microphone) with fiber-optic output designed to withstand hot, loud, structurally vibrating environment like that of jet engine. Features flat frequency response out to frequencies well beyond several-kilohertz range needed to test for acoustic-pressure loads on engine structures.

  13. A surface-micromachined capacitive microphone with improved sensitivity

    NASA Astrophysics Data System (ADS)

    Je, Chang Han; Lee, Jaewoo; Yang, Woo Seok; Kim, Jongdae; Cho, Young-Ho

    2013-05-01

    We present a surface-micromachined capacitive microphone with a membrane center hole and back-plate supports. The proposed membrane center hole reduces air damping at the center of the membrane and increases the sensitivity and frequency response. The back-plate supports allow for a deep back-chamber and prevent deformation of the back-plate. The proposed microelectromechanical-system (MEMS) microphone is fabricated using fully CMOS-compatible processes. The fabricated MEMS microphone has a membrane 500 µm in diameter and a center hole 30 µm in diameter. A deep back-chamber with a depth of 100 µm is formed by the back-plate supporting structures. During fabrication, the residual stress of the membrane is minimized using PECVD silicon nitride inserted in the metal membrane. The measured residual stress of the sensing membrane is 14.8 MPa. Acoustic measurements show that the sensitivity of the microphone is -49.1 dBV Pa-1 @1 kHz at a 12 V dc bias voltage, which is in good agreement with the calculated value.

  14. 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…

  15. Measurement of Gravitational Acceleration Using a Computer Microphone Port

    ERIC Educational Resources Information Center

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

    2012-01-01

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

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

    PubMed

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

    2012-06-01

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

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

    PubMed Central

    2012-01-01

    Objectives Many microphones have been developed to meet with the implantable requirement of totally implantable cochlear implant (TICI). However, a biocompatible one without destroying the intactness of the ossicular chain still remains under investigation. Such an implantable floating piezoelectric microphone (FPM) has been manufactured and shows an efficient electroacoustic performance in vitro test at our lab. We examined whether it pick up sensitively from the intact ossicular chain and postulated whether it be an optimal implantable one. Methods Animal controlled experiment: five adult cats (eight ears) were sacrificed as the model to test the electroacoustic performance of the FPM. Three groups were studied: (1) the experiment group (on malleus): the FPM glued onto the handle of the malleus of the intact ossicular chains; (2) negative control group (in vivo): the FPM only hung into the tympanic cavity; (3) positive control group (Hy-M30): a HiFi commercial microphone placed close to the site of the experiment ear. The testing speaker played pure tones orderly ranged from 0.25 to 8.0 kHz. The FPM inside the ear and the HiFi microphone simultaneously picked up acoustic vibration which recorded as .wav files to analyze. Results The FPM transformed acoustic vibration sensitively and flatly as did the in vitro test across the frequencies above 2.0 kHz, whereas inefficiently below 1.0 kHz for its overloading mass. Although the HiFi microphone presented more efficiently than the FPM did, there was no significant difference at 3.0 kHz and 8.0 kHz. Conclusions It is feasible to develop such an implantable FPM for future TICIs and TIHAs system on condition that the improvement of Micro Electromechanical System and piezoelectric ceramic material technology would be applied to reduce its weight and minimize its size. PMID:22353161

  18. 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

  19. Speech quality evaluation of subcutaneously implanted microphone using in vivo experiment.

    PubMed

    Woo, Seong Tak; Lee, Gihyoun; Jung, Eui Sung; Lim, Hyung-Gyu; Seong, Ki Woong; Lee, Jyung Hyun; Kim, Myoung Nam; Cho, Jin-Ho

    2014-01-01

    The microphone in a fully implantable hearing device (FIHD) is generally implanted under the skin covering the temporal bone. However, the implanted microphone can be affected by the skin, which causes both sound attenuation and distortion, particularly at high frequencies. As the degree of attenuation and distortion through the skin is severe, speech quality evaluation parameters are needed for the received signal when designing an implantable microphone. However, the performance of most implantable microphones is only assessed based on the sensitivity and frequency response. Thus, practical indicators based on human auditory characteristics are needed for an objective evaluation of the performance of implantable microphones. In this study, a subcutaneously implantable microphone was designed, and its frequency response investigated using an in vivo experiment. Plus, to evaluate the objective indicators, the speech quality of the signals measured by the implanted microphone was calculated using a MATLAB program, and the indicators compared before and after implantation.

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

    PubMed

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

    2014-06-01

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

  1. SVD-Based Optimal Filtering Technique for Noise Reduction in Hearing Aids Using Two Microphones

    NASA Astrophysics Data System (ADS)

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

    2002-12-01

    We introduce a new SVD-based (Singular value decomposition) strategy for noise reduction in hearing aids. This technique is evaluated for noise reduction in a behind-the-ear (BTE) hearing aid where two omnidirectional microphones are mounted in an endfire configuration. The behaviour of the SVD-based technique is compared to a two-stage adaptive beamformer for hearing aids developed by Vanden Berghe and Wouters (1998). The evaluation and comparison is done with a performance metric based on the speech intelligibility index (SII). The speech and noise signals are recorded in reverberant conditions with a signal-to-noise ratio of [InlineEquation not available: see fulltext.] and the spectrum of the noise signals is similar to the spectrum of the speech signal. The SVD-based technique works without initialization nor assumptions about a look direction, unlike the two-stage adaptive beamformer. Still, for different noise scenarios, the SVD-based technique performs as well as the two-stage adaptive beamformer, for a similar filter length and adaptation time for the filter coefficients. In a diffuse noise scenario, the SVD-based technique performs better than the two-stage adaptive beamformer and hence provides a more flexible and robust solution under speaker position variations and reverberant conditions.

  2. 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.

  3. 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

  4. 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.

  5. 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.

  6. Microphonics and Lorentz Transfer Function Measurements on the SNS Cryomodules

    SciTech Connect

    Jean Delayen; G. Davis

    2003-09-01

    As part of the acceptance tests, we have performed a number of measurements of microphonics levels and frequency spectra on the SNS cryomodules. These measurements are particularly important since those cryomodules may be used in the high-energy section of the RIA driver, a low-current CW accelerator. Measurements of the complete transfer functions between rf field modulation and cavity resonant frequency have also been performed.

  7. 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.

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

    PubMed

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

    2013-10-01

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

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

    PubMed

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

    2009-04-01

    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 x 2 mm(2) 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.

  10. 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

  11. The modal and flow velocity corrections of microphone turbulence screens

    NASA Astrophysics Data System (ADS)

    Davy, J. L.

    2007-09-01

    Microphone turbulence screens are used to suppress turbulent pressure fluctuations when measuring the acoustic pressure inside a duct with flow. They consist of a long tube with a slit covered with porous material, and thus are also called sampling tubes. Because they are not omnidirectional, it is necessary to calculate corrections when higher-order modes are propagating in the duct. In order to calculate these corrections it is necessary to know the directivity of the microphone turbulence screen, the propagation direction and energy of the duct modes and the flow velocity of the air in the duct. This paper derives a theoretical formula for the directivity of a microphone turbulence screen. It shows that this theoretical formula agrees better with experimental directivity data than the previously used empirical directivity formula. Because the empirical directivity formula is not a function of the flow velocity, previous research has separated the modal correction from the flow velocity correction. It is shown that it is not theoretically valid to separate the corrections, and that doing so can lead to large errors at high frequencies in the outlet duct. A new method of calculating the modal correction with flow is presented. This method uses a statistical room acoustics approach in contrast to the deterministic numerical approach of the older method. The new method requires much less computing. It is shown that the new method agrees fairly well with the old method for modal corrections without flow. The new method is compared with experimental measurements of the combined modal and flow velocity corrections. Although the trend is the same, the experimental results are higher than the theoretical results in the mid frequency range. The new method agrees reasonably well with the corrections given in ISO 5136:2003.

  12. MEMS capacitive accelerometer-based middle ear microphone.

    PubMed

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

    2012-12-01

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

  13. The Use of Miniature Electret Microphones in Small Optoacoustic Cells.

    DTIC Science & Technology

    1980-06-30

    OAA87 520 NEW MEXICO STATE UNIVA- 7 R LASR CRUCES F/6 9/5 THE USE OF MINIATURE ELECTRET MICROPHONES IN SMALL OPT’ACOUSTIC;-ETC(U) ,JUN AX S A...OPTOACOUSTIC CELLS it:".I FINAL REPORT STUART A. SCHLEUSENER JUNE 30, 1980 U.S. ARMY RESEARCH OFFICE Ic GRANT NO. DAAG29-78-G-0080 C NEW MEXICO STATE...ARMY RESEARCH OFFICE GRANT NO. DAAG29-78-G-0080 NEW MEXICO STATE UNIVERSITY APPROVED FOR PUBLIC RELEASE; DISTRIBUTION UNLIMITED U i) THE VIEW, OPINIONS

  14. 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.

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

    PubMed

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

    2014-09-01

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

  16. Isolated Word Recognition From In-Ear Microphone Data Using Hidden Markov Models (HMM)

    DTIC Science & Technology

    2006-03-01

    to more complicated and intrusive alternatives. The microphone earpiece contains a small passive 10 sensor, which is commercially available off-the...noisy surroundings and allow for a relatively simple speech recognition approach, an unconventional recording device, an earpiece microphone placed

  17. Practical considerations for a second-order directional hearing aid microphone system

    NASA Astrophysics Data System (ADS)

    Thompson, Stephen C.

    2003-04-01

    First-order directional microphone systems for hearing aids have been available for several years. Such a system uses two microphones and has a theoretical maximum free-field directivity index (DI) of 6.0 dB. A second-order microphone system using three microphones could provide a theoretical increase in free-field DI to 9.5 dB. These theoretical maximum DI values assume that the microphones have exactly matched sensitivities at all frequencies of interest. In practice, the individual microphones in the hearing aid always have slightly different sensitivities. For the small microphone separation necessary to fit in a hearing aid, these sensitivity matching errors degrade the directivity from the theoretical values, especially at low frequencies. This paper shows that, for first-order systems the directivity degradation due to sensitivity errors is relatively small. However, for second-order systems with practical microphone sensitivity matching specifications, the directivity degradation below 1 kHz is not tolerable. A hybrid order directive system is proposed that uses first-order processing at low frequencies and second-order directive processing at higher frequencies. This hybrid system is suggested as an alternative that could provide improved directivity index in the frequency regions that are important to speech intelligibility.

  18. 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.

  19. Fabrication of a dual-planar-coil dynamic microphone by MEMS techniques

    NASA Astrophysics Data System (ADS)

    Horng, Ray-Hua; Chen, Kuo-Feng; Tsai, Yao-Cheng; Suen, Cheng-You; Chang, Chao-Chih

    2010-06-01

    A dual-planar-coil miniature dynamic microphone, one of the electro-acoustic transducers working with the principle of the electromagnetic induction, has been realized by semiconductor micro-processing and micro-electro-mechanical system (MEMS) techniques. This MEMS microphone mainly consists of a 1 µm thick diaphragm sandwiched by two spiral coils and vibrating in the region with the highest magnetic flux density generated by a double magnetic system. In comparison with the traditional dynamic microphone, besides the miniaturized dimension, the MEMS microphone also provides 325 times the vibration velocity of the diaphragm faster than the traditional microphone. Measured by an audio analyzer, the frequency response of the MEMS microphone is only 4.5 dBV Pa-1 lower than that of the traditional microphone in the range between 50 Hz and 20 kHz. The responsivity of -54.8 dB Pa-1 (at 1 kHz) of the MEMS device is competitive to that of a traditional commercial dynamic microphone which typically ranges from -50 to -60 dBV Pa-1 (at 1 kHz).

  20. 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.

  1. 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

  2. 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.

  3. Microphonic versus end-tidal carbon dioxide nasal airflow detection in neonates with apnea.

    PubMed

    Toubas, P L; Duke, J C; Sekar, K C; McCaffree, M A

    1990-12-01

    Impedance pneumography in combination with expired CO2 monitoring are commonly used techniques for detecting central and obstructive apnea in infants. In this investigation an American Telephone and Telegraph StarSet-1 3000-ohm self-actuating microphone connected to the end of an infant cannula was used to monitor neonatal nasal airflow to detect breaths and apnea. The microphone was placed in a soundproof container to eliminate environmental sound artifacts. Analyses of 100 breaths from five patient samples during active and quiet sleep showed that there was no significant difference between microphone and expired CO2 recording of respiration. The techniques were 98% and 96% sensitive, respectively. Microphonic detection of nasal airflow identified 27 of the 32 episodes of upper airway obstruction (84.2%) registered by end-tidal CO2 recording. Inspiratory and expiratory events could also be well documented. Microphonic recording of nasal airflow is a reliable and inexpensive technique to detect apnea.

  4. 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.

  5. The robustness of hearing aid microphone preferences in everyday listening environments.

    PubMed

    Walden, Brian E; Surr, Rauna K; Cord, Mary T; Grant, Ken W; Summers, Van; Dittberner, Andrew B

    2007-05-01

    Automatic directionality algorithms currently implemented in hearing aids assume that hearing-impaired persons with similar hearing losses will prefer the same microphone processing mode in a specific everyday listening environment. The purpose of this study was to evaluate the robustness of microphone preferences in everyday listening. Two hearing-impaired persons made microphone preference judgments (omnidirectional preferred, directional preferred, no preference) in a variety of everyday listening situations. Simultaneously, these acoustic environments were recorded through the omnidirectional and directional microphone processing modes. The acoustic recordings were later presented in a laboratory setting for microphone preferences to the original two listeners and other listeners who differed in hearing ability and experience with directional microphone processing. The original two listeners were able to replicate their live microphone preferences in the laboratory with a high degree of accuracy. This suggests that the basis of the original live microphone preferences were largely represented in the acoustic recordings. Other hearing-impaired and normal-hearing participants who listened to the environmental recordings also accurately replicated the original live omnidirectional preferences; however, directional preferences were not as robust across the listeners. When the laboratory rating did not replicate the live directional microphone preference, listeners almost always expressed no preference for either microphone mode. Hence, a preference for omnidirectional processing was rarely expressed by any of the participants to recorded sites where directional processing had been preferred as a live judgment and vice versa. These results are interpreted to provide little basis for customizing automatic directionality algorithms for individual patients. The implications of these findings for hearing aid design are discussed.

  6. Cochlear microphonic potential recorded by transtympanic electrocochleography in normally-hearing and hearing-impaired ears

    PubMed Central

    Santarelli, R; Scimemi, P; Dal Monte, E; Arslan, E

    2006-01-01

    Summary The cochlear microphonic is a receptor potential believed to be generated primarily by outer hair cells. Its detection in surface recordings has been considered a distinctive sign of outer hair cell integrity in patients with auditory neuropathy. This report focuses on the results of an analysis performed on cochlear microphonic recorded by transtympanic electrocochleography in response to clicks in 502 subjects with normal hearing threshold or various degrees of hearing impairment, and in 20 patients with auditory neuropathy. Cochlear microphonics recorded in normally-hearing and hearing-impaired ears showed amplitudes decreasing by the elevation of compound action potential Cochlear microphonic responses were clearly detected in ears with profound hearing loss. After separating recordings according to the presence or absence of central nervous system pathology (CNS+ and CNS-, respectively), cochlear microphonic amplitude was significantly higher in CNS+ than in CNS- subjects with normally-hearing ears and at 70 dB nHL compound action potential threshold. Cochlear microphonic responses were detected in all auditory neuropathy patients, with similar amplitudes and thresholds to those calculated for normally-hearing CNS- subjects. Cochlear microphonic duration was significantly higher in auditory neuropathy and normally-hearing CNS+ patients compared to CNS- subjects. Our results show that: 1. cochlear microphonic detection is not a distinctive feature of auditory neuropathy; 2. CNS+ subjects showed enhancement in cochlear microphonic amplitude and duration, possibly due to efferent system dysfunction; 3. long-lasting, high frequency cochlear microphonics with amplitudes comparable to those obtained from CNS- ears were found in auditory neuropathy patients. This could result from a variable combination of afferent compartment lesion, efferent system dysfacilitation and loss of outer hair cells. PMID:16886850

  7. 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

  8. 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.

  9. 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.

  10. 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.

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

    PubMed

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

    2015-09-09

    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.

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

    PubMed

    Jeng, Yih-Nen; Yang, Tzung-Ming; Lee, Shang-Yin

    2011-01-01

    This study shows that a small electret condenser microphone connected to a notebook or a personal computer (PC) has a prominent response in the extremely low frequency region in a specific environment. It confines most acoustic waves within a tiny air cell as follows. The air cell is constructed by drilling a small hole in a digital versatile disk (DVD) plate. A small speaker and an electret condenser microphone are attached to the two sides of the hole. Thus, the acoustic energy emitted by the speaker and reaching the microphone is strong enough to actuate the diaphragm of the latter. The experiments showed that, once small air leakages are allowed on the margin of the speaker, the microphone captured the signal in the range of 0.5 to 20 Hz. Moreover, by removing the plastic cover of the microphone and attaching the microphone head to the vibration surface, the low frequency signal can be effectively captured too. Two examples are included to show the convenience of applying the microphone to pick up the low frequency vibration information of practical systems.

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

    PubMed Central

    Jeng, Yih-Nen; Yang, Tzung-Ming; Lee, Shang-Yin

    2011-01-01

    This study shows that a small electret condenser microphone connected to a notebook or a personal computer (PC) has a prominent response in the extremely low frequency region in a specific environment. It confines most acoustic waves within a tiny air cell as follows. The air cell is constructed by drilling a small hole in a digital versatile disk (DVD) plate. A small speaker and an electret condenser microphone are attached to the two sides of the hole. Thus, the acoustic energy emitted by the speaker and reaching the microphone is strong enough to actuate the diaphragm of the latter. The experiments showed that, once small air leakages are allowed on the margin of the speaker, the microphone captured the signal in the range of 0.5 to 20 Hz. Moreover, by removing the plastic cover of the microphone and attaching the microphone head to the vibration surface, the low frequency signal can be effectively captured too. Two examples are included to show the convenience of applying the microphone to pick up the low frequency vibration information of practical systems. PMID:22346594

  14. Some comparisons of binaural measurements made with different dummy heads and stereo microphone techniques

    NASA Astrophysics Data System (ADS)

    Mapp, Peter A.

    2004-10-01

    Binaural measurements have been made in a number of acoustic environments, and the results from different binaural heads and stereo microphones are compared. The object of the study was not only to establish what practical differences occurred between the various head formats, but also to see if a stereo microphone or pseudohead could be used for making auditorium binaural measurements. Five measurement platforms were employed. These included two binaural dummy heads, binaural in-ear probe microphones, an SAAS pseudohead stereo microphone and a M-S (midside) stereo microphone. In the latter case, three different midside ratios were employed and compared. The measurements were made in a reverberant recital hall (2.5-s RT) and small acoustically treated listening room (RT 0.2 s). Whereas relatively minor differences were found to occur between the heads, significant differences were found to occur with the stereo microphones. It is concluded that while useful information can be obtained from a stereo microphone, it is far from being the same as binaural.

  15. 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.

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

    PubMed

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

    2007-07-01

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

  17. Dummy head microphone and its application to the measurements of electro-acoustic equipments

    NASA Astrophysics Data System (ADS)

    1984-12-01

    An acoustic measurement system using a dummy head microphone was developed. The system consists of a dummy head microphone and an equipment for digital signal processing, and the system can be used to measure characteristics of hearing aids and headphones. The ear simulator, newly developed for practical use, is terminated in a simple resistance element of 320 cgs acoustic ohms. The dummy head microphone with the ear simulator shows the good agreement with the acoustical characteristics of the averaged human external ear. Several measured results of hearing aids and headphones using the measurement system are shown.

  18. Experimental Evaluation of an Adaptive Focusing Algorithm for a Microwave Planar Phased-Array Hyperthermia System at UCSF

    DTIC Science & Technology

    1993-05-17

    ESC"--=AD-A267 004 DT1C Tecnical Report EECTESAL 13 1993 I Experimental Evaluation of an Adaptive Focusing Algorithim for a Microwave Planar Phased... surgery , chemo-, and x-ray therapy [15]. One particular method used by itself or in conjunction with another is "tissue heating" or hyperthermia [15-19], a

  19. 78 FR 21977 - Certain Silicon Microphone Packages and Products Containing the Same; Commission Determination...

    Federal Register 2010, 2011, 2012, 2013, 2014

    2013-04-12

    ... From the Federal Register Online via the Government Publishing Office INTERNATIONAL TRADE COMMISSION Certain Silicon Microphone Packages and Products Containing the Same; Commission Determination Not To Review an Initial Determination Terminating Investigation Based on a Settlement...

  20. Evaluation of smartphone sound measurement applications (apps) using external microphones-A follow-up study.

    PubMed

    Kardous, Chucri A; Shaw, Peter B

    2016-10-01

    This follow-up study examines the accuracy of selected smartphone sound measurement applications (apps) using external calibrated microphones. The initial study examined 192 apps on the iOS and Android platforms and found four iOS apps with mean differences of ±2 dB of a reference sound level measurement system. This study evaluated the same four apps using external microphones. The results showed measurements within ±1 dB of the reference. This study suggests that using external calibrated microphones greatly improves the overall accuracy and precision of smartphone sound measurements, and removes much of the variability and limitations associated with the built-in smartphone microphones.

  1. Evaluation of smartphone sound measurement applications (apps) using external microphones – A follow-up Study

    PubMed Central

    Kardous, Chucri A.; Shaw, Peter B.

    2016-01-01

    This follow-up study examines the accuracy of selected smartphone sound measurement applications (apps) using external calibrated microphones. The initial study examined 192 apps on the iOS and Android platforms and found four iOS apps with mean differences of ±2 dB of a reference sound level measurement system. This study evaluated the same four apps using external microphones. The results showed measurements within ±1 dB of the reference. This study suggests that using external calibrated microphones greatly improves the overall accuracy and precision of smartphone sound measurements, and removes much of the variability and limitations associated with the built-in smartphone microphones. PMID:27794313

  2. 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.

  3. 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.

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

    DTIC Science & Technology

    2013-04-01

    Assessment of Operational Progress of NASA Langley Developed Windshield and Microphone for Infrasound by W.C. Kirkpatrick Alberts, II... Windshield and Microphone for Infrasound W.C. Kirkpatrick Alberts, II, Stephen M. Tenney, and John M. Noble Sensors and Electron Devices Directorate...NOT RETURN YOUR FORM TO THE ABOVE ADDRESS. 1. REPORT DATE (DD-MM-YYYY) April 2013 2. REPORT TYPE Final 3. DATES COVERED (From - To) Nov 2011–Jan

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

    PubMed

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

    2013-10-15

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

  6. 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

  7. 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.

  8. Design of a phased array for the generation of adaptive radiation force along a path surrounding a breast lesion for dynamic ultrasound elastography imaging.

    PubMed

    Ekeom, Didace; Hadj Henni, Anis; Cloutier, Guy

    2013-03-01

    This work demonstrates, with numerical simulations, the potential of an octagonal probe for the generation of radiation forces in a set of points following a path surrounding a breast lesion in the context of dynamic ultrasound elastography imaging. Because of the in-going wave adaptive focusing strategy, the proposed method is adapted to induce shear wave fronts to interact optimally with complex lesions. Transducer elements were based on 1-3 piezocomposite material. Three-dimensional simulations combining the finite element method and boundary element method with periodic boundary conditions in the elevation direction were used to predict acoustic wave radiation in a targeted region of interest. The coupling factor of the piezocomposite material and the radiated power of the transducer were optimized. The transducer's electrical impedance was targeted to 50 Ω. The probe was simulated by assembling the designed transducer elements to build an octagonal phased-array with 256 elements on each edge (for a total of 2048 elements). The central frequency is 4.54 MHz; simulated transducer elements are able to deliver enough power and can generate the radiation force with a relatively low level of voltage excitation. Using dynamic transmitter beamforming techniques, the radiation force along a path and resulting acoustic pattern in the breast were simulated assuming a linear isotropic medium. Magnitude and orientation of the acoustic intensity (radiation force) at any point of a generation path could be controlled for the case of an example representing a heterogeneous medium with an embedded soft mechanical inclusion.

  9. Development of a swallowing frequency meter using a laryngeal microphone.

    PubMed

    Tanaka, N; Nohara, K; Okuno, K; Kotani, Y; Okazaki, H; Matsumura, M; Sakai, T

    2012-06-01

    Disuse atrophy of swallowing-related organs is suspected when decreased swallowing frequency is seen in the elderly. However, swallowing frequency has not been examined in elderly people during daily life. We developed a swallowing frequency meter containing a laryngeal microphone that does not restrict the subject's ability to perform daily activities. In this study, the utility of the meter was assessed. Experiment 1: The ability of the meter to detect swallowing was examined. The subject was instructed to swallow saliva or foods at a voluntarily pace. During these procedures, swallowing events were simultaneously recorded by the meter, self-enumeration and videofluorography. As a result, all of the swallowing events identified by the meter coincided with the swallowing events identified by self-enumeration and videofluorography. Experiment 2: Swallowing sounds display various patterns both between and within individuals. Therefore, we examined the concordance rate between the number of swallowing events counted by the meter and that counted by self-enumeration in 15 subjects over a longer period than in experiment 1. The concordance rates calculated by two examiners between the meter and self-enumeration were 96·8 ± 4·5% and 98·9 ± 3·3% at rest and 95·2 ± 4·5% and 96·1 ± 4·1% during meals, respectively. Our findings indicate that this meter is useful for measuring the frequency of swallowing during daily situations.

  10. 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.

  11. 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.

  12. Robust Fusion of Multiple Microphone and Geophone Arrays in a Ground Sensor Network

    DTIC Science & Technology

    2006-10-01

    using modern pattern recognition methods that recognize patterns in, for instance, the emitted sound. Admittedly, there still are important obstacles ...d12, d13, and d23. Indeed, by knowing the distance d23 = t23/c, trigonometry gives the sought bearing angle θ. In theory, the t23 could be found as the

  13. Quantifying direct DQPSK receiver with integrated photodiode array by assessing an adapted common-mode rejection ratio

    NASA Astrophysics Data System (ADS)

    Wang, J.; Lauermann, M.; Zawadzki, C.; Brinker, W.; Zhang, Z.; de Felipe, D.; Keil, N.; Grote, N.; Schell, M.

    2011-12-01

    In this work, a direct DQPSK receiver was fabricated, which comprises a polymer waveguide based delay-line interferometer (DLI); a polymer based optical hybrid, and two monolithic pairs of > 25 GHz bandwidth photodiodes that are vertically coupled to the polymer planar lightwave circuit (PLC) via integrated 45° mirrors. The common mode rejection ratio (CMRR) is used to characterize the performance of coherent receivers, by indicating the electrical power balance between the balanced detectors. However, the standard CMRR can only be measured when the PDs can be illuminated separately. Also, the standard CMRR does not take into account the errors in the relative phases of the receiver outputs. We introduce an adapted CMRR to characterize the direct receiver, which takes into account the unequal responsivities of the PDs, the uneven split of the input power by the DLI and hybrid, the phase error and the extinction ratio of the DLI and hybrid.

  14. 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.

  15. 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.

  16. 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.

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

    PubMed

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

    2016-12-03

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

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

    PubMed

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

    2013-06-01

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

  19. 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.

  20. 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.

  1. 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.

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

    PubMed

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

    2013-04-20

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

  3. 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.

  4. 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.

  5. Audio source separation with multiple microphones on time-frequency representations

    NASA Astrophysics Data System (ADS)

    Sawada, Hiroshi

    2013-05-01

    This paper presents various source separation methods that utilize multiple microphones. We classify them into two classes. Methods that fall into the first class apply independent component analysis (ICA) or Gaussian mixture model (GMM) to frequency bin-wise observations, and then solve the permutation problem to reconstruct separated signals. The second type of method extends non-negative matrix factorization (NMF) to a multimicrophone situation, in which NMF bases are clustered according to their spatial properties. We have a unified understanding that all methods analyze a time-frequency representation with an additional microphone axis.

  6. Comparison of Theoretical Basics of Microphone and Piezoelectric Photothermal Spectroscopy of Semiconductors

    NASA Astrophysics Data System (ADS)

    Zakrzewski, J.; Maliński, M.; Chrobak, Ł.; Pawlak, M.

    2017-01-01

    Photothermal spectroscopy has found a wide range of applications as a method of monitoring thermal, optical and recombination parameters of semiconductors. We consider microphone detection, widely used in photoacoustic spectroscopy, and piezoelectric detection. Both methods require knowledge of the temperature distribution in the sample and in its surroundings, the support surface and gas. For the microphone signal, we simulated the temperature at one of the sample surfaces; for the piezoelectric signal, we simulated the spatial temperature distribution orthogonal to the sample surface. We modeled an idealized semiconducting sample and one with surface defects. We found that the amplitude and phase spectra vary between the methods, enabling determination of optical and thermal parameters.

  7. 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) Any person who manufactures, sells, leases, or offers for sale or lease, low power auxiliary...

  8. 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) Any person who manufactures, sells, leases, or offers for sale or lease, low power auxiliary...

  9. 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) Any person who manufactures, sells, leases, or offers for sale or lease, low power auxiliary...

  10. 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) Any person who manufactures, sells, leases, or offers for sale or lease, low power auxiliary...

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

    PubMed

    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 2012 were one-inch condenser microphones of three specific standard types: LS1Pn, LS1Po, and WS1P. Due to its long history of providing calibrations of such microphones to customers, NIST is in a unique position to analyze data concerning the long-term stability of these devices. This long history has enabled NIST to acquire and aggregate a substantial amount of repeat calibration data for a large number of microphones that belong to various other standards and calibration laboratories. In addition to determining microphone sensitivities at the time of calibration, it is important to have confidence that the microphones do not typically undergo significant drift as compared to the calibration uncertainty during the periods between calibrations. For each of the three microphone types, an average drift rate and approximate 95 % confidence interval were computed by two different statistical methods, and the results from the two methods were found to differ insignificantly in each case. These results apply to typical microphones of these types that are used in a suitable environment and handled with care. The average drift rate for Type LS1Pn microphones was -0.004 dB/year to 0.003 dB/year. The average drift rate for Type LS1Po microphones was -0.016 dB/year to 0.008 dB/year. The average drift rate for Type WS1P microphones was -0.004 dB/year to 0.018 dB/year. For each of these microphone types, the average drift rate is not significantly different from zero. This result is consistent with the performance expected of condenser microphones designed for use as transfer standards. In addition, the values that bound the confidence intervals are well within the limits specified for long-term stability in international standards. Even though these results show very good long-term stability historically

  12. Photoacoustic spectroscopy that uses a resonant characteristic of a microphone for in vitro measurements of glucose concentration.

    PubMed

    Joo Yong Sim; Chang-Geun Ahn; Eunju Jeong; Bong Kyu Kim

    2016-08-01

    Glucose measurements using photoacoustic spectroscopy have been highlighted to be a modality for non-invasive glucose monitoring. Previous photoacoustic spectroscopy for glucose measurements have used a resonant acoustic cell with a broadband capacitive microphone to increase sensitivity. However, a resonant characteristic of a microphone has not been investigated yet due to the working frequency range much lower than the resonance frequency of the microphone membrane. We, here, present a photoacoustic spectroscopy system that utilizes an ultrasound resonance of a microphone to increase sensitivity. We found that matching the resonance of a photoacoustic cell with the resonance of a microphone can increase signal-to-noise ratio and our system can distinguish the glucose concentration in liquid.

  13. 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.

  14. Recognition of In-Ear Microphone Speech Data Using Multi-Layer Neural Networks

    DTIC Science & Technology

    2006-03-01

    types, namely, the recurrent network [ Hopfield , 1982], and the backpropagation algorithm [Rumelhart, McClelland, 1986]. In particular, the discovery... Network Design, Campus Publishing Service, University of Colorado, Boulder, Colorado, 1996. Hopfield , J. J., “Neural networks and physical systems...MICROPHONE SPEECH DATA USING MULTI-LAYER NEURAL NETWORKS by Gokhan Bulbuller March 2006 Thesis Advisor: Monique P. Fargues Co

  15. 76 FR 4936 - Certain Silicon Microphone Packages and Products Containing the Same; Notice of Commission...

    Federal Register 2010, 2011, 2012, 2013, 2014

    2011-01-27

    ... From the Federal Register Online via the Government Publishing Office INTERNATIONAL TRADE COMMISSION Certain Silicon Microphone Packages and Products Containing the Same; Notice of Commission Determination To Review in Part an Initial Determination; On Review Taking No Position on Two Issues...

  16. Free-field reciprocity calibration of microphones in ultrasonic frequency range

    NASA Astrophysics Data System (ADS)

    Bouaoua, Nourreddine; Fedtke, Thomas; Mellert, Volker

    2005-09-01

    Exposure to airborne low-frequency ultrasound occurs in many industrial applications such as cleaning, welding plastics, measuring distances in buildings, and by using consumer devices such as camera rangefinders, automatic door openers, parametric ultrasound loudspeakers, etc. Exposure to ultrasound in air may be dangerous to the hearing and may have negative bio-effects on humans. Care should be taken in its use. In order to establish appropriate limits and to measure the output of ultrasound devices, there is a need to develop a sound-pressure standard in the frequency range from 20 kHz to about 160 kHz. A calibration project for quarter-inch microphones by the reciprocity method has been started in the PTB, and a new automated measurement setup for free-field reciprocity calibration of quarter-inch microphones has been established. A procedure for a free-field reciprocity calibration of quarter-inch microphones in the frequency range 20 to 160 kHz is described. First results of free-field calibration of quarter-inch microphones are presented, giving the sensitivities and the repeatability of results. The effects of reflections and cross talk on the accuracy of the measurements will be explained.

  17. 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

  18. 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.

  19. 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…

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

    PubMed Central

    Homentcovschi, Dorel; Miles, Ronald N.

    2011-01-01

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

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

    PubMed

    Homentcovschi, Dorel; Miles, Ronald N

    2011-12-01

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

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

    PubMed

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

    2015-08-18

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

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

    ERIC Educational Resources Information Center

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

    2014-01-01

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

  4. Determination of the Pressure Equivalent Noise Signal of Vector Sensors in a Hybrid Array

    DTIC Science & Technology

    2012-12-01

    Electromagnetic Interference HDPE High Density Polyethylene LOFAR Low Frequency Analysis and Recording MRA Maximum Response Axis NPS Naval... amplitude and phase of the array channels relative to the central microphone and then implemented the beamformer in the frequency domain. Under anechoic...rigid clamps to the stand helps to damp out high frequency vibration induced noise signals. The relatively large aspect presented by the array in

  5. 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.

  6. Data dependent random forest applied to screening for laryngeal disorders through analysis of sustained phonation: acoustic versus contact microphone.

    PubMed

    Verikas, A; Gelzinis, A; Vaiciukynas, E; Bacauskiene, M; Minelga, J; Hållander, M; Uloza, V; Padervinskis, E

    2015-02-01

    Comprehensive evaluation of results obtained using acoustic and contact microphones in screening for laryngeal disorders through analysis of sustained phonation is the main objective of this study. Aiming to obtain a versatile characterization of voice samples recorded using microphones of both types, 14 different sets of features are extracted and used to build an accurate classifier to distinguish between normal and pathological cases. We propose a new, data dependent random forests-based, way to combine information available from the different feature sets. An approach to exploring data and decisions made by a random forest is also presented. Experimental investigations using a mixed gender database of 273 subjects have shown that the perceptual linear predictive cepstral coefficients (PLPCC) was the best feature set for both microphones. However, the linear predictive coefficients (LPC) and linear predictive cosine transform coefficients (LPCTC) exhibited good performance in the acoustic microphone case only. Models designed using the acoustic microphone data significantly outperformed the ones built using data recorded by the contact microphone. The contact microphone did not bring any additional information useful for the classification. The proposed data dependent random forest significantly outperformed the traditional random forest.

  7. 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.

  8. A Detailed Gravitational Lens Model Based on Submillimeter Array and Keck Adaptive Optics Imaging of a Herschel-ATLAS Submillimeter Galaxy at z = 4.243

    NASA Astrophysics Data System (ADS)

    Bussmann, R. S.; Gurwell, M. A.; Fu, Hai; Smith, D. J. B.; Dye, S.; Auld, R.; Baes, M.; Baker, A. J.; Bonfield, D.; Cava, A.; Clements, D. L.; Cooray, A.; Coppin, K.; Dannerbauer, H.; Dariush, A.; De Zotti, G.; Dunne, L.; Eales, S.; Fritz, J.; Hopwood, R.; Ibar, E.; Ivison, R. J.; Jarvis, M. J.; Kim, S.; Leeuw, L. L.; Maddox, S.; Michałowski, M. J.; Negrello, M.; Pascale, E.; Pohlen, M.; Riechers, D. A.; Rigby, E.; Scott, Douglas; Temi, P.; Van der Werf, P. P.; Wardlow, J.; Wilner, D.; Verma, A.

    2012-09-01

    We present high-spatial resolution imaging obtained with the Submillimeter Array (SMA) at 880 μm and the Keck adaptive optics (AO) system at the K 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 ≈0farcs6) resolve the dust emission into multiple lensed images, while the Keck AO K S-band data (angular resolution ≈0farcs1) resolve the lens into a pair of galaxies separated by 0farcs3. We present an optical spectrum of the foreground lens obtained with the Gemini-South telescope that provides a lens redshift of z 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 μ = 4.1 ± 0.2 and has an intrinsic infrared (IR) luminosity of L IR = (2.1 ± 0.2) × 1013 L ⊙. We measure a half-light radius of the background source of r s = 4.4 ± 0.5 kpc which implies an IR luminosity surface density of ΣIR = (3.4 ± 0.9) × 1011 L ⊙ kpc-2, a value that is typical of z > 2 SMGs but significantly lower than IR luminous galaxies at z ~ 0. The two lens galaxies are compact (r lens ≈ 0.9 kpc) early-types with Einstein radii of θE1 = 0.57 ± 0.01 and θE2 = 0.40 ± 0.01 that imply masses of M lens1 = (7.4 ± 0.5) × 1010 M ⊙ and M lens2 = (3.7 ± 0.3) × 1010 M ⊙. 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 ~ 0.6. This work highlights the importance of high spatial resolution imaging in developing models of strongly lensed galaxies discovered by Herschel. Herschel is an ESA space observatory with science instruments provided by European-led Principal Investigator consortia and with important participation from NASA.

  9. 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.

  10. 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.

  11. 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.

  12. Some effects of the atmosphere and microphone placement on aircraft flyover noise measurements

    NASA Technical Reports Server (NTRS)

    Hosier, R. N.; Hilton, D. A.

    1975-01-01

    The effects of varying atmospheric conditions on certification-type noise measurements were studied. Tests were made under various atmospheric conditions at two test sites, Fresno, California, and Yuma, Arizona, using the same test aircraft, noise, and weather measuring equipment, and operating personnel. Measurements were made to determine the effects of the atmosphere and of microphone placement on aircraft flyover noise. The measurements were obtained for characterization of not only the acoustic signature of the test aircraft, but also specific atmospheric characteristics. Data are presented in the form of charts and tables which indicate that for a wide range of weather conditions, at both site locations, noise data were repeatable for similar aircraft operating conditions. The placement of microphones at ground level and at 1.2 m over both spaded sand and concrete illustrate the effects of ground reflections and surface impedance on the noise measurements.

  13. 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.

  14. 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.

  15. Estimation of Respiratory Rates Using the Built-in Microphone of a Smartphone or Headset.

    PubMed

    Nam, Yunyoung; Reyes, Bersain A; Chon, Ki H

    2016-11-01

    This paper proposes accurate respiratory rate estimation using nasal breath sound recordings from a smartphone. Specifically, the proposed method detects nasal airflow using a built-in smartphone microphone or a headset microphone placed underneath the nose. In addition, we also examined if tracheal breath sounds recorded by the built-in microphone of a smartphone placed on the paralaryngeal space can also be used to estimate different respiratory rates ranging from as low as 6 breaths/min to as high as 90 breaths/min. The true breathing rates were measured using inductance plethysmography bands placed around the chest and the abdomen of the subject. Inspiration and expiration were detected by averaging the power of nasal breath sounds. We investigated the suitability of using the smartphone-acquired breath sounds for respiratory rate estimation using two different spectral analyses of the sound envelope signals: The Welch periodogram and the autoregressive spectrum. To evaluate the performance of the proposed methods, data were collected from ten healthy subjects. For the breathing range studied (6-90 breaths/min), experimental results showed that our approach achieves an excellent performance accuracy for the nasal sound as the median errors were less than 1% for all breathing ranges. The tracheal sound, however, resulted in poor estimates of the respiratory rates using either spectral method. For both nasal and tracheal sounds, significant estimation outliers resulted for high breathing rates when subjects had nasal congestion, which often resulted in the doubling of the respiratory rates. Finally, we show that respiratory rates from the nasal sound can be accurately estimated even if a smartphone's microphone is as far as 30 cm away from the nose.

  16. 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.

  17. 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

  18. 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.

  19. 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.

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

    NASA Technical Reports Server (NTRS)

    1990-01-01

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

  1. 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.

  2. Effect of Power Line Interference on Microphone Calibration Measurements Made at or Near Harmonics of the Power Line Frequency.

    PubMed

    Wagner, Randall P; Nedzelnitsky, Victor

    2007-01-01

    The electrical measurements required during the primary calibrations of laboratory standard microphones by the reciprocity method can be influenced by power line interference. Because of this influence, the protocols of international inter-laboratory key comparisons of microphone calibrations usually have not included measurements at power line frequencies. Such interference has been observed in microphone output voltage measurements made with a microphone pressure reciprocity calibration system under development at NIST. This system was configured for a particular type of standard microphone in such a way that measurements of relatively small signal levels, which are more susceptible to the effect of power line interference, were required. This effect was investigated by acquiring microphone output voltage measurement data with the power line frequency adjusted to move the frequency of the interference relative to the center frequency of the measurement system passband. These data showed that the effect of power line interference for this system configuration can be more than one percent at test frequencies harmonically related to the power line frequency. These data also showed that adjusting the power line frequency to separate the interference and test frequencies by as little as 1.0 Hz can reduce the effect of the interference by at least an order of magnitude. Adjustment of the power line frequency could enable accurate measurements at test frequencies that otherwise might be avoided.

  3. 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.

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

    PubMed Central

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

    2012-01-01

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

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

    PubMed

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

    2011-01-01

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

  6. Speech Recognition for Environmental Control: Effect of Microphone Type, Dysarthria, and Severity on Recognition Results.

    PubMed

    Fager, Susan Koch; Burnfield, Judith M

    2015-01-01

    This study examines the use of commercially available automatic speech recognition (ASR) across microphone options as access to environmental control for individuals with and without dysarthria. A study of two groups of speakers (typical speech and dysarthria), was conducted to understand their performance using ASR and various microphones for environmental control. Specifically, dependent variables examined included attempts per command, recognition accuracy, frequency of error type, and perceived workload. A further sub-analysis of the group of participants with dysarthria examined the impact of severity. Results indicated a significantly larger number of attempts were required (P = 0.007), and significantly lower recognition accuracies were achieved by the dysarthric participants (P = 0.010). A sub-analysis examining severity demonstrated no significant differences between the typical speakers and participants with mild dysarthria. However, significant differences were evident (P = 0.007, P = 0.008) between mild and moderate-severe dysarthric participants. No significant differences existed across microphones. A higher frequency of threshold errors occurred for typical participants and no response errors for moderate-severe dysarthrics. There were no significant differences on the NASA Task Load Index.

  7. 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.

  8. 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.

  9. 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.

  10. 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.

  11. 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.

  12. 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.

  13. Realization of a High Sensitivity Microphone for a Hearing Aid Using a Graphene-PMMA Laminated Diaphragm.

    PubMed

    Woo, SeongTak; Han, Jae-Hyung; Lee, Jyung Hyun; Cho, Sunghun; Seong, Ki-Woong; Choi, Muhan; Cho, Jin-Ho

    2017-01-18

    Microphones for hearing aid systems are required to have high sensitivity, an appropriate bandwidth, and a wide dynamic range. In this paper, a high sensitivity microphone, 4 mm in diameter and using a multilayer graphene-PMMA laminated diaphragm that can be applied in hearing aids, is designed, optimized, and implemented. Typically, polyphenylene sulfide (PPS) has been used for the diaphragm of electret condenser microphones (ECM), and this method provides simple, low cost mass production. Generally, the sensitivity of the commercial 4 mm diameter ECM is about -30 to 35 dB (0 dB = 1 V/Pa). A microphone using a nanometer-thick graphene diaphragm has been found to have higher sensitivity than the conventional ECM. However, nanometer-thick multilayer graphene is vulnerable to large mechanical shocks or high sound pressures, and the practical production of nanometer-thick diaphragms also poses a challenge. However, if a multilayer graphene diaphragm of the same thickness as the conventional ECM is used, displacement during diaphragm vibration will be severely attenuated due to the high elastic modulus of graphene, and the microphone sensitivity will be greatly reduced. In this paper, we fabricate a multilayer graphene/poly(methyl methacrylate) (PMMA) laminated diaphragm with sensitivity higher than that of any other microphones currently available for hearing aids, with the appropriate bandwidth in the auditory range. The high sensitivity arises from the laminated structure of the thin graphene membrane with high elastic modulus and from the PMMA membrane with lower elastic modulus and higher dielectric constant. The optimal thickness ratio of the graphene-PMMA layered diaphragm was studied by both analytical and experimental methods, and then a fabricated diaphragm was assembled in a 4 mm diameter microphone package. The performance of the implemented microphone was evaluated, including the sensitivity and total harmonic distortion. It is demonstrated that the

  14. 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

  15. 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.

  16. 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.

  17. 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.

  18. Kokkos Array

    SciTech Connect

    Edwards Daniel Sunderland, Harold Carter

    2012-09-12

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

  19. Effect of Free Stream Turbulence on the Flow-Induced Background Noise of In-Flow Microphones

    NASA Technical Reports Server (NTRS)

    Allen, Christopher S.; Olson, Lawrence E. (Technical Monitor)

    1998-01-01

    When making noise measurements of sound sources in flow using microphones immersed in an air stream or wind tunnel, the factor limiting the dynamic range of the measurement is, in many cases, the noise caused by the flow over the microphone. To lower this self-noise, and to protect the microphone diaphragm, an aerodynamic microphone forebody is usually mounted on the tip of the omnidirectional microphone. The microphone probe is then pointed into the wind stream. Even with a microphone forebody, however, the self-noise persists, prompting further research in the area of microphone forebody design for flow-induced self-noise reduction. The magnitude and frequency characteristics of in-flow microphone probe self-noise is dependent upon the exterior shape of the probe and on the level of turbulence in the onset flow, among other things. Several recent studies present new designs for microphone forebodies, some showing the forbodies' self-noise characteristics when used in a given facility. However, these self-noise characteristics may change when the probes are used in different facilities. The present paper will present results of an experimental investigation to determine an empirical relationship between flow turbulence and self-noise levels for several microphone forebody shapes as a function of frequency. As a result, the microphone probe self-noise for these probes will be known as a function of freestream turbulence, and knowing the freestream turbulence spectra for a given facility, the probe self-noise can be predicted. Flow-induced microphone self-noise is believed to be related to the freestream. turbulence by three separate mechanisms. The first mechanism is produced by large scale, as compared to the probe size, turbulence which appears to the probe as a variation in the angle of attack of the freestream. flow. This apparent angle of attack variation causes the pressure along the probe surface to fluctuate, and at the location of the sensor orifice this

  20. 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.

  1. 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.

  2. Non-Contact Methods for Measuring Front Cavity Depths of Laboratory Standard Microphones Using a Depth-Measuring Microscope

    PubMed Central

    Nedzelnitsky, Victor; Wagner, Randall P.

    2008-01-01

    To achieve an acceptable degree of accuracy at high frequencies in some standardized methods for primary calibration of laboratory standard (LS) microphones, the front cavity depth lfc of each microphone must be known. This dimension must be measured using non-contact methods to prevent damage to the microphone diaphragm. The basic capabilities of an optical depth-measuring microscope were demonstrated by the agreement of its measurements within 0.7 μm of the known values of reference gage blocks. Using this microscope, two basic methods were applied to measure lfc. One (D) uses direct measurements at the microphone front surface annulus and conventional data reduction techniques. The other (GB) uses measurements at the surface of a gage block placed on the annulus, and plane-fitting data reduction techniques intended to reduce the effects of the slightly imperfect geometries of the microphones. The GB method was developed to provide a smoother surface of measurement than the relatively rough surface of the annulus, and to simulate the contact that occurs between the annulus and the smooth, plane surface of an acoustic coupler during microphone calibration. Using these methods, full data sets were obtained at 33 measurement positions (D), or 25 positions (GB). In addition, D and GB subsampling methods were applied by using subsamples of either the D or the GB full data sets. All these methods were applied to six LS microphones, three each of two different types. The GB subsampling methods are preferred for several reasons. The measurement results for lfc obtained by these methods agree well with those obtained by the GB method using the full data set. The expanded uncertainties of results from the GB subsampling methods are not very different from the expanded uncertainty of results from the GB method using the full data set, and are smaller than the expanded uncertainties of results from the D subsampling methods. Measurements of lfc using the GB subsampling

  3. 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.

  4. 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

    2016-08-17

    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.

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

    PubMed Central

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

    2016-01-01

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

  6. 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

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

    PubMed

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

    2016-12-26

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

  8. 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.

  9. 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.

  10. On the statistical errors in the estimate of acoustical energy density by using two microphones in a one dimensional field.

    PubMed

    Pascal, Jean-Claude; Thomas, Jean-Hugh; Li, Jing-Fang

    2008-10-01

    It was recently shown that the statistical errors of the measurement in the acoustic energy density by the two microphone method in waveguide have little variation when the losses of coherence between microphones increase. To explain these intervals of uncertainty, the variance of the measurement is expressed in this paper as a function of the various energy quantities of the acoustic fields--energy densities and sound intensities. The necessary conditions to reach the lower bound are clarified. The results obtained are illustrated by an example of a one-dimensional partially coherent field, which allows one to specify the relationship between the coherence functions of the pressure and particle velocity and those of the two microphone signals.

  11. 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.

  12. Characterization of condenser microphones under different environmental conditions for accurate speed of sound measurements with acoustic resonators

    SciTech Connect

    Guianvarc'h, Cecile; Pitre, Laurent; Bruneau, Michel

    2009-07-15

    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.

  13. 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.

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

    PubMed

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

    2016-12-17

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

  15. Selecting the gain for radio microphone (FM) systems: theoretical considerations and practical limitations.

    PubMed

    Rowson, V J; Bamford, J M

    1995-06-01

    This paper provides theoretical estimates of the signal-to-noise (S/N) ratio advantage that can be gained from a radio microphone (FM) system used with an environmental (EV) microphone, compared with the S/N ratio delivered by a conventional hearing aid alone. These estimates show that the S/N advantage gained is a function of the speaker/listener distance and the FM/EV gain difference, the greater the distance and the closer the FM gain is to the EV gain, the greater the S/N advantage. The implications for choice of FM and EV gain when fitting FM systems are discussed. The two goals of equating the FM and EV outputs and maximizing the S/N ratio are mutually incompatible. It is concluded that the FM gain should be set to be 10dB less than the EV gain unless the situation can be closely monitored, in which case the FM gain should be set as high as possible consistent with loudness tolerance and acceptable levels of distortion.

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

    PubMed Central

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

    2016-01-01

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

  17. 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

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

    NASA Astrophysics Data System (ADS)

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

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

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

    PubMed

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

    2008-06-01

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

  20. 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.

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

    PubMed

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

    2007-03-01

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

  2. Dual-microphone Sounds of Daily Life classification for telemonitoring in a noisy environment.

    PubMed

    Maunder, David; Ambikairajah, Eliathamby; Epps, Julien; Celler, Branko

    2008-01-01

    Telemonitoring of elderly people in their homes using video cameras is complicated by privacy concerns, and hence sound has emerged as a promising alternative that is more acceptable to users. We investigate methods to address the accuracy degradation of sound classification that arises in the presence of background noise typical of a practical telemonitoring situation. A dual microphone configuration is used to record a database of Sounds of Daily Life (SDL) in a kitchen. We introduce a new algorithm employing the eigenvalues of the cross-spectral matrix of the recorded signals to detect the endpoints of a SDL in the presence of background noise. Independent component analysis is also used to improve the signal to noise ratio of the SDL. Results on a 7-class noisy SDL classification problem show that the error rate the proposed SDL classification system can be improved by up to 97% relative to a single-microphone system without noise reduction techniques, when evaluated on a large SDL database with SNRs in the range 0-28 dB.

  3. 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.

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

    PubMed

    Lien, Yu-An S; Stepp, Cara E

    2014-05-01

    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.

  5. 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.

  6. Coherence analysis using canonical coordinate decomposition with applications to sparse processing and optimal array deployment

    NASA Astrophysics Data System (ADS)

    Azimi-Sadjadi, Mahmood R.; Pezeshki, Ali; Wade, Robert L.

    2004-09-01

    Sparse array processing methods are typically used to improve the spatial resolution of sensor arrays for the estimation of direction of arrival (DOA). The fundamental assumption behind these methods is that signals that are received by the sparse sensors (or a group of sensors) are coherent. However, coherence may vary significantly with the changes in environmental, terrain, and, operating conditions. In this paper canonical correlation analysis is used to study the variations in coherence between pairs of sub-arrays in a sparse array problem. The data set for this study is a subset of an acoustic signature data set, acquired from the US Army TACOM-ARDEC, Picatinny Arsenal, NJ. This data set is collected using three wagon-wheel type arrays with five microphones. The results show that in nominal operating conditions, i.e. no extreme wind noise or masking effects by trees, building, etc., the signals collected at different sensor arrays are indeed coherent even at distant node separation.

  7. Estimation of the far-field directivity of broadband aeroengine fan noise using an in-duct axial microphone array

    NASA Astrophysics Data System (ADS)

    Lowis, C. R.; Joseph, P. F.; Kempton, A. J.

    2010-09-01

    This paper presents a measurement technique for estimating the far-field directivity of the sound radiated from a duct using measurements of acoustic pressure made inside the duct. The technique is restricted to broadband, multi-mode sound fields whose directivity patterns are axi-symmetric, and whose modes are mutually uncorrelated. The technique uses a transfer function to relate the output from an in-duct axial beamformer to measurements of the far-field polar directivity. A transfer function for a hollow cylindrical duct with no flow is derived, and investigated in detail. Transfer functions for practical cases concerning aeroengine exhausts are also presented. The transfer function is shown to be insensitive to the mode-amplitude distribution inside the duct, and hence can be used to predict the directivity in practice where the noise source distribution is unknown. The technique is then validated using a no-flow facility, and is shown to be able to predict variations in the far-field directivity pattern and also estimate the far-field sound pressure levels to within 2 dB. It is suggested that the proposed technique will be especially useful for fan rig experiments, where direct measurement of directivity, for example by use of an anechoic chamber, is impossible.

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

    NASA Technical Reports Server (NTRS)

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

    1990-01-01

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

  9. High resolution beamforming for small aperture arrays

    NASA Astrophysics Data System (ADS)

    Clark, Chris; Null, Tom; Wagstaff, Ronald A.

    2003-04-01

    Achieving fine resolution bearing estimates for multiple sources using acoustic arrays with small apertures, in number of wavelengths, is a difficult challenge. It requires both large signal-to-noise ratio (SNR) gains and very narrow beam responses. High resolution beamforming for small aperture arrays is accomplished by exploiting acoustical fluctuations. Acoustical fluctuations in the atmosphere are caused by wind turbulence along the propagation path, air turbulence at the sensor, source/receiver motion, unsteady source level, and fine scale temperature variations. Similar environmental and source dependent phenomena cause fluctuations in other propagation media, e.g., undersea, optics, infrared. Amplitude fluctuations are exploited to deconvolve the beam response functions from the beamformed data of small arrays to achieve high spatial resolution, i.e., fine bearing resolution, and substantial SNR gain. Results are presented for a six microphone low-frequency array with an aperture of less than three wavelengths. [Work supported by U.S. Army Armament Research Development and Engineering Center.

  10. Vocabulary and Environment Adaptation in Vocabulary-Independent Speech Recognition

    DTIC Science & Technology

    1992-01-01

    normalization (ISDCN) proposed by Acero et al. [2] for microphone adaptation are incorporated into the our VI system to achieve environmental...reverberation from surface reflec- tions, etc. Acero at al. [1,2] proposed a series of environment normal- ization algorithms based on joint...support. References [1] Acero , A. Acoustical and Environmental Robustness in Auto- matic Speech Recognition. Department of Electrical Engineer- ing

  11. 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.

  12. 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.

  13. 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…

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

    PubMed

    Ziegelwanger, Harald; Majdak, Piotr; Kreuzer, Wolfgang

    2015-07-01

    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.

  15. ACR 2016 Open-Microphone Session: ACR Membership 3.0-Moving From Member Volume to Value.

    PubMed

    Stern, Eric; Metter, Darlene; Everett, Catherine; Flug, Jonathan; Friedberg, Eric; Kotsenas, Amy; Nathan, Jennifer; Herrington, William

    2017-04-02

    This report of the 2016 ACR Council Open Microphone session reviews the discussion around interests and concerns of council members and state chapter leaders as to the perceived and real value of their ACR membership, and how the ACR might further enhance membership value and meaningful engagement with members.

  16. 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

    ... 47 Telecommunication 1 2010-10-01 2010-10-01 false Disclosure requirements for wireless... 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. 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.

  18. A Novel Technique for Maximum Power Point Tracking of a Photovoltaic Based on Sensing of Array Current Using Adaptive Neuro-Fuzzy Inference System (ANFIS)

    NASA Astrophysics Data System (ADS)

    El-Zoghby, Helmy M.; Bendary, Ahmed F.

    2016-10-01

    Maximum Power Point Tracking (MPPT) is now widely used method in increasing the photovoltaic (PV) efficiency. The conventional MPPT methods have many problems concerning the accuracy, flexibility and efficiency. The MPP depends on the PV temperature and solar irradiation that randomly varied. In this paper an artificial intelligence based controller is presented through implementing of an Adaptive Neuro-Fuzzy Inference System (ANFIS) to obtain maximum power from PV. The ANFIS inputs are the temperature and cell current, and the output is optimal voltage at maximum power. During operation the trained ANFIS senses the PV current using suitable sensor and also senses the temperature to determine the optimal operating voltage that corresponds to the current at MPP. This voltage is used to control the boost converter duty cycle. The MATLAB simulation results shows the effectiveness of the ANFIS with sensing the PV current in obtaining the MPPT from the PV.

  19. 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

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

    PubMed Central

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

    2011-01-01

    Objectives Cochlear implant microphones differ in placement, frequency response, and other characteristics such as whether they are directional. Although normal hearing individuals are often used as controls in studies examining cochlear implant users’ binaural benefits, the considerable differences across cochlear implant microphones make such comparisons potentially misleading. The goal of this study was to examine binaural benefits for speech perception in noise for normal hearing individuals using stimuli processed by head-related transfer functions (HRTFs) based on the different cochlear implant microphones. Design HRTFs were created for different cochlear implant microphones and used to test participants on the Hearing in Noise Test. Experiment 1 tested cochlear implant users and normal hearing individuals with HRTF-processed stimuli and with sound field testing to determine whether the HRTFs adequately simulated sound field testing. Experiment 2 determined the measurement error and performance-intensity function for the Hearing in Noise Test with normal hearing individuals listening to stimuli processed with the various HRTFs. Experiment 3 compared normal hearing listeners’ performance across HRTFs to determine how the HRTFs affected performance. Experiment 4 evaluated binaural benefits for normal hearing listeners using the various HRTFs, including ones that were modified to investigate the contributions of interaural time and level cues. Results The results indicated that the HRTFs adequately simulated sound field testing for the Hearing in Noise Test. They also demonstrated that the test-retest reliability and performance-intensity function were consistent across HRTFs, and that the measurement error for the test was 1.3 dB, with a change in signal-to-noise ratio of 1 dB reflecting a 10% change in intelligibility. There were significant differences in performance when using the various HRTFs, with particularly good thresholds for the HRTF based on the

  1. A novel vibration mode testing method for cylindrical resonators based on microphones.

    PubMed

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

    2015-01-16

    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.

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

    PubMed

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

    2015-08-18

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

  3. 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.

  4. 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.

  5. 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.

  6. 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.

  7. SNP Arrays

    PubMed Central

    Louhelainen, Jari

    2016-01-01

    The papers published in this Special Issue “SNP arrays” (Single Nucleotide Polymorphism Arrays) focus on several perspectives associated with arrays of this type. The range of papers vary from a case report to reviews, thereby targeting wider audiences working in this field. The research focus of SNP arrays is often human cancers but this Issue expands that focus to include areas such as rare conditions, animal breeding and bioinformatics tools. Given the limited scope, the spectrum of papers is nothing short of remarkable and even from a technical point of view these papers will contribute to the field at a general level. Three of the papers published in this Special Issue focus on the use of various SNP array approaches in the analysis of three different cancer types. Two of the papers concentrate on two very different rare conditions, applying the SNP arrays slightly differently. Finally, two other papers evaluate the use of the SNP arrays in the context of genetic analysis of livestock. The findings reported in these papers help to close gaps in the current literature and also to give guidelines for future applications of SNP arrays. PMID:27792140

  8. 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.

  9. Nanocoax Arrays for Sensing Devices

    NASA Astrophysics Data System (ADS)

    Rizal, Binod

    We have adapted a nanocoax array architecture for high sensitivity, all-electronic, chemical and biological sensing. Arrays of nanocoaxes with various dielectric annuli were developed using polymer replicas of Si nanopillars made via soft lithography. These arrays were implemented in the development of two different kinds of chemical detectors. First, arrays of nanocoaxes constructed with different porosity dielectric annuli were employed to make capacitive detectors for gaseous molecules and to investigate the role of dielectric porosity in the sensitivity of the device. Second, arrays of nanocoaxes with partially hollowed annuli were used to fabricate three-dimensional electrochemical biosensors within which we studied the role of nanoscale gap between electrodes on device sensitivity. In addition, we have employed a molecular imprint technique to develop a non-conducting molecularly imprinted polymer thin film of thickness comparable to size of biomolecules as an "artificial antibody" architecture for the detection of biomolecules.

  10. Enthalpy arrays

    PubMed Central

    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.

    2004-01-01

    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. PMID:15210951

  11. 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.

  12. 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.

  13. Infrared Arrays

    NASA Astrophysics Data System (ADS)

    McLean, I.; Murdin, P.

    2000-11-01

    Infrared arrays are small electronic imaging devices subdivided into a grid or `array' of picture elements, or pixels, each of which is made of a material sensitive to photons (ELECTROMAGNETIC RADIATION) with wavelengths much longer than normal visible light. Typical dimensions of currently available devices are about 27-36 mm square, and formats now range from 2048×2048 pixels for the near-infra...

  14. 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.

  15. 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.

  16. Real-ear measurements in conductive hearing loss: discrepancies between probe-tube microphone measurements and sound field test results.

    PubMed

    Cleaver, V C

    1998-06-01

    This study was designed to investigate earlier observations that probe-tube microphone measurements of insertion gain overestimates the functional gain received from hearing aids by users with significant conductive hearing losses. This was originally thought to be due to artefacts in the probe-tube measurement caused by middle ear pathology, but is now believed to be the result of the bone conduction stimulation of the ear exposed to high intensities of airborne sound during sound field threshold measurements. Since the functional gain must relate to the true aided benefit in such cases, these findings suggest that probe-tube microphone measurements in ears with significant air-bone gaps should be interpreted with caution.

  17. 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.

  18. 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.

  19. Linear Methods for Efficient and Fast Separation of Two Sources Recorded with a Single Microphone.

    PubMed

    Bhargava, Saurabh; Blättler, Florian; Kollmorgen, Sepp; Liu, Shih-Chii; Hahnloser, Richard H R

    2015-10-01

    This letter addresses the problem of separating two speakers from a single microphone recording. Three linear methods are tested for source separation, all of which operate directly on sound spectrograms: (1) eigenmode analysis of covariance difference to identify spectro-temporal features associated with large variance for one source and small variance for the other source; (2) maximum likelihood demixing in which the mixture is modeled as the sum of two gaussian signals and maximum likelihood is used to identify the most likely sources; and (3) suppression-regression, in which autoregressive models are trained to reproduce one source and suppress the other. These linear approaches are tested on the problem of separating a known male from a known female speaker. The performance of these algorithms is assessed in terms of the residual error of estimated source spectrograms, waveform signal-to-noise ratio, and perceptual evaluation of speech quality scores. This work shows that the algorithms compare favorably to nonlinear approaches such as nonnegative sparse coding in terms of simplicity, performance, and suitability for real-time implementations, and they provide benchmark solutions for monaural source separation tasks.

  20. Development and Evaluation of Single-Microphone Noise Reduction Algorithms for Digital Hearing Aids

    NASA Astrophysics Data System (ADS)

    Marzinzik, Mark; Kollmeier, Birger

    In this study single-microphone noise reduction procedures were investigated for use in digital hearing aids. One widely reported artifact of most noise suppression systems, the musical noise phenomenon, can partly be overcome by the Ephraim-Malah noise suppression algorithms [1,2]. Based on these algorithms, three different versions have been implemented together with a procedure for automatically updating the noise-spectrum estimate. To evaluate the algorithms, different tests have been performed with six normal-hearing and six hearing-impaired subjects. With `standard' measurement methods no increase in speech intelligibility was found compared to the unprocessed signal. However, benefits with respect to reductions in listener fatigue and in the mental effort needed to listen to speech in noise over longer periods of time were found in this study by use of a newly developed ease-of-listening test. Subsequent paired comparison tests also revealed a clear preference of the hearing-impaired subjects for the noise-reduced signals in situations with rather stationary noise. In the case of strongly fluctuating noise at low SNR, however, the subjects preferred the unprocessed signal due to speech distortions caused by the noise reduction algorithms.

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

    PubMed

    Nishida, H; Komatsuzaki, A; Noguchi, Y

    1997-01-01

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

  2. System identification of velocity mechanomyogram measured with a capacitor microphone for muscle stiffness estimation.

    PubMed

    Uchiyama, Takanori; Tomoshige, Taiki

    2017-04-01

    A mechanomyogram (MMG) measured with a displacement sensor (displacement MMG) can provide a better estimation of longitudinal muscle stiffness than that measured with an acceleration sensor (acceleration MMG), but the displacement MMG cannot provide transverse muscle stiffness. We propose a method to estimate both longitudinal and transverse muscle stiffness from a velocity MMG using a system identification technique. The aims of this study are to show the advantages of the proposed method. The velocity MMG was measured using a capacitor microphone and a differential circuit, and the MMG, evoked by electrical stimulation, of the tibialis anterior muscle was measured five times in seven healthy young male volunteers. The evoked MMG system was identified using the singular value decomposition method and was approximated with a fourth-order model, which provides two undamped natural frequencies corresponding to the longitudinal and transverse muscle stiffness. The fluctuation of the undamped natural frequencies estimated from the velocity MMG was significantly smaller than that from the acceleration MMG. There was no significant difference between the fluctuations of the undamped natural frequencies estimated from the velocity MMG and that from the displacement MMG. The proposed method using the velocity MMG is thus more advantageous for muscle stiffness estimation.

  3. 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.

  4. 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

  5. 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.

  6. 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.

  7. 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.

  8. Optical Processing for Adaptive Phased Array Radar.

    DTIC Science & Technology

    1981-06-01

    the system. A porn I Id model YieIled pooir resuilts , with aI fal clv constant C 70-80 pF value but with a large R range ( 500-50 ohms) from 25-45...associated maps on a disk file. The routine to compute s :wd M is given in Table 3.5. In Fig. 3.10, we see that a 25 element 5 x 5 space-time tenn

  9. Adaptive Arrays for Multiple Simultaneous Desired Signals.

    DTIC Science & Technology

    1983-08-01

    weights [Equation (4)]. Using Equation (6), the inverse of the covariance matrix is given by 5 4 i *ŕm * T ". -1 1 I d Z dij (7) L -I + UT U* 4 di x di...Equations (11) and (12) p k = A k Ik d (14) dk aki*~~* ( ~ dk LJk Udk) and 1 t IjI II di l() 27 x = (1 + t UT U*) Thus, the output SNR of the kth desired...signals are assumed to be of the same frequency. There is no jammer 9 0 dB SIGNAL 10 dB SIGNAL 90 % 90 180 Fiur .dptdpatrnofalier rayo tn strpc lmet

  10. Assistive technology evaluations: Remote-microphone technology for children with Autism Spectrum Disorder.

    PubMed

    Schafer, Erin C; Wright, Suzanne; Anderson, Christine; Jones, Jessalyn; Pitts, Katie; Bryant, Danielle; Watson, Melissa; Box, Jerrica; Neve, Melissa; Mathews, Lauren; Reed, Mary Pat

    The goal of this study was to conduct assistive technology evaluations on 12 children diagnosed with Autism Spectrum Disorder (ASD) to evaluate the potential benefits of remote-microphone (RM) technology. A single group, within-subjects design was utilized to explore individual and group data from functional questionnaires and behavioral test measures administered, designed to assess school- and home-based listening abilities, once with and once without RM technology. Because some of the children were unable to complete the behavioral test measures, particular focus was given to the functional questionnaires completed by primary teachers, participants, and parents. Behavioral test measures with and without the RM technology included speech recognition in noise, auditory comprehension, and acceptable noise levels. The individual and group teacher (n=8-9), parent (n=8-9), and participant (n=9) questionnaire ratings revealed substantially less listening difficulty when RM technology was used compared to the no-device ratings. On the behavioral measures, individual data revealed varied findings, which will be discussed in detail in the results section. However, on average, the use of the RM technology resulted in improvements in speech recognition in noise (4.6dB improvement) in eight children, higher auditory working memory and comprehension scores (12-13 point improvement) in seven children, and acceptance of poorer signal-to-noise ratios (8.6dB improvement) in five children. The individual and group data from this study suggest that RM technology may improve auditory function in children with ASD in the classroom, at home, and in social situations. However, variability in the data and the inability of some children to complete the behavioral measures indicates that individualized assistive technology evaluations including functional questionnaires will be necessary to determine if the RM technology will be of benefit to a particular child who has ASD.

  11. 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.

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

    NASA Astrophysics Data System (ADS)

    Cobos, Maximo; Lopez, JoseJ; Spors, Sascha

    2010-12-01

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

  13. Global Arrays

    SciTech Connect

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

    2015-11-01

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

  14. 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.

  15. 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.

  16. 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.

  17. 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.

  18. 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.

  19. 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.

  20. 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.

  1. Difference Between the Default Telecoil (T-Coil) and Programmed Microphone Frequency Response in Behind-the-Ear (BTE) Hearing Aids

    PubMed Central

    Putterman, Daniel B.; Valente, Michael

    2013-01-01

    Background A telecoil (t-coil) is essential for hearing aid users when listening on the telephone because using the hearing aid microphone when communicating on the telephone can cause feedback due to telephone handset proximity to the hearing aid microphone. Clinicians may overlook the role of the t-coil due to a primary concern of matching the microphone frequency response to a valid prescriptive target. Little has been published to support the idea that the t-coil frequency response should match the microphone frequency response to provide “seamless” and perhaps optimal performance on the telephone. If the clinical goal were to match both frequency responses, it would be useful to know the relative differences, if any, which currently exist between these two transducers. Purpose The primary purpose of this study was to determine if statistically significant differences were present between the mean output (in dB SPL) of the programmed microphone program and the hearing aid manufacturer’s default t-coil program as a function of discrete test frequencies. In addition, pilot data are presented on the feasibility of measuring the microphone and t-coil frequency response with real-ear measures using a digital speech-weighted noise. Research Design A repeated-measures design was utilized for a 2-cc coupler measurement condition. Independent variables were the transducer (microphone; t-coil), and eleven discrete test frequencies (fifteen discrete frequencies in the real-ear pilot condition). Study Sample The study sample was comprised of behind-the-ear (BTE) hearing aids from one manufacturer. Fifty-two hearing aids were measured in a coupler condition, 39 of which were measured in the real-ear pilot condition. Hearing aids were previously programmed and verified using real-ear measures to the NAL-NL1 prescriptive target by a licensed audiologist. Data Collection and Analysis Hearing aid output was measured with a Fonix 7000 hearing aid analyzer (Frye Electronics

  2. Acoustical Direction Finding with Time-Modulated Arrays.

    PubMed

    Clark, Ben; Flint, James A

    2016-12-11

    Time-Modulated Linear Arrays (TMLAs) offer useful efficiency savings over conventional phased arrays when applied in parameter estimation applications. The present paper considers the application of TMLAs to acoustic systems and proposes an algorithm for efficiently deriving the arrival angle of a signal. The proposed technique is applied in the frequency domain, where the signal and harmonic content is captured. Using a weighted average method on harmonic amplitudes and their respective main beam angles, it is possible to determine an estimate for the signal's direction of arrival. The method is demonstrated and evaluated using results from both numerical and practical implementations and performance data is provided. The use of Micro-Electromechanical Systems (MEMS) sensors allows time-modulation techniques to be applied at ultrasonic frequencies. Theoretical predictions for an array of five isotropic elements with half-wavelength spacing and 1000 data samples suggest an accuracy of ± 1 ∘ within an angular range of approximately ± 50 ∘ . In experiments of a 40 kHz five-element microphone array, a Direction of Arrival (DoA) estimation within ± 2 . 5 ∘ of the target signal is readily achieved inside a ± 45 ∘ range using a single switched input stage and a simple hardware setup.

  3. Acoustical Direction Finding with Time-Modulated Arrays

    PubMed Central

    Clark, Ben; Flint, James A.

    2016-01-01

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

  4. 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.

  5. Reconstruction of the Acoustic Field Using a Conformal Array

    NASA Technical Reports Server (NTRS)

    Valdivia, Nichlas P.; Williams, Earl G.; Klos, Jacob

    2006-01-01

    Near-field acoustical holography (NAH) requires the measurement of the near-field pressure field over a conformal and closed surface in order to recover the acoustic field on a nearby surface. We are interested in the reconstruction of the acoustic field over the fuselage of a Boeing 757 airplane when pressure data is available over an array of microphones that are conformal to the fuselage surface. In this case the strict NAH theory does not hold, but still there are techniques used to overcome this difficulty. The best known is patch NAH, which has been used for planar surfaces. In this work we will discuss two new techniques used for surfaces with an arbitrarily shape: patch inverse boundary element methods (IBEM) and patch equivalent sources method (ESM). We will discuss the theoretical justification of the method and show reconstructions for in-flight data taken inside a Boeing 757 airplane.

  6. 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.

  7. 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.

  8. Hair-Cell Versus Afferent Adaptation in the Semicircular Canals

    PubMed Central

    Rabbitt, R. D.; Boyle, R.; Holstein, G. R.; Highstein, S. M.

    2010-01-01

    The time course and extent of adaptation in semicircular canal hair cells was compared to adaptation in primary afferent neurons for physiological stimuli in vivo to study the origins of the neural code transmitted to the brain. The oyster toadfish, Opsanus tau, was used as the experimental model. Afferent firing-rate adaptation followed a double-exponential time course in response to step cupula displacements. The dominant adaptation time constant varied considerably among afferent fibers and spanned six orders of magnitude for the population (~1 ms to >1,000 s). For sinusoidal stimuli (0.1–20 Hz), the rapidly adapting afferents exhibited a 90° phase lead and frequency-dependent gain, whereas slowly adapting afferents exhibited a flat gain and no phase lead. Hair-cell voltage and current modulations were similar to the slowly adapting afferents and exhibited a relatively flat gain with very little phase lead over the physiological bandwidth and dynamic range tested. Semicircular canal microphonics also showed responses consistent with the slowly adapting subset of afferents and with hair cells. The relatively broad diversity of afferent adaptation time constants and frequency-dependent discharge modulations relative to hair-cell voltage implicate a subsequent site of adaptation that plays a major role in further shaping the temporal characteristics of semicircular canal afferent neural signals. PMID:15306633

  9. Evolutionary Technique for Designing Optimized Arrays

    NASA Astrophysics Data System (ADS)

    Villazón, J.; Ibañez, A.

    2011-06-01

    Many ultrasonic inspection applications in the industry could benefit from the use of phased array distributions specifically designed for them. Some common design requirements are: to adapt the shape of the array to that of the part to be inspected, to use large apertures for increasing lateral resolution, to find a layout of elements that avoids artifacts produced by lateral and/or grating lobes, to maintain the total number of independent elements (and the number of control channels) as low as possible to reduce complexity and cost of the inspection system. Recent advances in transducer technology have made possible to design and build arrays whit non-regular layout of elements. In this paper we propose to use Evolutionary Algorithms to find layouts of ultrasonic arrays (whether 1D or 2D array) that approach a set of specified beampattern characteristics using a low number of elements.

  10. Adaptive aperture synthesis

    NASA Astrophysics Data System (ADS)

    Johnson, A. M.; Zhang, S.; Mudassar, A.; Love, G. D.; Greenaway, A. H.

    2005-12-01

    High-resolution imaging can be achieved by optical aperture synthesis (OAS). Such an imaging process is subject to aberrations introduced by instrumental defects and/or turbulent media. Redundant spacings calibration (RSC) is a snapshot calibration technique that can be used to calibrate OAS arrays without use of assumptions about the object being imaged. Here we investigate the analogies between RSC and adaptive optics in passive imaging applications.

  11. MITRE Adaptive Processing Capability

    DTIC Science & Technology

    1994-06-01

    gathering, Funded Research and Development transfer, processing , and interpretation of Center (FFRDC) under the primary data are provided. A strong state-of...1988: Unisys Reston Technology Center, Reston, VA Dr. Bronez was a Member of the Technical Staff. He performed research on signal processing and... processing , mathematical research , and sensor array processing . He was Project Leader and Principal Investigator for projects in adaptive beamforming

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

    NASA Astrophysics Data System (ADS)

    Avison, Janine; Barham, Richard

    2014-01-01

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

  13. An experimental assessment of the use of ground-level microphones to measure the fly-over noise of jet-engined aircraft

    NASA Astrophysics Data System (ADS)

    Payne, R. C.

    1993-01-01

    During aircraft flight trials to measure the noise levels of six different military jet aircraft types in low altitude high speed operations, noise measurements were performed using microphones at ground level and at a height of 1.2 m. The program provided reliable data on the difference between sound pressure levels from the two microphone arrangements, for sound incident over a range of angles, from 0 deg (aircraft overhead) to approximately 80 deg. Substantial differences from ground level to 1.2 m were observed in measurements of maximum perceived noise level, effective perceived noise level and maximum A-weighted sound pressure level. For sound waves incident to the ground at angles less than approximately 60 deg from vertical, these differences were found to be independent of angle of incidence for all the six aircraft and all flight procedures. Within this range of sound incidence angles the ground plane arrangement produced data that closely approximated pressure doubled values. The conventional 1.2 m high microphone gave rise to noise levels approximately 4 dB lower. For sound incident at angles greater than 60 deg from vertical, the difference between noise levels measured using the two microphone configurations was found to depend on angle of incidence, reducing to zero at approximately 75 deg. When noise measurements are made using the ground plane arrangement, the effects of meteorological conditions must be considered in relation to sound incident at angles greater than approximately 60 deg.

  14. Diode Laser Arrays

    NASA Astrophysics Data System (ADS)

    Botez, Dan; Scifres, Don R.

    2005-11-01

    Contributors; 1. Monolithic phase-locked semiconductor laser arrays D. Botez; 2. High power coherent, semiconductor laser master oscillator power amplifiers and amplifier arrays D. F. Welch and D. G. Mehuys; 3. Microoptical components applied to incoherent and coherent laser arrays J. R. Leger; 4. Modeling of diode laser arrays G. R. Hadley; 5. Dynamics of coherent semiconductor laser arrays H. G. Winfuland and R. K. Defreez; 6. High average power semiconductor laser arrays and laser array packaging with an emphasis for pumping solid state lasers R. Solarz; 7. High power diode laser arrays and their reliability D. R. Scifres and H. H. Kung; 8. Strained layer quantum well heterostructure laser arrays J. J. Coleman; 9. Vertical cavity surface emitting laser arrays C. J. Chang-Hasnain; 10. Individually addressed arrays of diode lasers D. Carlin.

  15. Improving quality and intelligibility of speech using single microphone for the broadband fMRI noise at low SNR.

    PubMed

    Vahanesa, Chetan; Reddy, Chandan K A; Panahi, Issa M S

    2016-08-01

    Functional Magnetic Resonance Imaging (fMRI) is used in many diagnostic procedures for neurological related disorders. Strong broadband acoustic noise generated during fMRI scan interferes with the speech communication between the physician and the patient. In this paper, we propose a single microphone Speech Enhancement (SE) technique which is based on the supervised machine learning technique and a statistical model based SE technique. The proposed algorithm is robust and computationally efficient and has capability to run in real-time. Objective and Subjective evaluations show that the proposed SE method outperforms the existing state-of-the-art algorithms in terms of quality and intelligibility of the recovered speech at low Signal to Noise Ratios (SNRs).

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

    NASA Astrophysics Data System (ADS)

    Gibian, G. L.

    1980-03-01

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

  17. 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.

  18. Reduced Complexity High Performance Array Processing

    DTIC Science & Technology

    1998-01-01

    constrained beamforming,” IEEE Transactions on Aerospace and Electronic Systems, January 1982. [5] J . Scott Goldstein and Irving S. Reed, “Theory of...partially adaptive radar,” IEEE Transactions on Aerospace and Electronic Systems, October 1997. [6] J . Scott Goldstein, Irving S. Reed, and John A...Computers. [7] Barry D. Van Veen, “Eigenstructure-based partially adaptive array design,” IEEE Transactions on Antennas and Propagation, March 1988. [8] J

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

    PubMed Central

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

    2013-01-01

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

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

    PubMed

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

    2014-02-01

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